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 .
Response to Arguments
Applicant’s arguments with respect to claim(s) 1 and 3-29 have been considered and are addressed below.
As to Applicant’s argument, on pages 16 and 17, that “Jensen does not show different modifiers”, the Examiner agrees with this assessment, as Jensen, at FIG. 4A, block P(Θ)(g(k’,m,Θ’)), and col. 33 ln. 26 – col. 34 ln. 36, describes a single processing algorithm that determines the optimized parameter setting Θ’, which, as described a col. 34 ln. 65 – col. 35 ln. 1, is determined using the iterative process described in FIG. 1B and FIG. 6.
However, this argument is now moot, as Usher (US 2019/0313196) is now relied upon to teach different signal modifiers. See the updated 35 U.S.C. 103 rejections below.
As to Applicant’s arguments, on page 17, that “it is impossible to compare the two signals, since according to Jensen there isn’t any possibility to generate two comparable signals” and “Jensen fails to disclose a multi-value, multi-criteria selection between a first and a second modified audio signal”, the examiner contests that Jensen does describe taking an input audio signal and iteratively modifying using a plurality of beamformer settings in order to determine the optimal beamformer setting that produces the desired speech intelligibility level while also optimizing other criteria, such as least modification of the original signal, least total noise power reduction, and best maintained direction of the spatial minima of the beam pattern (see col. 28 lns. 48-65, FIG. 1B, col. 22 lns. 12-18 and col. 22 ln. 60 – col. 23 ln. 5).
Thus, Jensen does describe comparing processed signals via an iterative process of generating signals to compare, where the comparison is a multi-value, multi-criteria selection between at least a first and a second modified audio signal.
Claim Objections
Claims 1, 4, 11, 15, 20, 21, 28 and 29 are objected to because of the following informalities:
Regarding claims 1 and 20, lines 9 and 12 describe, “a first evaluation values” and “a second evaluation values”. The examiner recommends “first evaluation values” and “second evaluation values”.
Regarding claim 4, lines 1-2 describe, “wherein the evaluation criterions is the perceptual similarity…” The examiner recommends, “wherein one of the at least two independent evaluation criterions is the perceptual similarity…”
Regarding claim 11, lines 1-2 describes, “wherein the first and/or second evaluation values is dependent on…” The examiner recommends, “wherein the first and/or second evaluation values are dependent on…”
Regarding claim 15, line 3 describes, “preference; wherein the evaluation criterions is dependent on the…” The examiner recommends, “preference; wherein the evaluation criterions are dependent on the…”
Regarding claim 20, line 15 describes, “electing the first or second modified audio signal”. The examiner recommends, “selecting the first or second modified audio signal”.
Regarding claim 21, lines 7-8 describe, “an evaluator for evaluating the first modified audio signal with respect to evaluation criteria to acquire first evaluation values…” The examiner recommends, “an evaluator for evaluating the first modified audio signal with respect to evaluation criterions to acquire first evaluation values…” in order to maintain consistency with the later use of “evaluation criterions” in the claim set.
Regarding claims 28 and 29, line 1 describes, “wherein for a fist timeframe of the initial audio” The examiner recommends, “wherein for a first timeframe of the initial audio”
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 and 3-29 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.
Claims 1, 20, and 21 recite the limitation "the at least two independent evaluation criterions" in the last limitation. There is insufficient antecedent basis for this limitation in the claim.
Claim 3 recites, “The method according to claim 1, evaluation criterions, such that respective first evaluation values describing a degree of fulfilment…” It is unclear as to why claim 3 starts with “evaluation criterions”, since it doesn’t appear to fit with the rest of the claim.
Claims 22 and 25 recite, “wherein the first signal modifier and the second signal modifier processing said received initial audio signal or the same received initial audio signal”. It is unclear as to how processing said received initial audio signal is different from processing the same received initial audio signal, since they both appear to be referencing the initial audio signal introduced in claims 21, and 1, respectively. Also, there is insufficient antecedent basis for “the same received initial audio signal”.
The remaining dependent claims are rejected due to their respective dependencies, as they inherit the indefiniteness issues from the independent claims and do not correct them.
The following is a quotation of 35 U.S.C. 112(d):
(d) REFERENCE IN DEPENDENT FORMS.—Subject to subsection (e), a claim in dependent form shall contain a reference to a claim previously set forth and then specify a further limitation of the subject matter claimed. A claim in dependent form shall be construed to incorporate by reference all the limitations of the claim to which it refers.
The following is a quotation of pre-AIA 35 U.S.C. 112, fourth paragraph:
Subject to the following paragraph [i.e., the fifth paragraph of pre-AIA 35 U.S.C. 112], a claim in dependent form shall contain a reference to a claim previously set forth and then specify a further limitation of the subject matter claimed. A claim in dependent form shall be construed to incorporate by reference all the limitations of the claim to which it refers.
Claims 22 and 25 rejected under 35 U.S.C. 112(d) or pre-AIA 35 U.S.C. 112, 4th paragraph, as being of improper dependent form for failing to further limit the subject matter of the claim upon which it depends, or for failing to include all the limitations of the claim upon which it depends.
Claims 22 and 25 describe, “the first signal modifier and the second signal modifier processing said received initial audio signal or the same received initial audio signal.” It is unclear as to how these claims further limit claims 21 and 1, respectively, since claim 1 describes “modifying the received initial audio signal by use of a first signal modifier” and “modifying said received initial audio signal by use of a second signal modifier”, claim 21 describes equivalent language, and “processing” the initial audio signal has a broader scope than “modifying” it.
Applicant may cancel the claim(s), amend the claim(s) to place the claim(s) in proper dependent form, rewrite the claim(s) in independent form, or present a sufficient showing that the dependent claim(s) complies with the statutory requirements.
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, 4-6, 8, and 11-29 are rejected under 35 U.S.C. 103 as being unpatentable over Jensen et al. (US 10,701,494), hereinafter “Jensen” in view of Usher (US 2019/0313196), hereinafter “Usher”.
