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 filed September 10, 2025 have been fully considered but they are not persuasive.
Applicants submit that in the frequency-domain equalizer disclosed in Faruk, the input signal with 2xoversampling is separated to a 1x sampling signal containing only even-numbered samples and a 1x sampling signal containing only odd-numbered samples, and, by expanding the MIMO filter (operating at 1x sampling), the equalizer with 2x oversampling input and 1x sampling output is implemented. The approach employed in Faruk is an approach that focuses on refining the filter input signals and the filter itself.
The grounds of rejection rely on allegations that Faruk teaches wherein an input signal of oversampling of the first predetermined multiple is a signal of M/L times oversampling, where M and L are integers satisfying 1<M/L<2 (the received sequence is sampled by free-running analog-to-digital converters (ADCs) operated at the rate twice the symbol rate (i.e., twofold oversampling), Page 2, Paragraph 2).
Re claim 1, Examiner submits that Faruk teaches of the input signal being an input signal of oversampling of a first predetermined multiple (2xoversampling, Page 3). Therefore, Faruk, Arikawa and Ke teach all the limitations of claim 1. Examiner disagrees with the statement that Faruk teaches of 2xoversampling being separated to a 1x sampling signal containing only even-numbered samples and a 1x sampling signal containing only odd-numbered samples, and, by expanding the MIMO filter (operating at 1x sampling), since Faruk mentions that the input sequences ux(n) and uy(n) are 2xoversampled and the filter output is downsampled by 2. Re claim 3, Faruk teaches of 2xoversampling (M/L=2) that satisfies the condition
1
<
M
/
L
≤
2
. Therefore, the combination of Faruk, Arikawa and Ke teach all the limitations of claim 3.
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.
The factual inquiries for establishing a background for determining obviousness under 35 U.S.C. 103 are summarized as follows:
1. Determining the scope and contents of the prior art.
2. Ascertaining the differences between the prior art and the claims at issue.
3. Resolving the level of ordinary skill in the pertinent art.
4. Considering objective evidence present in the application indicating obviousness or nonobviousness.
Claims 1 – 7, 10 – 17 and 19 are rejected under 35 U.S.C. 103 as being unpatentable over Faruk et al (“Adaptive frequency-domain equalization in digital coherent optical receivers”, The University of Tokyo, Vol. 19, No. 13, OPTICS EXPRESS, 20 June 2011) in view of Arikawa et al (“Adaptive equalization of transmitter and receiver IQ skew by multi-layer linear and widely linear filters with deep unfolding”, Vol. 28, No. 16, Optics Express, 3 August 2020) and further in view of Ke et al (US 2021/0328680).
Re claim 1, Faruk teaches of an equalization signal processing circuit comprising: circuitry configured to: convert (FFT, Fig.1) an input signal of oversampling of a first predetermined multiple into a signal (the received sequence is sampled by free-running analog-to-digital converters (ADCs) operated at the rate twice the symbol rate (i.e., twofold oversampling), Page 2, Paragraph 2) in a frequency domain (FFT output, Fig.1); input an input signal being converted into the frequency domain to a frequency domain filter (frequency domain equalizer, Pages 3 – 4, as shown in Fig.1), and multiply the input signal being converted into the frequency domain by a coefficient for each frequency by the frequency domain filter (frequency domain filter-tap coefficients or weights, Page 3 – 4 and Fig.1); convert (the filter output is down-sampled by a factor 2, Page 3) the signal being multiplied by a coefficient for each frequency by the frequency domain filter into a signal of oversampling of a second predetermined multiple (1x oversampling, Page 3); convert the signal being converted into the signal of oversampling of the second predetermined multiple into a signal in a time domain (IFFT, Fig.1); calculate a gradient (equations 15 – 16) of a loss function for the coefficient (CMA, equation 14, Page 4 and Fig.1); and update the coefficient of the frequency domain filter, based on the calculated gradient of the loss function for the coefficient (equation 17 and Fig.1). However, Faruk does not specifically teach of calculating the gradient of the loss function for the coefficient by using an error back propagation method using, as the loss function, magnitude of a difference between a signal of oversampling of the second predetermined multiple being converted into a signal in the time domain, and a predetermined value determined by oversampling of the second predetermined multiple. Faruk does not specifically teach of the equalization signal processing circuit comprising: at least one memory storing instructions; and at least one processor configured to perform the functions of the equalization signal processing circuit.
