DETAILED ACTION
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 1-16 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. Regarding independent claim 1, line 5 recites “N time” and, lines 7,14,16 and 17 recites “M+1”. The claim languages of M delay elements, M+1 filter coefficients and M+1 error estimators are not defined what determines M and how M relates to N (the impulse response of the communication channel). Claim 1, line 11 recites “k-th time” is not defined what determines k. Therefore, the scope of the claim is not clear. Further, structural relation is unclear. The relation between the finite impulse response filter, the interval selector, the delay elements, and the error estimators is not clearly defined the relationship between these structures which makes the scope of the claim indefinite. Independent claim 16 is reject due to a similar issue shown above in claim 1.
Claim 2-15 are rejected because they depend on rejected claim 1.
Claim Rejections - 35 USC § 102
In the event the determination of the status of the application as subject to AIA 35 U.S.C. 102 and 103 (or as subject to pre-AIA 35 U.S.C. 102 and 103) is incorrect, any correction of the statutory basis (i.e., changing from AIA to pre-AIA ) for the rejection will not be considered a new ground of rejection if the prior art relied upon, and the rationale supporting the rejection, would be the same under either status.
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 –
(a)(1) the claimed invention was patented, described in a printed publication, or in public use, on sale, or otherwise available to the public before the effective filing date of the claimed invention.
Claims 1-16 are rejected under 35 U.S.C. 102 (a) (1) as being anticipated by U.S Pub. No. 2009/0060167 A1 to Deng et al. (hereinafter “Deng”).
Regarding claim 1, Deng teaches an apparatus for determining filter coefficients for an echo compensation filter for reduction of an echo signal in a communication channel upon a basis of a received signal, the received signal being a receivable version of a known reference signal transmittable over the communication channel, the receivable version of the reference signal being affected by the echo signal, an impulse response of the communication channel extending over N time intervals (Abstract, paragraphs [0053]; the output of adder 322 is an error signal, which signal is the difference between the estimated signal and signal S.sub.in. This error signal is used both as feedback for signal estimator 320, and for the output signal S.sub.out, wherein S.sub.out should have a significantly reduced echo signal once full rate adaptive filter 318 has converged. The convergence time can be defined as the interval between the instant a signal is applied to the R.sub.in input 308 of echo canceller 302 with the estimated echo path impulse response (e.g., the filter coefficients) initially set to zero, and the instant the returned echo level in error signal at output 314 reaches a defined level), the apparatus comprising:
a finite impulse response filter comprising a set of M+1 filter coefficients, the finite
impulse response filter being configured to filter values of the reference signal using the M+1filter coefficients to obtain a first filtered reference signal (paragraphs [0020], [0029] and [0051]- [0053]; the pure delay can be defined as a time distance between the beginning of the impulse response and the time location of the impulse response absolute value that is significantly greater than the preceding values (e.g., impulse response absolute values that exceed a threshold). Alternatively, the pure delay can be a time distance between the beginning of the impulse response and the time location of the maximum value of the impulse response, reduced by the length of the preceding impulse segment that contains intermediate values between the maximum values and the average noise level value);
an interval selector configured to select values of the reference signal associated with a k-th time interval of the N time intervals (paragraph [0053]; error signal is used both as feedback for signal estimator 320, and for the output signal S.sub.out, wherein S.sub.out should have a significantly reduced echo signal once full rate adaptive filter 318 has converged. The convergence time can be defined as the interval between the instant a signal is applied to the R.sub.in input 308 of echo canceller 302 with the estimated echo path impulse response (e.g., the filter coefficients) initially set to zero, and the instant the returned echo level in error signal at output 314 reaches a defined level);
M delay elements configured to successively delay the selected values of the reference signal, the M delay elements being arranged after the interval selector (paragraphs [0020] and [0051]- [0053]; a tapped-delay-line finite impulse response (FIR) filter which is adapted by changing or adapting the filter coefficients (FIR filter is part of Box with filter coefficients 118). Filter coefficients 118 can adapt to describe an impulse response of a connected linear system, which can produce an echo signal); and
M+1 error estimators, a first error estimator of the M+1 error estimators being coupled to an output of the interval selector, each of the remaining M error estimators being coupled to an output of a respective delay element, the M+1 error estimators being configured to determine the M+1 filter coefficients such that a measure of a deviation between the first filtered reference signal and the received signal is minimized (Abstract, paragraphs [0030] and [0050]-[0053]; the process calculates error sample e(k) according to the formula e(k)=d(k)-y(k), as depicted at 206. In one embodiment, calculating the error sample can be implemented using an adder, such as adder 128 in FIG. 1, wherein sample y(k) is input into an inverting input of the adder and sample d(k) is input into a noninverting input. The samples output by adder 128, which can be referred to as error signal e(k), represent the difference between estimated signal y(k) and the signal that can contain echo d(k)).