Regarding Claim 1, Jensen teaches a method for processing an initial audio signal comprising a target portion and a side portion, comprising:
receiving of the initial audio signal (col. 35 lns. 45-50: “FIG. 5 shows a flow diagram for a method of operating a hearing aid according to a first embodiment of the present disclosure. The hearing aid is adapted for being worn by a user. The method comprises S1. receiving sound comprising speech from the environment of the user;”);
modifying the received initial audio signal by use of a first signal modifier to acquire a first modified audio signal (col. 28 lns. 48-63: “We now outline a procedure to find the desired beamformer settings Θ which achieve a desired speech intelligibility level. In principle, the search for these settings may be divided into the following three situations:
i) the desired speech intelligibility level can be achieved (or is exceeded) without any beamforming,
ii) the set of most aggressive beamformers are not sufficient to achieve the desired speech intelligibility, and
iii) one or more beamformer settings exist, that lead to the desired speech intelligibility level. In this situation, the beamformer setting (amongst the settings leading to the desired intelligibility) is chosen, which optimize other criteria, e.g. least modification of the original signal, least total noise power reduction (e.g. to maintain awareness of the acoustic environment), the setting that maintain the direction of the spatial minima of the beam pattern, etc.”
Also see col. 22 lns. 12-21: “A speech intelligibility measure of one or more processed or un-processed signals is determined at successive points in time t. As indicated in FIG. 1B by unit or process step ‘t=t+1’. The successive points in time may e.g. be every successive time frame (defined by time frame index m) of the respective signals. Alternatively, successive points in time may indicate a lower rate, e.g. every 10.sup.th time frame.
The controller is configured to control the processor to provide that the resulting signal y.sub.res at a current point in time t is equal to one of the electric input signals y”.
Finally, see col. 22 ln. 60 – col. 23 ln. 5: “In case the statement ‘I(y.sub.p(Θ1,t))≤I.sub.des?’ is false (branch ‘No’), i.e. if the speech intelligibility measure I of the processed signal y.sub.p(Θ1,t) is larger than the desired value I.sub.des, the controller is further configured to determine a second parameter setting Θ′ of the processing algorithm under the constraint that the second processed signal y.sub.p(Θ′) exhibits the desired value I.sub.des of the speech intelligibility measure, and to control the processor to provide that the resulting signal y.sub.res at the current point in time t is equal to the second, optimized, processed signal y.sub.p(Θ′) (cf. respective units or process steps, ‘Find Θ′ providing I(y.sub.p(Θ,t)=I.sub.des. Set y.sub.res=y.sub.p(Θ′,t)’, and advance time to the next time index ‘t=t+1’).”);
modifying said received initial audio signal by use of [the first] signal modifier to acquire a second modified audio signal (Multiple beamformer settings are determined - col. 28 lns. 48-63: “We now outline a procedure to find the desired beamformer settings Θ which achieve a desired speech intelligibility level. In principle, the search for these settings may be divided into the following three situations:
i) the desired speech intelligibility level can be achieved (or is exceeded) without any beamforming,
ii) the set of most aggressive beamformers are not sufficient to achieve the desired speech intelligibility, and
iii) one or more beamformer settings exist, that lead to the desired speech intelligibility level. In this situation, the beamformer setting (amongst the settings leading to the desired intelligibility) is chosen, which optimize other criteria, e.g. least modification of the original signal, least total noise power reduction (e.g. to maintain awareness of the acoustic environment), the setting that maintain the direction of the spatial minima of the beam pattern, etc.”)
Also see col. 22 lns. 12-21: “A speech intelligibility measure of one or more processed or un-processed signals is determined at successive points in time t. As indicated in FIG. 1B by unit or process step ‘t=t+1’. The successive points in time may e.g. be every successive time frame (defined by time frame index m) of the respective signals. Alternatively, successive points in time may indicate a lower rate, e.g. every 10.sup.th time frame.
The controller is configured to control the processor to provide that the resulting signal y.sub.res at a current point in time t is equal to one of the electric input signals y”.
Finally, see col. 22 ln. 60 – col. 23 ln. 5: “In case the statement ‘I(y.sub.p(Θ1,t))≤I.sub.des?’ is false (branch ‘No’), i.e. if the speech intelligibility measure I of the processed signal y.sub.p(Θ1,t) is larger than the desired value I.sub.des, the controller is further configured to determine a second parameter setting Θ′ of the processing algorithm under the constraint that the second processed signal y.sub.p(Θ′) exhibits the desired value I.sub.des of the speech intelligibility measure, and to control the processor to provide that the resulting signal y.sub.res at the current point in time t is equal to the second, optimized, processed signal y.sub.p(Θ′) (cf. respective units or process steps, ‘Find Θ′ providing I(y.sub.p(Θ,t)=I.sub.des. Set y.sub.res=y.sub.p(Θ′,t)’, and advance time to the next time index ‘t=t+1’).”);
evaluating the first modified audio signal with respect to evaluation criterions to acquire a first evaluation values describing a degree of fulfilment of the evaluation criterions (col. 29 lns. 33-46: “3) a) Identify the (potentially multiple) parameter settings Θ which achieve I=I.sub.desired, and which process the incoming signal the least, e.g., the beamformer settings which reduce the total noise power at the output of the beamformer the least, or the beamformer settings which lead to maximum total signal loudness, the beamformer settings that best maintain the direction and value of the spatial minima of the beam pattern, etc. (several such secondary requirements may be envisioned). This may, e.g., be done by introducing the Karush-Kuhn Tucker conditions (cf. p 243 in [4]) and identifying the beamformer parameter settings, which satisfy these conditions, see [2, 3] for examples.”);
evaluating the second modified audio signal with respect to the evaluation criterions to acquire a second evaluation values describing a degree of fulfilment of the evaluation criterions (Multiple beamformer settings are evaluated - col. 29 lns. 33-46: “3) a) Identify the (potentially multiple) parameter settings Θ which achieve I=I.sub.desired, and which process the incoming signal the least, e.g., the beamformer settings which reduce the total noise power at the output of the beamformer the least, or the beamformer settings which lead to maximum total signal loudness, the beamformer settings that best maintain the direction and value of the spatial minima of the beam pattern, etc. (several such secondary requirements may be envisioned). This may, e.g., be done by introducing the Karush-Kuhn Tucker conditions (cf. p 243 in [4]) and identifying the beamformer parameter settings, which satisfy these conditions, see [2, 3] for examples.”); and
selecting the first or second modified audio signal dependent on the respective first or second evaluation values; wherein selecting is performed based on a plurality of independent first evaluation values and independent second evaluation values (col. 28 lns. 56-63: “iii) one or more beamformer settings exist, that lead to the desired speech intelligibility level. In this situation, the beamformer setting (amongst the settings leading to the desired intelligibility) is chosen, which optimize other criteria, e.g. least modification of the original signal, least total noise power reduction (e.g. to maintain awareness of the acoustic environment), the setting that maintain the direction of the spatial minima of the beam pattern, etc.”);
wherein the at least two independent evaluation criterions are evaluated separately, and wherein the two independent evaluation criterions are two different evaluation criterions out of the group comprising:
- perceptual similarity, described by a first and second perceptual similarity value, the first and the second perceptual similarity value describing a perceptual similarity between the respective first and second modified audio signal and the initial audio signal AS (col. 28 lns. 56-63: “iii) one or more beamformer settings exist, that lead to the desired speech intelligibility level. In this situation, the beamformer setting (amongst the settings leading to the desired intelligibility) is chosen, which optimize other criteria, e.g. least modification of the original signal [emphasis added], least total noise power reduction (e.g. to maintain awareness of the acoustic environment), the setting that maintain the direction of the spatial minima of the beam pattern, etc.”);
- speech intelligibility, in the form of calculated values of speech intelligibility to be compared with targets or thresholds (col. 29 lns. 1-4: “Let us assume that a value I.sub.desired reflecting the desired level of speech intelligibility is available. This value could, for example, have been established when the hearing aid system was fitted by the audiologist.”