Arikawa teaches of calculating a gradient of a loss function for the coefficient by using an error back propagation method (see Fig.2) using, as the loss function (CMA, Page 7), magnitude of a difference (equation 17) between a signal being filtered by the frequency domain filter, (Fig.2 and magnitude of yi(k), Page 7 and equation 17), and a predetermined value (amplitude criterion r, equation 17); and update the coefficient of the frequency domain filter, based on the calculated gradient of the loss function for the coefficient (filter coefficient update algorithm by SGD with back propagation, Pages 3 and 8).
Ke teaches of equalization signal processing circuit comprising: at least one memory storing instructions; and at least one processor configured to execute the instructions to perform the functions of the equalization signal processing circuit (Paragraphs 0062 – 0063, Fig.5).
It would have been obvious to one having ordinary skill in the art before the effective filing date of the claimed invention to have calculated the gradient of the loss function for the coefficient by using an error back propagation method as taught by Arikawa to control the filter coefficients so that the magnitude of deviation of the last outputs from the desired state is stochastically minimized. It would have been obvious to one having ordinary skill in the art before the effective filing date of the claimed invention to have the equalization signal processing circuit comprise a memory storing instructions and at least one processor configured to execute the instructions so as to efficiently perform the functions of the equalization signal processing circuit.
Re claim 2, Faruk teaches of wherein the second predetermined multiple is one time (downsampling the 2x oversampled signal would output a 1x oversampled signal, Page 3).
Re claim 3, Faruk teaches of wherein an input signal of oversampling of the first predetermined multiple is a signal of M/L times oversampling, where M and L are integers satisfying
1
<
M
/
L
≤
2
(the received sequence is sampled by free-running analog-to-digital converters (ADCs) operated at the rate twice the symbol rate (i.e., twofold oversampling), Page 2, Paragraph 2).
Re claim 4, Faruk teaches of wherein the input signal is a signal acquired by coherently receiving a signal being transmitted via a transmission path by a receiver (digital coherent optical receiver, Abstract and 1. Introduction, Page 2).
Re claim 5, Faruk teaches of wherein the input signal is a polarization multiplexed signal, and the frequency domain filter compensates for a change in a polarization state in the transmission path (proposed adaptive FDE for polarization-multiplexed transmission systems is shown in Fig. 1, Pages 4 – 5, Figures 1 – 2).
Re claim 6, Faruk, Arikawa and Ke teach all the limitations of claim 1 as well as Arikawa teaches of wherein the frequency domain filter is a multi-input multi-output (MIMO) filter (MIMO filter, Pages 23479, 23481 – 23482 and 23485).
It would have been obvious to one having ordinary skill in the art before the effective filing date of the claimed invention to have the frequency domain filter be a multi-input multi-output (MIMO) filter to compensate for IQ skew and CD simultaneously in one filter.
Re claim 7, Faruk teaches of wherein the at least one processor is configured to execute the instructions to update the coefficient by a stochastic gradient descent method (gradient decent algorithm, Page 4), based on the gradient of the loss function for the coefficient (Fig.1).
Re claim 10, Faruk teaches of a receiver (receiver part of Fig.2) comprising: a detector configured to coherently receive a signal being transmitted from a transmitter via a transmission path (as shown in Fig.2); and the equalization signal processing circuit according to claim 1, the equalization signal processing circuit according to wherein the equalization signal processing circuit performs equalization signal processing on the coherently received input signal of oversampling of a first predetermined multiple (ADC, Fig.2 and Page 2, Paragraph 2).