Regarding claim 2, Deng teaches the apparatus of claim 1, wherein the M+1 error estimators are mean squares error estimators (paragraphs [0050]- [ [0052]; the least mean square (LMS) algorithm, the Normalized Least Mean Squares (NLMS) algorithm, the Proportionate Normalized Least Mean Squares (PNLMS) algorithm, the Affine Projection (AP) algorithm, the Recursive Least Squares (RLS) algorithm, or the like. Setting a pure delay for the full rate main adaptive filter configures the filter to adapt filter coefficients in the relevant portion of the echo path impulse response domain (e.g., the filter is set to adapt to a delayed R.sub.in signal near the time of the undesired echo signal), thereby enabling the filter to converge faster and be implemented with a shorter length adaptive filter (e.g., a "sparse" adaptive filter, which is a filter having a length shorter than the actual echo path impulse response length that includes the pure delay portion)).
Regarding claim 3, Deng teaches the apparatus of claim 1, wherein a measure of a deviation between the first filtered reference signal and the received signal is a mean square deviation between values of the first filtered reference signal and the received signal (paragraphs [0050]- [ [0052];the least mean square (LMS) algorithm, the Normalized Least Mean Squares (NLMS) algorithm, the Proportionate Normalized Least Mean Squares (PNLMS) algorithm, the Affine Projection (AP) algorithm, the Recursive Least Squares (RLS) algorithm, or the like. Setting a pure delay for the full rate main adaptive filter configures the filter to adapt filter coefficients in the relevant portion of the echo path impulse response domain (e.g., the filter is set to adapt to a delayed R.sub.in signal near the time of the undesired echo signal), thereby enabling the filter to converge faster and be implemented with a shorter length adaptive filter (e.g., a "sparse" adaptive filter, which is a filter having a length shorter than the actual echo path impulse response length that includes the pure delay portion)).
Regarding claim 4, Deng teaches the apparatus of claim 1, comprising:
a memory configured to store the M+1 filter coefficients for the values of the reference signal selected by the interval selector (paragraphs [0051]- [0053]; error signal is used both as feedback for signal estimator 320, and for the output signal S.sub.out, wherein S.sub.out should have a significantly reduced echo signal once full rate adaptive filter 318 has converged. The convergence time can be defined as the interval between the instant a signal is applied to the R.sub.in input 308 of echo canceller 302 with the estimated echo path impulse response (e.g., the filter coefficients) initially set to zero, and the instant the returned echo level in error signal at output 314 reaches a defined level).
Regarding claim 5, Deng teaches the apparatus of claim 4, wherein the memory is configured to store the M+1filter coefficients for each time interval of the N time intervals applied by the interval selector (paragraphs [0051]- [0053]; error signal is used both as feedback for signal estimator 320, and for the output signal S.sub.out, wherein S.sub.out should have a significantly reduced echo signal once full rate adaptive filter 318 has converged. The convergence time can be defined as the interval between the instant a signal is applied to the R.sub.in input 308 of echo canceller 302 with the estimated echo path impulse response (e.g., the filter coefficients) initially set to zero, and the instant the returned echo level in error signal at output 314 reaches a defined level).
Regarding claim 6, Deng teaches the apparatus of claim 4, wherein the memory is configured to store the M+1 filter coefficients together with the value of the k-th time interval applied by the interval selector (paragraphs [0051]- [0053]; error signal is used both as feedback for signal estimator 320, and for the output signal S.sub.out, wherein S.sub.out should have a significantly reduced echo signal once full rate adaptive filter 318 has converged. The convergence time can be defined as the interval between the instant a signal is applied to the R.sub.in input 308 of echo canceller 302 with the estimated echo path impulse response (e.g., the filter coefficients) initially set to zero, and the instant the returned echo level in error signal at output 314 reaches a defined level).