Also see col. 29 lns. 33-36: “3) a) Identify the (potentially multiple) parameter settings Θ which achieve I=I.sub.desired, and which process the incoming signal the least,”);
- loudness, described by a loudness value (col. 29 lns. 33-46: “3) a) Identify the (potentially multiple) parameter settings Θ which achieve I=I.sub.desired, and which process the incoming signal the least, e.g., the beamformer settings which reduce the total noise power at the output of the beamformer the least, or the beamformer settings which lead to maximum total signal loudness [emphasis added], the beamformer settings that best maintain the direction and value of the spatial minima of the beam pattern, etc.);
- sound pattern (col. 29 lns. 33-46: “3) a) Identify the (potentially multiple) parameter settings Θ which achieve I=I.sub.desired, and which process the incoming signal the least, e.g., the beamformer settings which reduce the total noise power at the output of the beamformer the least, or the beamformer settings which lead to maximum total signal loudness, the beamformer settings that best maintain the direction and value of the spatial minima of the beam pattern, [emphasis added] etc.);
- spatiality (col. 29 lns. 33-46: “3) a) Identify the (potentially multiple) parameter settings Θ which achieve I=I.sub.desired, and which process the incoming signal the least, e.g., the beamformer settings which reduce the total noise power at the output of the beamformer the least, or the beamformer settings which lead to maximum total signal loudness, the beamformer settings that best maintain the direction and value of the spatial minima of the beam pattern, [emphasis added] etc.).
Jensen doesn’t describe modifying said received initial audio signal by use of a second signal modifier to acquire a second modified audio signal.
However, Usher describes a system and method including modifying said received initial audio signal by use of a first signal modifier to acquire a first modified audio signal and modifying said received initial audio signal by use of a second signal modifier to acquire a second modified audio signal (FIG. 5, ¶ [0058]-[0065], and claim 1: “[0058] FIG. 5 shows a detailed exemplary method to generate a DRCF curve to optimize speech intelligibility, and comprises the steps of:
[0059] 1. 502 Receiving a selected audio signal to the earphone DSP. The audio signal is reproduced from a digital storage file, and may be a speech or music audio signal.
[0060] 2. 504 Applying a gain to the received audio signal to generate a modified input audio signal.
[0061] 3. 506 Generating a first dynamic range compression parameter set A, where the parameters comprise a compression ratio value, an expansion ratio value, threshold value, and gate value 508.
[0062] 4. 510 Generating a second dynamic range compression parameter set B, where the parameters also comprise a compression ratio value, an expansion ratio value, threshold value, and gate value 512.
[0063] 5. The modified input signal is processed with a first dynamic range compressor using the DRC parameter set A 514 to produce an output signal A.
[0064] 6. The modified input signal is processed with a [second] dynamic range compressor using the DRC parameter set B 516 to produce an output signal B.
[0065] 7. A preference test is conducted 518 by the user with a user selection interface 520. The preference test can be in the form of a standard paired comparison AB test, where two audio signals are presented A and B, A and O, or B and O, and the user determines which signal they prefer. In one exemplary embodiment, the user is asked to determine which signal, A or B, sounds the clearest in terms of speech intelligibility. Using this methodology, an optimum DRCF can be determined that optimizes speech intelligibility.”).
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to include in Jensen a system and method including modifying said received initial audio signal by use of a first signal modifier to acquire a first modified audio signal and modifying said received initial audio signal by use of a second signal modifier to acquire a second modified audio signal, as taught by Usher, in order to reduce the time associated with determining the optimal beamformer setting by testing out multiple beamformer settings in parallel, instead of iteratively, which improves the speech intelligibility of the system, as it can reach the optimal beamformer setting in less time.
Regarding Claim 4, Jensen teaches the method to according to claim 1, wherein the evaluation criterions is the perceptual similarity, and wherein step c comprises the substeps of
comparing received initial audio signal with the first modified audio signal to acquire a first perceptual similarity value as first evaluation value describing the perceptual similarity between the initial audio signal and the first modified audio signal (col. 28 lns. 56-63: “iii) one or more beamformer settings exist, that lead to the desired speech intelligibility level. In this situation, the beamformer setting (amongst the settings leading to the desired intelligibility) is chosen, which optimize other criteria, e.g. least modification of the original signal [emphasis added], least total noise power reduction (e.g. to maintain awareness of the acoustic environment), the setting that maintain the direction of the spatial minima of the beam pattern, etc.”); and
comparing the received initial audio signal with the second modified audio signal to acquire a second perceptual similarity value as second evaluation value describing the perceptual similarity between the initial audio signal and the second modified audio signal (col. 28 lns. 56-63: “iii) one or more beamformer settings exist, that lead to the desired speech intelligibility level. In this situation, the beamformer setting (amongst the settings leading to the desired intelligibility) is chosen, which optimize other criteria, e.g. least modification of the original signal [emphasis added], least total noise power reduction (e.g. to maintain awareness of the acoustic environment), the setting that maintain the direction of the spatial minima of the beam pattern, etc.”).