Re claim 11, Faruk teaches of wherein the second predetermined multiple is one time (see claim 2).
Re claim 12, Faruk teaches of wherein an input signal of oversampling of the first predetermined multiple is a signal of M/L times oversampling, where M and L are integers satisfying
1
<
M
/
L
≤
2
(see claim 3).
Re claim 13, Faruk, Arikawa and Ke teach all the limitations of claim 10 as well as Arikawa teaches of wherein the frequency domain filter is a multi-input multi-output (MIMO) filter (MIMO filter, Pages 23479, 23481 – 23482 and 23485).
It would have been obvious to one having ordinary skill in the art before the effective filing date of the claimed invention to have the frequency domain filter be a multi-input multi-output (MIMO) filter to compensate for IQ skew and CD simultaneously in one filter.
Re claim 14, Faruk teaches of a communication system comprising: a transmitter configured to transmit a signal via a transmission path (Transmitter of Fig.2); and the receiver according to claim 10.
Re claim 15, Faruk teaches of wherein the second predetermined multiple is one time (see claim 2).
Re claim 16, Faruk teaches of wherein an input signal of oversampling of the first predetermined multiple is a signal of M/L times oversampling, where M and L are integers satisfying
1
<
M
/
L
≤
2
(see claim 3).
Re claim 17, Faruk, Arikawa and Ke teach all the limitations of claim 14 as well as Arikawa teaches of wherein the frequency domain filter is a multi-input multi-output (MIMO) filter (MIMO filter, Pages 23479, 23481 – 23482 and 23485).
It would have been obvious to one having ordinary skill in the art before the effective filing date of the claimed invention to have the frequency domain filter be a multi-input multi-output (MIMO) filter to compensate for IQ skew and CD simultaneously in one filter.
Re claim 19, Faruk teaches of an equalization signal processing circuit comprising: circuitry configured to: convert (FFT, Fig.1) an input signal of oversampling of a first predetermined multiple into a signal (the received sequence is sampled by free-running analog-to-digital converters (ADCs) operated at the rate twice the symbol rate (i.e., twofold oversampling), Page 2, Paragraph 2) in a frequency domain (FFT output, Fig.1); input an input signal being converted into the frequency domain to a frequency domain filter (frequency domain equalizer, Pages 3 – 4, as shown in Fig.1), and multiply the input signal being converted into the frequency domain by a coefficient for each frequency by the frequency domain filter (frequency domain filter-tap coefficients or weights, Page 3 – 4 and Fig.1); convert (the filter output is down-sampled by a factor 2, Page 3) a signal being multiplied by a coefficient by the frequency domain filter into a signal of oversampling of a second predetermined multiple (1x oversampling, Page 3); convert a signal being converted into a signal of oversampling of the second predetermined multiple into a signal in a time domain (IFFT, Fig.1); calculate a gradient (equations 15 – 16) of a loss function for the coefficient (CMA, equation 14, Page 4 and Fig.1); and update the coefficient of the frequency domain filter, based on the calculated gradient of the loss function for the coefficient (equation 17 and Fig.1). However, Faruk does not specifically teach of calculating the gradient of the loss function for the coefficient by using an error back propagation method using, as the loss function, magnitude of a difference between a signal of oversampling of the second predetermined multiple being converted into a signal in the time domain, and a predetermined value determined by oversampling of the second predetermined multiple. Faruk does not specifically teach of a non-transitory computer readable medium storing a program for causing a processor to execute the processing.
Arikawa teaches of calculating a gradient of a loss function for the coefficient by using an error back propagation method (see Fig.2) using, as the loss function (CMA, Page 7), magnitude of a difference (equation 17) between a signal being filtered by the frequency domain filter, (Fig.2 and magnitude of yi(k), Page 7 and equation 17), and a predetermined value (amplitude criterion r, equation 17); and update the coefficient of the frequency domain filter, based on the calculated gradient of the loss function for the coefficient (filter coefficient update algorithm by SGD with back propagation, Pages 3 and 8).