Regarding claim 7, Deng teaches the apparatus of claim 1, wherein the M+1 error estimators are configured to determine filter coefficients of the M+1 filter coefficients for which the measure of deviation between the first filtered reference signal and the received signal is below a predefined threshold (Abstract, paragraphs [0024]- [0026] and [0050]- [0053]; a coefficient threshold is determined. Thereafter, for each filter coefficient, a first step size is assigned to filter coefficients with a magnitude less than the coefficient threshold, and a second step size is assigned to filter coefficients with a magnitude greater than or equal to the coefficient threshold).
Regarding claim 8, Deng teaches the apparatus of claim 7, further comprising: a subtractor configured to determine the measure of deviation based on a subtraction of the first filtered reference signal from a digital representation of the received signal (paragraphs [0027], [0030] and [0050]-[0053]; a noninverting input of adder 128 can be coupled to an output of down sampler 108 in order to receive signal d(k), which can be a signal containing echo. The output of adder 128 can be an error signal, e(k), which can indicate the difference between signals y(k) and d(k), and which can be used as a feedback signal for signal estimator 11).
Regarding claim 9, Deng teaches the apparatus of claim 1, further comprising: an echo compensation filter having a set of N times M+1 filter coefficients, the echo compensation filter being configured to filter values of a transmit signal transmittable over the communication channel by using the M+1 filter coefficients for each time interval of the N time intervals (paragraphs [0050]-[0053]; the output of adder 322 is an error signal, which signal is the difference between the estimated signal and signal S.sub.in. This error signal is used both as feedback for signal estimator 320, and for the output signal S.sub.out, wherein S.sub.out should have a significantly reduced echo signal once full rate adaptive filter 318 has converged. The convergence time can be defined as the interval between the instant a signal is applied to the R.sub.in input 308 of echo canceller 302 with the estimated echo path impulse response (e.g., the filter coefficients) initially set to zero, and the instant the returned echo level in error signal at output 314 reaches a defined level).
Regarding claim 10, Deng teaches the apparatus of claim 1, further comprising: a controller configured to send a select signal to the interval selector for selecting a value of the k-th time interval applied by the interval selector (paragraphs [0050]-[0053]; the output of adder 322 is an error signal, which signal is the difference between the estimated signal and signal S.sub.in. This error signal is used both as feedback for signal estimator 320, and for the output signal S.sub.out, wherein S.sub.out should have a significantly reduced echo signal once full rate adaptive filter 318 has converged. The convergence time can be defined as the interval between the instant a signal is applied to the R.sub.in input 308 of echo canceller 302 with the estimated echo path impulse response (e.g., the filter coefficients) initially set to zero, and the instant the returned echo level in error signal at output 314 reaches a defined level).
Regarding claim 11, Deng teaches the apparatus of claim 10, further comprising: a plurality of spare resources configured to increase a number of filter coefficients of the finite impulse response filter (paragraphs [0050]-[0053]; increasing the adaptive filter convergence speed, which contributes to better voice quality of the echo canceller system).
Regarding claim 12, Deng teaches the apparatus of claim 11, wherein the spare resources comprise at least one additional filter coefficient; and wherein the finite impulse response filter is configured to filter the values of the reference signal using the M+1 filter coefficients and the at least one additional filter coefficient to obtain the first filtered reference signal (Abstract, (paragraphs [0020], [0029] and [0051]- [0053]; the pure delay can be defined as a time distance between the beginning of the impulse response and the time location of the impulse response absolute value that is significantly greater than the preceding values (e.g., impulse response absolute values that exceed a threshold). Alternatively, the pure delay can be a time distance between the beginning of the impulse response and the time location of the maximum value of the impulse response, reduced by the length of the preceding impulse segment that contains intermediate values between the maximum values and the average noise level value).
Regarding claim 13, Deng teaches the apparatus of claim 11, further comprising: a combination element for combining the finite impulse response filter with the plurality of spare resources (paragraphs [0050]-[0053]; the pure delay can be a time distance between the beginning of the impulse response and the time location of the maximum value of the impulse response, reduced by the length of the preceding impulse segment that contains intermediate values between the maximum values and the average noise level value. Further analysis of the filter coefficients can be used to determine whether the filter coefficients have converged on an echo signal).