Regarding Claim 5, Jensen teaches the method according to claim 4, wherein the first modified audio signal is selected, when the first perceptual similarity value is higher than the second perceptual similarity value so as to indicate a higher perceptual similarity of the first modified audio signal (col. 28 lns. 56-63: “iii) one or more beamformer settings exist, that lead to the desired speech intelligibility level. In this situation, the beamformer setting (amongst the settings leading to the desired intelligibility) is chosen, which optimize other criteria, e.g. least modification of the original signal [emphasis added], least total noise power reduction (e.g. to maintain awareness of the acoustic environment), the setting that maintain the direction of the spatial minima of the beam pattern, etc.”); and
wherein the second modified audio signal is selected when the second perceptual similarity value is higher than the first perceptual similarity value so as to indicate a higher perceptual similarity of the second modified audio signal (col. 28 lns. 56-63: “iii) one or more beamformer settings exist, that lead to the desired speech intelligibility level. In this situation, the beamformer setting (amongst the settings leading to the desired intelligibility) is chosen, which optimize other criteria, e.g. least modification of the original signal [emphasis added], least total noise power reduction (e.g. to maintain awareness of the acoustic environment), the setting that maintain the direction of the spatial minima of the beam pattern, etc.”).
Regarding Claim 6, Jensen teaches the method to according to claim 1, further comprising outputting the first or second modified audio signal dependent on the selection of step d (col. 21 lns. 10-29: “the adjustment unit is configured to adjust the parameter setting Θ to provide a second (preferably optimized) parameter setting Θ′ that provides the desired speech intelligibility I.sub.des of the second processed signal y.sub.p(Θ′) to be presented to the user as the resulting signal y.sub.res”).
Regarding Claim 8, Jensen teaches the method according to claim 1, wherein the target portion is a speech portion of the initial audio signal and the side portion is an ambient noise portion of the audio signal (col. 36 lns. 11-21: “FIG. 6 shows a flow diagram for a method of operating a hearing aid according to a second embodiment of the present disclosure. FIG. 6 shows a flow diagram for a method of operating a hearing aid comprising a multi-input beamformer and providing a resulting signal y.sub.res according to an embodiment of the present disclosure. The method comprises—at a given point in time t the following processes
A1. Determine SNR for an electric input signal y.sub.ref received at a reference microphone;
A2. Determine a measure I of a users' speech intelligibility I(y.sub.ref) of the unprocessed electric input signal y.sub.ref;”).
Regarding Claim 11, Jensen teaches the method according to claim 1, wherein the first and/or second evaluation values is dependent on a physical parameter of the first or second modified audio signal, a volume level of the first or second modified audio signal, a psychoacoustic acoustic parameter for the first or second modified audio signal, a loudness information of the first or second modified audio signal, a pitch information of the first or second modified audio signal, and/or a perceived source width information of the first or second modified audio signal (col. 29 lns. 33-46: “3) a) Identify the (potentially multiple) parameter settings Θ which achieve I=I.sub.desired, and which process the incoming signal the least, e.g., the beamformer settings which reduce the total noise power at the output of the beamformer the least, or the beamformer settings which lead to maximum total signal loudness, the beamformer settings that best maintain the direction and value of the spatial minima of the beam pattern, etc. (several such secondary requirements may be envisioned). This may, e.g., be done by introducing the Karush-Kuhn Tucker conditions (cf. p 243 in [4]) and identifying the beamformer parameter settings, which satisfy these conditions, see [2, 3] for examples.”).
Regarding Claim 12, Jensen describes the method according to claim 1, wherein the first and/or second signal modifier is configured to perform an SNR increase, a dynamic compression, an SNR increase for the initial audio signal, and/or a dynamic compression of the initial audio signal (col. 3 lns. 23-37: “In an embodiment, the first parameter setting Θ1 is a default setting. The first parameter setting Θ1 may be a setting that maximizes a signal to noise ratio (SNR) or the speech intelligibility measure I of the first processed signal y.sub.p(Θ1). In an embodiment, the second (optimized) parameter setting Θ′ is used by the one or more processing algorithms to process the number of electric input signal(s), and to provide a second (optimized) processed signal y.sub.p(Θ′) (yielding the desired level of speech intelligibility to the user, as reflected in the desired value I.sub.des of the speech intelligibility measure). The SNR may preferably be determined in a time-frequency framework, e.g. per TF-unit, cf. e.g. FIG. 3B). In an embodiment, the speech intelligibility measure I is a monotonous function of the signal to noise ratio.”); and/or
wherein modifying comprises increasing the target portion, increasing a frequency weighting for the target portion, dynamically compressing the target portion, decreasing the side portion, decreasing a frequency weighting for the side portion, if the initial audio signal comprises a separate target portion and a separate side portion; and/or
wherein modifying comprises performing a separation of the target portion and the side portion, if the initial audio signal comprises a combined target portion and side portion.
Regarding Claim 13, Jensen describes the method according to claim 1, wherein selecting is performed taking into consideration one or more of the below factors:
- grade of hardness of hearing for hearing-impaired persons (col 5 lns. 18-28: “The hearing device may be adapted to a users' hearing profile, e.g. to compensate for a hearing impairment of the user. The hearing profile of the user may be defined by a parameter set Φ. The parameter set Φ may e.g. define the user's (frequency dependent) hearing thresholds (or their deviation from normal; e.g. reflected in an audiogram). In an embodiment, one of the ‘one or more processing algorithms’, is configured to compensate for a hearing loss of the user. In an embodiment, a compressive amplification algorithm (for adapting the input signal(s) to a user's needs) forms part of the ‘one or more processing algorithms’.”);
- individual hearing performance;
- individual frequency-dependent hearing performance;
- individual preference;
- individual preference regarding signal modification rate.