Ke teaches of a non-transitory computer readable medium storing a program for causing a processor to execute equalization signal processing (Paragraphs 0062 – 0063, Fig.5).
It would have been obvious to one having ordinary skill in the art before the effective filing date of the claimed invention to have calculated the gradient of the loss function for the coefficient by using an error back propagation method as taught by Arikawa to control the filter coefficients so that the magnitude of deviation of the last outputs from the desired state is stochastically minimized. It would have been obvious to one having ordinary skill in the art before the effective filing date of the claimed invention to have a non-transitory computer-readable medium storing computer-executable instructions for the receiver for its storage capacity, portability, where data cannot be changed and for preventing accidental erasure of programs or files.
Claim 8 is rejected under 35 U.S.C. 103 as being unpatentable over Faruk, Arikawa and Ke in view of Abe (US 2016/0308579).
Re claim 8, Faruk, Arikawa and Ke teach all the limitations of claim 14 as well as Faruk teaches of wherein the at least one processor is further configured to execute the instructions to perform processing in a time domain (IFFT, Fig.1) on a signal being converted into the signal of one-time oversampling (downsampling the 2x oversampled signal would output a 1x oversampled signal, Page 3) and to calculate, as a loss function, magnitude of a difference between an output signal and the predetermined value (as taught by Arikawa in claim 1). However, Faruk, Arikawa and Ke does not specifically teach of performing, using a time domain filter, filter processing in a time domain on the signal being converted into the signal of one-time oversampling, and the at least one processor is configured to execute the instructions to calculate, as a loss function, magnitude of a difference between an output signal of the time domain filter and the predetermined value.
Abe teaches of performing, using a time domain filter, filter processing in a time domain on a filtered signal (time domain equalization, #300B, Fig.4), and to calculate the coefficients of the frequency domain filter (#440, Fig.4) based on an output signal of the time domain filter (as shown in Fig.4).
It would have been obvious to one having ordinary skill in the art before the effective filing date of the claimed invention to have performed filter processing in a time domain on a filtered signal to compensate for signal distortion in the time domain.
Claim 9 is rejected under 35 U.S.C. 103 as being unpatentable over Faruk, Arikawa and Ke in view of Arikawa et al (“Transmission of a 127 Gb/s PM-QPSK Signal over a 3350 km SMF-Only Line with Chromatic Dispersion Compensation using Real-Time DSP”, Vol 4, No 11, November 2012) (Arikawa(2)).
Re claim 9, Faruk, Arikawa and Ke teach all the limitations of claim 1 as well as Faruk teaches of wherein the at least one processor is further configured to execute the instructions to: convert the input signal of oversampling of a first predetermined multiple from a serial signal into a block signal while providing a constant overlap between blocks (overlap-save, Page 5), convert the converted block signal into a signal in the frequency domain (equation 11), and input the converted signal to the frequency domain filter (Fig.1), and convert a block signal to be converted into a signal in the time domain (IFFT, Fig.1) into a serial signal (P/S, Fig.1). Faruk, Arikawa and Ke does not specifically teach of converting the block signal to be converted into a signal in the time domain into a serial signal, while leaving a domain not being affected by assumption of periodicity included in a signal being converted into a signal in the time domain and removing a domain that is possibly affected by assumption of periodicity included in a signal being converted into a signal in the time domain.
Arikawa(2) teaches of converting the input signal from a serial signal into a block signal while providing a constant overlap between blocks (overlap-save, Col 1, Page 2), convert the converted block signal into a signal in the frequency domain (DFT, Fig.1), and input the converted signal to the frequency domain filter (FDE, Fig.2), and convert a block signal to be converted into a signal in the time domain (IDFT, Fig.2), while leaving a domain not being affected by assumption of periodicity included in a signal being converted into a signal in the time domain and removing a domain that is possibly affected by assumption of periodicity included in a signal being converted into a signal in the time domain (Col 1, Page 2 and Fig.1).