Regarding claim 14, Deng teaches the apparatus of claim 11, wherein the controller is configured to send an enable signal to the plurality of spare resources and the combination element, the enable signal being configured to enable the plurality of spare resources and to enable the combination element (paragraphs [0050]-[0053]; the pure delay can be a time distance between the beginning of the impulse response and the time location of the maximum value of the impulse response, reduced by the length of the preceding impulse segment that contains intermediate values between the maximum values and the average noise level value. Further analysis of the filter coefficients can be used to determine whether the filter coefficients have converged on an echo signal).
Regarding claim 15, Deng teaches the apparatus of claim 1, further comprising: an analog front end configured to receive the received signal; and an analog-to-digital converter configured to convert the received signal processed by the analog front end into a digital representation (paragraph [0016]; network element is a channel bank, which is a device that converts analog to digital signals or vice-a-versa).
Regarding claim 16, Deng teaches a method for determining filter coefficients for an echo compensation filter for reduction of an echo signal in a communication channel upon a basis of a received signal, the received signal being a receivable version of a known reference signal transmittable over the communication channel, the receivable version of the reference signal being affected by the echo signal, an impulse response of the communication channel extending over N time intervals , the method comprising:
filtering values of the reference signal, by using a finite impulse response filter having a set of M+1 filter coefficients to obtain a first filtered reference signal (Abstract and paragraphs [0050]- [0053]; error signal is used both as feedback for signal estimator 320, and for the output signal S.sub.out, wherein S.sub.out should have a significantly reduced echo signal once full rate adaptive filter 318 has converged. The convergence time can be defined as the interval between the instant a signal is applied to the R.sub.in input 308 of echo canceller 302 with the estimated echo path impulse response (e.g., the filter coefficients) initially set to zero, and the instant the returned echo level in error signal at output 314 reaches a defined level);
selecting values of the reference signal associated with a k-th time interval of the N time intervals by an interval selector (paragraphs [0020] and [0051]- [0053]; a tapped-delay-line finite impulse response (FIR) filter which is adapted by changing or adapting the filter coefficients (FIR filter is part of Box with filter coefficients 118). Filter coefficients 118 can adapt to describe an impulse response of a connected linear system, which can produce an echo signal);
successively delaying the selected values of the reference signal by M delay elements, the M delay elements being arranged after the interval selector (paragraphs [0020] and [0051]- [0053]; a tapped-delay-line finite impulse response (FIR) filter which is adapted by changing or adapting the filter coefficients (FIR filter is part of Box with filter coefficients 118). Filter coefficients 118 can adapt to describe an impulse response of a connected linear system, which can produce an echo signal); and
determining, by M+1 error estimators, the M+1 filter coefficients in order to minimize a measure of a deviation between the first filtered reference signal and the received signal, wherein a first error estimator is coupled to an output of the interval selector, and wherein each of the remaining M error estimators is coupled to an output of a respective delay element (Abstract, paragraphs [0030] and [0050]-[0053]; the process calculates error sample e(k) according to the formula e(k)=d(k)-y(k), as depicted at 206. In one embodiment, calculating the error sample can be implemented using an adder, such as adder 128 in FIG. 1, wherein sample y(k) is input into an inverting input of the adder and sample d(k) is input into a noninverting input. The samples output by adder 128, which can be referred to as error signal e(k), represent the difference between estimated signal y(k) and the signal that can contain echo d(k)).
Conclusion
The prior art made of record and not relied upon is considered pertinent to applicant's disclosure.
U.S Patent No. 5,796,820 to Sasada discloses echo canceler includes an N-tap FIR filter and a coefficient controller. Previous filter coefficients which were generated before an estimation error occurrence when the transmitting output signal becomes greater in power than the transmitting input signal are obtained by using only a number S of coefficient correction values and another number (N-1+S) of sequentially received signals (Abstract).
U.S Patent No. 8,139,760 B2 to Dyba et al. discloses reducing an echo signal include assigning a subsegment delay to each of a plurality of subsegment adaptive filters. A send signal and a delayed receive signal are coupled to each of the subsegment adaptive filters, wherein the delayed receive signal is delayed according to a respective subsegment delay. A set of filter coefficients in each of the subsegment adaptive filters are adapted, in parallel, to correspond to a respective subsegment impulse response of a connected system. Each set of filter coefficients is analyzed to produce a pure delay, and the pure delay is used to delay the receive signal for a main adaptive filter (Abstract).
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AKELAW TESHALE
Primary Examiner
Art Unit 2694
/AKELAW TESHALE/ Primary Examiner, Art Unit 2694