Regarding Claim 14, Jensen describes the method according to claim 1, wherein modifying and/or comparing is performed taking into consideration one or more of the below factors:
- grade of hardness of hearing for hearing-impaired persons (col 5 lns. 18-28: “The hearing device may be adapted to a users' hearing profile, e.g. to compensate for a hearing impairment of the user. The hearing profile of the user may be defined by a parameter set Φ. The parameter set Φ may e.g. define the user's (frequency dependent) hearing thresholds (or their deviation from normal; e.g. reflected in an audiogram). In an embodiment, one of the ‘one or more processing algorithms’, is configured to compensate for a hearing loss of the user. In an embodiment, a compressive amplification algorithm (for adapting the input signal(s) to a user's needs) forms part of the ‘one or more processing algorithms’.”);
- individual hearing performance;
-individual frequency-dependent hearing performance;
-individual preference;
-individual preference regarding signal modification rate.
Regarding Claim 15, Jensen describes the method according to claim 1, wherein the method further comprises receiving an information on an optimization target defining individual preference; wherein the evaluation criterions is dependent on the optimization target (col. 20 lns. 15-20: “The inputs further comprise c) a desired value I.sub.des of the speech intelligibility measure (e.g. stored in a memory, e.g. configurable via a user interface), d) a parameter set Φ indicative of a hearing profile of the user (e.g. reflecting a normal hearing or a hearing impairment).”); or
wherein modifying and/or evaluating and/or selecting is dependent on the optimization target; or
wherein a weighting of independent first and second evaluation values describing independent evaluation criterions for selecting is dependent on the optimization target.
Regarding Claim 16, Jensen describes the method according to claim 4, wherein comparing is performed for the entire initial audio signal and the entire first and second modified audio signal (col. 28 lns. 56-63: “iii) one or more beamformer settings exist, that lead to the desired speech intelligibility level. In this situation, the beamformer setting (amongst the settings leading to the desired intelligibility) is chosen, which optimize other criteria, e.g. least modification of the original signal, least total noise power reduction (e.g. to maintain awareness of the acoustic environment), the setting that maintain the direction of the spatial minima of the beam pattern, etc.”)); and/or
for the target portion of the individual audio signal and a respective target portion of the first and second modified audio signal; and/or
for the side portion of the initial audio signal and the side portion on the first and second modified audio portion.
Regarding Claim 17, Jensen describes the method according to claim 1, wherein the initial audio signal comprises a plurality of time frames and wherein steps a-d are repeated for each time frame (col. 12 lns. 46-55: “The method is repeated over time, e.g. according to a predefined scheme, e.g. periodically, e.g. every time instance m, e.g. for every time frame of a signal of the forward path. In an embodiment, the method is repeated every N.sup.th time frame, e.g. every N=10 time frames or every N=100 time frames. In an embodiment, N is adaptively determined in dependence of the electric input signal, and/or of one or more sensor signals (e.g. indicative of a current acoustic environment of the user, and/or of a mode of operation of the hearing device, e.g. a battery status indication).”); and/or
wherein the steps a-d are repeated for a time portion or time frame of a scene of the initial audio signal.
Regarding Claim 18, Jensen describes the method according to claim 1, wherein an adaption of the initial audio signal comprising a plurality of time frames is performed for the time frames for which the adaption is applied and for the other time frames in order to maintain a perceptual continuity (col. 10 lns. 43-58: “In a still further aspect, a hearing device, e.g. a hearing aid, is provided. The hearing device comprises a processor for applying one or more processing algorithms to an electric input signal y representing sound, e.g. speech, a speech intelligibility estimator providing an estimate I of a user's intelligibility of said sound at a current time m from said electric input signal y (m), a predictor of a current value, e.g. a current time frame, of the electric input signal y(m) from previous values of the input signal y(m−1), . . . , y(m−N), e.g. N previous time frames, of the electric input signal, a controller configured to control the speech intelligibility estimator in dependence of the estimated predictability of the sound signal, to thereby provide a modified speech intelligibility estimate.”) or
wherein an adaption of the initial audio signal comprising a plurality of time frames is performed for the time frames for which the adaption is applied and in an interpolated manner for the other time frames in order to maintain a perceptual continuity; and/or
wherein the adaption of a first and a second subsequent time frame is performed such that a transition between the first and the second subsequent time frame is formed in order to maintain a perceptual continuity.
Regarding Claim 19, Jensen describes the method according to claim 1, wherein the method further comprises the initial steps of:
analyzing the initial audio portion in order to determine a speech portion (col. 8 lns. 42-56: “In a particular embodiment, the hearing device comprises a voice detector (VD) for estimating whether or not (or with what probability) an input signal comprises a voice signal (at a given point in time). A voice signal is in the present context taken to include a speech signal from a human being. It may also include other forms of utterances generated by the human speech system (e.g. singing). In an embodiment, the voice detector unit is adapted to classify a current acoustic environment of the user as a VOICE or NO-VOICE environment. This has the advantage that time segments of the electric microphone signal comprising human utterances (e.g. speech) in the user's environment can be identified, and thus separated from time segments only (or mainly) comprising other sound sources (e.g. artificially generated noise).”);
comparing the speech portion and the ambient noise portion in order to evaluate on a speech intelligibility of the initial audio signal (col. 29 lns. 6-11: “1) a) Compute SNR (k,m,Φ,Θ) for the situations where the beamforming system is absent (for the example above, this situation is described by Θ={α.sub.k,m=0}. b) Compute the resulting estimated speech intelligibility I=ƒ(SNR(k,m,Φ,Θ)).”); and
activating the first and/or second signal modifier for modifying, if a value indicative for the speech intelligibility is below a threshold (col. 29 lns. 12-22: “ c) If I≥I.sub.desired, the unprocessed signal is already sufficiently understandable, and the beamforming system should remain absent. Otherwise, continue to Step 2 below.
2) a) Compute SNR (k,m,Φ,Θ) for the situations where the beamforming system is in its most aggressive setting (for the example above, this situation is described by Θ={α.sub.k,m=1}. b) Compute the resulting estimated speech intelligibility I=ƒ(SNR(k,m,Φ,Θ)).”).
