DETAILED ACTION
Notice of Pre-AIA or AIA Status
The present application is being examined under the pre-AIA first to invent provisions.
Double Patenting
The nonstatutory double patenting rejection is based on a judicially created doctrine grounded in public policy (a policy reflected in the statute) so as to prevent the unjustified or improper timewise extension of the “right to exclude” granted by a patent and to prevent possible harassment by multiple assignees. A nonstatutory double patenting rejection is appropriate where the conflicting claims are not identical, but at least one examined application claim is not patentably distinct from the reference claim(s) because the examined application claim is either anticipated by, or would have been obvious over, the reference claim(s). See, e.g., In re Berg, 140 F.3d 1428, 46 USPQ2d 1226 (Fed. Cir. 1998); In re Goodman, 11 F.3d 1046, 29 USPQ2d 2010 (Fed. Cir. 1993); In re Longi, 759 F.2d 887, 225 USPQ 645 (Fed. Cir. 1985); In re Van Ornum, 686 F.2d 937, 214 USPQ 761 (CCPA 1982); In re Vogel, 422 F.2d 438, 164 USPQ 619 (CCPA 1970); In re Thorington, 418 F.2d 528, 163 USPQ 644 (CCPA 1969).
A timely filed terminal disclaimer in compliance with 37 CFR 1.321(c) or 1.321(d) may be used to overcome an actual or provisional rejection based on nonstatutory double patenting provided the reference application or patent either is shown to be commonly owned with the examined application, or claims an invention made as a result of activities undertaken within the scope of a joint research agreement. See MPEP § 717.02 for applications subject to examination under the first inventor to file provisions of the AIA as explained in MPEP § 2159. See MPEP § 2146 et seq. for applications not subject to examination under the first inventor to file provisions of the AIA . A terminal disclaimer must be signed in compliance with 37 CFR 1.321(b).
The filing of a terminal disclaimer by itself is not a complete reply to a nonstatutory double patenting (NSDP) rejection. A complete reply requires that the terminal disclaimer be accompanied by a reply requesting reconsideration of the prior Office action. Even where the NSDP rejection is provisional the reply must be complete. See MPEP § 804, subsection I.B.1. For a reply to a non-final Office action, see 37 CFR 1.111(a). For a reply to final Office action, see 37 CFR 1.113(c). A request for reconsideration while not provided for in 37 CFR 1.113(c) may be filed after final for consideration. See MPEP §§ 706.07(e) and 714.13.
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Claims 1-13 are rejected on the ground of nonstatutory double patenting as being unpatentable over claims 1-13 of U.S. Patent No. 12,154,583. Although the claims at issue are not identical, they are not patentably distinct from each other because of the following:
Pending Application No. 18/926,883
U.S. Patent No. 12,154,583
Claims 1, 12 and 13:
An apparatus for processing an audio signal, comprising:
an analysis filter bank comprising a first number of analysis filter bank channels for transforming the audio signal from a time domain to another domain,
a synthesis filter bank comprising a second number of synthesis filter bank channels for transforming the audio signal from the other domain to the time domain, and
a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels, so that an audio signal output of the synthesis filter bank comprises the predetermined sampling rate or a sampling rate being different from the predetermined sampling rate and being closer to the predetermined sampling rate than a sampling rate of an analysis filter bank input signal.
Claims 1, 12 and 13:
An apparatus for processing an audio signal, comprising:
a configurable first audio signal processor for processing an audio signal to acquire a processed audio signal,
an analysis filter bank comprising a first number of analysis filter bank channels,
a synthesis filter bank comprising a second number of synthesis filter bank channels, wherein the synthesis filter bank is adapted to transform an output of the analysis filter bank from the time-frequency domain to the time domain,
a second audio processor being adapted to receive and process the audio signal comprising a predetermined sampling rate, and
a controller for controlling the first number of analysis filter bank channels in accordance with a configuration setting provided to the configurable first audio signal processor, so that an audio signal output of the synthesis filter bank comprises the predetermined sampling rate or a sampling rate being different from the predetermined sampling rate and being closer to the predetermined sampling rate than a sampling rate of an analysis filter bank input signal,
wherein the apparatus is adapted to receive the configuration setting at run time.