It would have been obvious to one having ordinary skill in the art before the effective filing date of the claimed invention to have removed a domain that is possibly affected by assumption of periodicity included in a signal being converted into a signal in the time domain so as to eliminate residual distortion.
Claim 18 is rejected under 35 U.S.C. 103 as being unpatentable over Faruk in view of Arikawa.
Re claim 18, Faruk teaches of an equalization signal processing circuit comprising: circuitry configured to: convert (FFT, Fig.1) an input signal of oversampling of a first predetermined multiple into a signal (the received sequence is sampled by free-running analog-to-digital converters (ADCs) operated at the rate twice the symbol rate (i.e., twofold oversampling), Page 2, Paragraph 2) in a frequency domain (FFT output, Fig.1); input an input signal being converted into the frequency domain to a frequency domain filter (frequency domain equalizer, Pages 3 – 4, as shown in Fig.1), and multiply the input signal being converted into the frequency domain by a coefficient for each frequency by the frequency domain filter (frequency domain filter-tap coefficients or weights, Page 3 – 4 and Fig.1); convert (the filter output is down-sampled by a factor 2, Page 3) a signal being multiplied by a coefficient by the frequency domain filter into a signal of oversampling of a second predetermined multiple (1x oversampling, Page 3); convert a signal being converted into a signal of oversampling of the second predetermined multiple into a signal in a time domain (IFFT, Fig.1); calculate a gradient (equations 15 – 16) of a loss function for the coefficient (CMA, equation 14, Page 4 and Fig.1); and update the coefficient of the frequency domain filter, based on the calculated gradient of the loss function for the coefficient (equation 17 and Fig.1). However, Faruk does not specifically teach of calculating the gradient of the loss function for the coefficient by using an error back propagation method using, as the loss function, magnitude of a difference between a signal of oversampling of the second predetermined multiple being converted into a signal in the time domain, and a predetermined value determined by oversampling of the second predetermined multiple.
Arikawa teaches of calculating a gradient of a loss function for the coefficient by using an error back propagation method (see Fig.2) using, as the loss function (CMA, Page 7), magnitude of a difference (equation 17) between a signal being filtered by the frequency domain filter, (Fig.2 and magnitude of yi(k), Page 7 and equation 17), and a predetermined value (amplitude criterion r, equation 17); and update the coefficient of the frequency domain filter, based on the calculated gradient of the loss function for the coefficient (filter coefficient update algorithm by SGD with back propagation, Pages 3 and 8).
It would have been obvious to one having ordinary skill in the art before the effective filing date of the claimed invention to have calculated the gradient of the loss function for the coefficient by using an error back propagation method as taught by Arikawa to control the filter coefficients so that the magnitude of deviation of the last outputs from the desired state is stochastically minimized.
Conclusion
THIS ACTION IS MADE FINAL. Applicant is reminded of the extension of time policy as set forth in 37 CFR 1.136(a).
A shortened statutory period for reply to this final action is set to expire THREE MONTHS from the mailing date of this action. In the event a first reply is filed within TWO MONTHS of the mailing date of this final action and the advisory action is not mailed until after the end of the THREE-MONTH shortened statutory period, then the shortened statutory period will expire on the date the advisory action is mailed, and any nonprovisional extension fee (37 CFR 1.17(a)) pursuant to 37 CFR 1.136(a) will be calculated from the mailing date of the advisory action. In no event, however, will the statutory period for reply expire later than SIX MONTHS from the mailing date of this final action.
Any inquiry concerning this communication or earlier communications from the examiner should be directed to ARISTOCRATIS FOTAKIS whose telephone number is (571)270-1206. The examiner can normally be reached M-F 8:30am-5:00pm.
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, Sam K Ahn can be reached at (571) 272-3044. 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.
/ARISTOCRATIS FOTAKIS/
Primary Examiner, Art Unit 2633
Prosecution Insights