Claim 20 is a non-transitory digital storage medium Claim with limitations similar to the limitations of Claim 1 and is rejected under similar rationale. Additionally, the non-transitory digital storage medium of Claim 20 is taught by Jensen (col. 16 lns. 7-15: “In some hearing devices, an amplifier and/or compressor may constitute the signal processing circuit. The signal processing circuit typically comprises one or more (integrated or separate) memory elements for executing programs and/or for storing parameters used (or potentially used) in the processing and/or for storing information relevant for the function of the hearing device and/or for storing information (e.g. processed information, e.g. provided by the signal processing circuit),”).
Claim 21 is an apparatus for processing an initial audio signal claim with limitations similar to the limitations of Claim 1 and is rejected under similar rationale. Additionally, Jensen describes the apparatus for processing an initial audio signal, the apparatus comprising
an interface (col. 29 lns. 47-49: “FIG. 2 shows an embodiment of a hearing aid according to the present disclosure comprising a multitude of input transducers”);
a first signal modifier (col. 30 lns. 6-15: “The signal processor (HAPU) is configured to execute one or more processing algorithms. The signal processor (HAPU) comprises a beamformer filtering unit (BF) and is configured to execute a beamformer algorithm. The beamformer filtering unit (BF) receives the multitude of electric input signals y.sub.r, r=1, . . . , M from the input unit (IU), or processed versions thereof, and is configured to provide a spatially filtered, beamformed, signal y.sub.BF. The beamformer algorithm and thus the beamformed signal, is controlled by beamformer parameter settings Θ.”);
an evaluator (col. 30 lns. 23-37: “The first parameter setting Θ1, and/or the beamformed signal y.sub.BF(Θ1) based thereon, is/are fed to the control unit (CONT) together with at least one (here all) of the electric input signals y.sub.r, r=1, . . . , M. An estimate of the intelligibility I(y.sub.BF(Θ)) of the beamformed signal y.sub.BF(Θ) based on the first parameter setting Θ1 (and the user's hearing profile, e.g. reflecting an impairment, Φ) is provided by the speech intelligibility estimator (ESI, cf. FIG. 1A) and fed to the adjustment unit (ADJ, cf. FIG. 1A) for (in dependence on predefined criteria, and if possible, cf. FIG. 1B and description thereof) adjusting (optimizing) the parameter setting Θ to provide a second parameter setting Θ′ that provides the desired speech intelligibility I.sub.des of the processed signal y.sub.res presented to the user.”); and
a selector (col. 30 lns. 37-50: “The controller, e.g. the adjustment unit (ADJ, cf. FIG. 1A), receives as inputs a) the multitude of electric input signals y.sub.r, r=1, . . . , M, b) the estimated speech intelligibility I(y.sub.r) of at least one of the multitude of electric input signals y.sub.r, c) the first parameter setting Θ1, and/or the beamformed signal y.sub.BF(Θ1) based thereon, d) the desired speech intelligibility I.sub.des, and e) the estimated speech intelligibility I(y.sub.BF(Θ1)) of the beamformed signal y.sub.BF(Θ1) based on the first parameter setting Θ1. Based on these inputs (a, b, c, d), the controller provides a second parameter setting Θ′ that is fed to the beamformer filtering unit (BF) and applied to the electric input signals y.sub.r, r=1, . . . , M, to provide the optimized beamformed signal y.sub.BF(Θ′) based thereon (under the conditions discussed above).”).
Jensen doesn’t describe but Usher describes a second signal modifier (FIG. 5, ¶ [0058]-[0065], and claim 1 describe a first dynamic range compressor 514 and a second dynamic range compressor 516).
See the rejection of claim 1 for rationale to modify Jensen based on the teachings of Usher.
Regarding claims 22 and 25, Jensen doesn’t describe but Usher describes the apparatus according to claim 21, and the method according to claim 1, wherein the first signal modifier and the second signal modifier processing said received initial audio signal or the same received initial audio signal (FIG. 5, ¶ [0058]-[0064], and claim 1: “[0058] FIG. 5 shows a detailed exemplary method to generate a DRCF curve to optimize speech intelligibility, and comprises the steps of:
[0059] 1. 502 Receiving a selected audio signal to the earphone DSP. The audio signal is reproduced from a digital storage file, and may be a speech or music audio signal.
[0060] 2. 504 Applying a gain to the received audio signal to generate a modified input audio signal.
[0061] 3. 506 Generating a first dynamic range compression parameter set A, where the parameters comprise a compression ratio value, an expansion ratio value, threshold value, and gate value 508.
[0062] 4. 510 Generating a second dynamic range compression parameter set B, where the parameters also comprise a compression ratio value, an expansion ratio value, threshold value, and gate value 512.
[0063] 5. The modified input signal is processed with a first dynamic range compressor using the DRC parameter set A 514 to produce an output signal A.
[0064] 6. The modified input signal is processed with a [second] dynamic range compressor using the DRC parameter set B 516 to produce an output signal B.).
See the rejection of claim 1 for rationale to modify Jensen based on the teachings of Usher.
Regarding claims 23 and 26, Jensen doesn’t describe but Usher describes the apparatus according to claim 21, and the method according to claim 1, wherein the first signal modifier and the second signal modifier processing said received initial audio signal in parallel (FIG. 5, ¶ [0058]-[0065], and claim 1: “[0058] FIG. 5 shows a detailed exemplary method to generate a DRCF curve to optimize speech intelligibility, and comprises the steps of:
[0059] 1. 502 Receiving a selected audio signal to the earphone DSP. The audio signal is reproduced from a digital storage file, and may be a speech or music audio signal.
[0060] 2. 504 Applying a gain to the received audio signal to generate a modified input audio signal.
[0061] 3. 506 Generating a first dynamic range compression parameter set A, where the parameters comprise a compression ratio value, an expansion ratio value, threshold value, and gate value 508.
[0062] 4. 510 Generating a second dynamic range compression parameter set B, where the parameters also comprise a compression ratio value, an expansion ratio value, threshold value, and gate value 512.
[0063] 5. The modified input signal is processed with a first dynamic range compressor using the DRC parameter set A 514 to produce an output signal A.