Claim 2 corresponds to
Claim 2
Claim 3 corresponds to
Claim 3
Claim 4 corresponds to
Claim 4
Claim 5 corresponds to
Claim 5
Claim 6 corresponds to
Claim 6
Claim 7 corresponds to
Claim 7
Claim 8 corresponds to
Claim 8
Claim 9 corresponds to
Claim 9
Claim 10 corresponds to
Claim 10
Claim 11 corresponds to
Claim 11
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, 12 and 13 recites the limitation "the predetermined sampling rate". There is insufficient antecedent basis for this limitation in the claim.
Claim Rejections - 35 USC § 103
The following is a quotation of pre-AIA 35 U.S.C. 103(a) which forms the basis for all obviousness rejections set forth in this Office action:
(a) A patent may not be obtained though the invention is not identically disclosed or described as set forth in section 102, if the differences between the subject matter sought to be patented and the prior art are such that the subject matter as a whole would have been obvious at the time the invention was made to a person having ordinary skill in the art to which said subject matter pertains. Patentability shall not be negated by the manner in which the invention was made.
Claims 1-5, 12 and 13 is/are rejected under pre-AIA 35 U.S.C. 103(a) as being unpatentable over Liljeryd et al. (US 7,483,758) in view of Brennan et al. (US 6,236,731).
Claims 1 and 12-13,
Liljeryd teaches an apparatus for processing an audio signal, comprising ([col. 6 lines 41-59] Liljeryd states: “The present invention may be implemented in various kinds of systems for storage or transmission of audio signals using arbitrary codecs. FIG. 1 shows the decoder of an audio coding system”; Liljeryd further teaches “The digital wideband output signal is finally converted 107 to an analogue output signal”):
an analysis filter bank comprising a first number of analysis filter bank channels for transforming the audio signal from a time domain to another domain ([col. 4 lines 8-14] Liljeryd states: “The signal under consideration is decomposed into a series of subband signals by the analysis part of the filterbank”; [Fig. 2] [col. 4 lines 15-26] Liljeryd further states: “The analysis filter bank 201 splits the input signal into several subband signals”; [Fig. 2] [col. 4 lines 27-39] Liljeryd also states: “an L-channel filter bank splits the input signal x(n) into L subband signals”; Liljeryd teaches an analysis filter bank having a first number of channels, namely L channels, that transforms the input signal x(n) from the time domain into subband signals, i.e., another domain),
a synthesis filter bank comprising a second number of synthesis filter bank channels for transforming the audio signal from the other domain to the time domain ([col. 4 lines 15-39] Liljeryd states: “The synthesis filter bank 202 combines the subband samples in order to recreate the original signal”; Liljeryd further states: “The synthesis section, with the synthesis filters denoted F.sub.k(z), reassembles the subband signals after interpolation 205 and filtering 206 to produce {circumflex over (x)}(n)”; [col. 5 lines 7-25] Liljeryd also teaches “a QL-channel synthesis filter bank”; Liljeryd teaches a synthesis filter bank having a second number of synthesis filter bank reassembles subband-domain signals and produces a time-domain output signal, x^(n)), and
so that an audio signal output of the synthesis filter bank comprises the predetermined sampling rate or a sampling rate being different from the predetermined sampling rate and being closer to the predetermined sampling rate than a sampling rate of an analysis filter bank input signal ([col. 5 lines 7-43] Liljeryd teaches that filtering with a “QL-channel synthesis filter bank” results in “an output signal with sampling frequency Qf.sub.s”; Liljeryd further states: “any size of the synthesis filter bank may be used, resulting in different sampling rates of the output signal”; in a specific sample, Liljeryd states that subbands “may then be synthesized using a 28-channel filterbank” to produce “sampling frequency 28/16 f.sub.s=1.75 f.sub.s,” and that the subband signals “could also be synthesized using a 32-channel filterbank” to produce “sampling frequency 2 f.sub.s”).
The difference between the prior art and the claimed invention is that Liljeryd does not explicitly teach a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels.
Brennan teaches a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels ([Background] [Summary of the Invention] [col. 1 line 24 to col. 3 line 39] Brennan teaches that “the number of frequency bands and their bandwidth will usually be parameters that can be adjusted by the selection input.”; Brennan further states that “both of the first analysis filterbank means and the second synthesis filterbank means are connected to the selection input”; Brenna also teaches “a programmable digital signal processor for controlling the number of frequency bands and the bandwidth of each frequency band”; [col. 9 lines 47-57] Brennan further equates channels and bands by stating that the invention allows “the number of channels or bands and the width of those to be selected”).