[0064] 6. The modified input signal is processed with a [second] dynamic range compressor using the DRC parameter set B 516 to produce an output signal B.
[0065] 7. A preference test is conducted 518 by the user with a user selection interface 520. The preference test can be in the form of a standard paired comparison AB test, where two audio signals are presented A and B, A and O, or B and O, and the user determines which signal they prefer. In one exemplary embodiment, the user is asked to determine which signal, A or B, sounds the clearest in terms of speech intelligibility. Using this methodology, an optimum DRCF can be determined that optimizes speech intelligibility.”).
See the rejection of claim 1 for rationale to modify Jensen based on the teachings of Usher.
Regarding claims 24 and 27, Jensen doesn’t describe but Usher describes the apparatus according to claim 21, and the method according claim 1, wherein the first signal modifier and the second signal modifier comprise different signal-processing structures configured to modify said received initial audio signal in different ways (FIG. 5, ¶ [0058]-[0065], and claim 1: “[0058] FIG. 5 shows a detailed exemplary method to generate a DRCF curve to optimize speech intelligibility, and comprises the steps of:
[0059] 1. 502 Receiving a selected audio signal to the earphone DSP. The audio signal is reproduced from a digital storage file, and may be a speech or music audio signal.
[0060] 2. 504 Applying a gain to the received audio signal to generate a modified input audio signal.
[0061] 3. 506 Generating a first dynamic range compression parameter set A, where the parameters comprise a compression ratio value, an expansion ratio value, threshold value, and gate value 508.
[0062] 4. 510 Generating a second dynamic range compression parameter set B, where the parameters also comprise a compression ratio value, an expansion ratio value, threshold value, and gate value 512.
[0063] 5. The modified input signal is processed with a first dynamic range compressor using the DRC parameter set A 514 to produce an output signal A.
[0064] 6. The modified input signal is processed with a [second] dynamic range compressor using the DRC parameter set B 516 to produce an output signal B.
[0065] 7. A preference test is conducted 518 by the user with a user selection interface 520. The preference test can be in the form of a standard paired comparison AB test, where two audio signals are presented A and B, A and O, or B and O, and the user determines which signal they prefer. In one exemplary embodiment, the user is asked to determine which signal, A or B, sounds the clearest in terms of speech intelligibility. Using this methodology, an optimum DRCF can be determined that optimizes speech intelligibility.”).
See the rejection of claim 1 for rationale to modify Jensen based on the teachings of Usher.
Regarding claims 28 and 29, Jensen describes the apparatus according to claim 21, and the method according to claim 1, wherein for a first timeframe of the initial audio signal the first [modified audio signal] is selected; and wherein for a second timeframe of the initial audio signal the second [modified audio signal] is selected (see col. 22 lns. 12-21: “A speech intelligibility measure of one or more processed or un-processed signals is determined at successive points in time t. As indicated in FIG. 1B by unit or process step ‘t=t+1’. The successive points in time may e.g. be every successive time frame (defined by time frame index m) of the respective signals. Alternatively, successive points in time may indicate a lower rate, e.g. every 10.sup.th time frame.
The controller is configured to control the processor to provide that the resulting signal y.sub.res at a current point in time t is equal to one of the electric input signals y”.
Further, see col. 22 ln. 60 – col. 23 ln. 5: “In case the statement ‘I(y.sub.p(Θ1,t))≤I.sub.des?’ is false (branch ‘No’), i.e. if the speech intelligibility measure I of the processed signal y.sub.p(Θ1,t) is larger than the desired value I.sub.des, the controller is further configured to determine a second parameter setting Θ′ of the processing algorithm under the constraint that the second processed signal y.sub.p(Θ′) exhibits the desired value I.sub.des of the speech intelligibility measure, and to control the processor to provide that the resulting signal y.sub.res at the current point in time t is equal to the second, optimized, processed signal y.sub.p(Θ′) (cf. respective units or process steps, ‘Find Θ′ providing I(y.sub.p(Θ,t)=I.sub.des. Set y.sub.res=y.sub.p(Θ′,t)’, and advance time to the next time index ‘t=t+1’).”).
Jensen doesn’t describe that the first modified signal comes from the first modifier and the second modified signal comes from the second modifier.
However, Usher describes a system and method wherein the first modified signal comes from the first modifier and the second modified signal comes from the second modifier (FIG. 5, ¶ [0058]-[0064], and claim 1: “[0058] FIG. 5 shows a detailed exemplary method to generate a DRCF curve to optimize speech intelligibility, and comprises the steps of:
[0059] 1. 502 Receiving a selected audio signal to the earphone DSP. The audio signal is reproduced from a digital storage file, and may be a speech or music audio signal.
[0060] 2. 504 Applying a gain to the received audio signal to generate a modified input audio signal.
[0061] 3. 506 Generating a first dynamic range compression parameter set A, where the parameters comprise a compression ratio value, an expansion ratio value, threshold value, and gate value 508.
[0062] 4. 510 Generating a second dynamic range compression parameter set B, where the parameters also comprise a compression ratio value, an expansion ratio value, threshold value, and gate value 512.
[0063] 5. The modified input signal is processed with a first dynamic range compressor using the DRC parameter set A 514 to produce an output signal A.
[0064] 6. The modified input signal is processed with a [second] dynamic range compressor using the DRC parameter set B 516 to produce an output signal B.).
See the rejection of claim 1 for rationale to modify Jensen based on the teachings of Usher.
Claim 3 is rejected under 35 U.S.C. 103 as being unpatentable over Jensen in view of Usher and further in view of Bertelsen et al. (US 10,165,373), hereinafter “Bertelsen”.
Regarding Claim 3, Jensen teaches the method according to claim 1, evaluation criterions, such that respective first evaluation values describing a degree of fulfilment of for at least two independent evaluation criterions for the first modified audio signal and respective second evaluation values describing a degree of fulfilment of for the at least two independent evaluation criterions for the second modified audio signal are determined (col. 29 lns. 33-46: “3) a) Identify the (potentially multiple) parameter settings Θ which achieve I=I.sub.desired, and which process the incoming signal the least, e.g., the beamformer settings which reduce the total noise power at the output of the beamformer the least, or the beamformer settings which lead to maximum total signal loudness, the beamformer settings that best maintain the direction and value of the spatial minima of the beam pattern, etc. (several such secondary requirements may be envisioned). This may, e.g., be done by introducing the Karush-Kuhn Tucker conditions (cf. p 243 in [4]) and identifying the beamformer parameter settings, which satisfy these conditions, see [2, 3] for examples.”),
Jensen in view of Usher doesn’t describe a system or method wherein then the selection is performed based on weighted first and second evaluation values.
However, Bertelsen describes a system and method wherein then the selection is performed based on weighted first and second evaluation values (col. 4 lns. 17-26: “In an embodiment, the resulting adaptation parameter β.sub.mix is determined as a linear combination of the adaptation parameters β.sub.opt and β.sub.fix according to the expression
β.sub.mix=αβ.sub.opt+(1−α)β.sub.fix,
where the weighting parameter α is a real number between 0 and 1. This has the advantage of providing a computationally simple solution. In an embodiment,
β.sub.mix=w.sub.1β.sub.opt+w.sub.2β.sub.fix,
where w.sub.1 and w.sub.2 are complex or real weighting factors.”).
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to include in Jensen in view of Usher a system and method wherein then the selection is performed based on weighted first and second evaluation values, as taught by Bertelsen, in order to enable a user to indicate which evaluation values are more important and less important when determining optimal beamforming settings, which enables the system to be tuned such that it can achieve better speech intelligibility.
Claim 7 is rejected under 35 U.S.C. 103 as being unpatentable over Jensen in view of Usher in view of Bertelsen, and further in view of Kristjansson et al. (US 10,937,441), hereinafter “Kristjansson”.
Regarding Claim 7, Jensen describes the method according to claim 3, wherein outputting the initial audio signal is performed instead of outputting the first or second modified audio signal, when the respective first or second [speech intelligibility] value is below a threshold, below which threshold a respective first or second modified audio signal is indicated as not [at the desired level of speech intelligibility] to the initial audio signal (col. 36 lns. 57-63: “B4. If I.sub.max-SNR(=I(y.sub.p(Θ1))≤I.sub.des (path ‘Yes’ in FIG. 6), where I.sub.des is the desired value of the speech intelligibility measure I, set y.sub.res=y.sub.sel, where y.sub.sel is a selectable signal e.g. equal to an unprocessed input signal y.sub.ref [emphasis added] or to the first processed signal y.sub.res=y.sub.p(Θ1), or to a combination of one of them with an information signal y.sub.inf indicating that the intelligibility situation is difficult.”).
Jensen in view of Usher in view of Bertelsen doesn’t describe a system or method wherein outputting the initial audio signal is performed instead of outputting the first or second modified audio signal, when the respective first or second perceptual similarity value is below a threshold, below which threshold a respective first or second modified audio signal is indicated as not sufficiently similar to the initial audio signal.
However, Kristjansson describes a system and method wherein in certain situations noise processing of a signal attenuates portions of the local speech and degrades the audio data. The system disables the noise processing in those situations, in order to improve the quality of the speech (see col. 17 lns. 21-37).
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to include in Jensen in view of Usher in view of Bertelsen a system and method wherein outputting the initial audio signal is performed instead of outputting the first or second modified audio signal, when the respective first or second perceptual similarity value is below a threshold, below which threshold a respective first or second modified audio signal is indicated as not sufficiently similar to the initial audio signal, as suggested by Kristjansson, in order to output the original audio signal when beamforming results in a processed signal that is lower quality than the original, which improves the perceived performance of the system, since it doesn’t output a signal that is lower quality than the original incoming signal.
Claim 9 is rejected under 35 U.S.C. 103 as being unpatentable over Jensen in view of Usher in view of Norris et al. (US 9,226,090), hereinafter “Norris”.
Regarding Claim 9, Jensen in view of Usher doesn’t teach, but Norris teaches the method according to claim 1, wherein the first and/or second modified audio signal comprises the target portion moved into the foreground and the side portion moved into the background and/or a speech portion as the target portion moved into the foreground and an ambient noise portion as the side portion moved into the background (col. 36 lns. 17-22 “Further yet, this user can arrange the SLPs of the window/process in any order and/or move any of them to the foreground (such as bringing the associated SLP toward him to make the sound louder), moving any of them to the background (such as bringing the SLP away from him to make the sound softer)”).
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to include in Jensen in view of Usher a system and method wherein the first and/or second modified audio signal comprises the target portion moved into the foreground and the side portion moved into the background and/or a speech portion as the target portion moved into the foreground and an ambient noise portion as the side portion moved into the background, as suggested by Norris, in order to further improve the beamforming by amplifying speech signals, by moving them into the foreground, and reducing noise signals, by moving them into the background, which further improves speech intelligibility without losing the context of the background noise.
Claim 10 is rejected under 35 U.S.C. 103 as being unpatentable over Jensen in view of Usher in view of Francombe et al. (US 20150264507), hereinafter “Francombe”.
Regarding Claim 10, Jensen in view of Usher doesn’t teach, but Francombe discloses:
the method according to claim 1, wherein comparing comprises extracting the first and/or second evaluation value by use of a perceptual model, PEAQ model, POLQA model, and/or a PEMO-Q model (¶ [0069]— “…After this, four quantities are produced by calculating the perceptual similarity measure PSM (of Huber and Kollmeier's PEMO-Q) of four pairs of signals…” This illustrates the extraction of evaluation values using perceptual models, including the PEMO-Q model.
It would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to include in Jensen in view of Usher a system and method wherein comparing comprises extracting the first and/or second evaluation value by use of a perceptual model, PEAQ model, POLQA model, and/or a PEMO-Q model, as suggested by Francombe, in order to evaluate the perceived quality of the audio signal through perceptual similarity measure (PSM) , focusing on quality assessment based on unique audio characteristics and spatial perceptions [Francombe: 0278].
Conclusion
The prior art made of record and not relied upon is considered pertinent to applicant's disclosure: See additional references cited on PTO-892.
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/DANIEL C WASHBURN/Supervisory Patent Examiner, Art Unit 2657
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