Therefore, it would have been obvious to one of ordinary skill in the art, before the effective filing date of the claimed invention, to modify the teachings of Liljeryd with teachings of Brennan by modifying the spectral translation/folding in the subband domain as taught by Liljeryd to include a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels as taught by Brennan for the benefit of reducing memory usage (Brennan [Abstract]).
Claim 2,
Liljeryd further teaches the apparatus according to claim 1, wherein the analysis filter bank is adapted to transform the analysis filter bank input signal being represented in a time-domain into a first time-frequency domain audio signal comprising a plurality of first subband signals ([col. 4 lines 8-39] Liljeryd states: “The signal under consideration is decomposed into a series of subband signals by the analysis part of the filterbank”; Liljeryd further states: “The analysis filter bank 201 splits the input signal into several subband signals”; see Fig. 2; Liljeryd also states: “an L-channel filter bank splits the input signal x(n) into L subband signals”),
wherein the number of first subband signals is equal to the first number of analysis filter bank channels ([col. 4 lines 27-39] Liljeryd states: “an L-channel filter bank splits the input signal x(n) into L subband signals”; Liljeryd further identifies the analysis filters as “H.sub.k(z) 203, where k=0, 1, . . . , L-1”; see Fig. 2),
wherein the apparatus further comprises a signal adjuster being adapted to generate a second time-frequency domain audio signal comprising a plurality of second subband signals from the first time-frequency domain audio signal ([col. 6 lines 41-59] Liljeryd describes “the integrated translation or folding and envelope adjusting filterbank unit 105”; see Fig. 1; Liljeryd states that “the resulting spectral envelope information is fed together with the subband samples from the analysis filterbank to the integrated translation or folding and envelope adjusting filterbank unit 105”; Liljeryd further states: “This unit translates or folds the lowband signal, according to the present invention, to form a wideband signal and applies the transmitted spectral envelope”)
based on the configuration setting ([col. 6 lines 41-59] Liljeryd states that demultiplexer 101 separates “the envelope data and other HFR related control signals from the bitstream”; [col. 5 lines 26-43] Liljeryd also teaches parameter-based repatching: “frequency translation according to Eq. (3) is used with the following parameters: M=16, S=7 and P=1,” and later “Eq. (3) is used with the new parameter: M-23, S-5 and P=3”; see Fig. 3),
such that the number of second subband signals of the second time-frequency domain audio signal is equal to the number of synthesis filter bank channels ([col. 6 lines 41-59] Liljeryd states:” “the 16 subbands are extended to 23 subbands,” and then “the 23 subbands are extended to 28 subbands”; Liljeryd then states: “The so-produced subbands may then be synthesized using a 28-channel filterbank”), and
wherein the number of second subband signals of the second time-frequency domain audio signal is different from the number of subband signals of the first time-frequency domain audio signal ([col. 6 lines 41-59] Liljeryd states: “consider the subband channels from a 16-channel analysis filterbank,” and then explains that “the 16 subbands are extended to 23 subbands” and “the 23 subbands are extended to 28 subbands”; see Fig. 3), and
wherein the synthesis filter bank is adapted to transform the second time-frequency domain audio signal into a time domain audio signal as the audio signal output of the synthesis filter bank ([col. 4 lines 15-39] Liljeryd states: “The synthesis filter bank 202 combines the subband samples in order to recreate the original signal”; see Fig. 2; Liljeryd further states: “The synthesis section, with the synthesis filters denoted F.sub.k(z), reassembles the subband signals after interpolation 205 and filtering 206 to produce {circumflex over x(n)}(n)”; [col. 6 lines 41-59] Liljeryd also states that “processed subband samples are then fed to the synthesis filterbank 106,” and that “the digital wideband output signal is finally converted 107 to an analogue output signal”).
Claim 3,
Liljeryd further teaches the apparatus according to claim 2, wherein the signal adjuster is adapted to generate the second time-frequency domain audio signal by generating at least one additional subband signal ([col. 6 lines 41-59] Liljeryd identifies the signal-adjusting unit as the “integrated translation or folding and envelope adjusting filterbank unit 105”; see Fig. 1; Liljeryd states: “This unit translates or folds the lowband signal, according to the present invention, to form a wideband signal and applies the transmitted spectral envelope”; [col. 5 lines 7-43] Liljeryd further teaches that “the number of subband channels may be increase after the analysis filtering”; In the specific Fig. 3 example, Liljeryd states: “the 16 subbands are extended to 23 subbands,” and then “the 23 subbands are extended to 28 subbands”).
Claim 4,
Liljeryd further teaches the apparatus according to claim 3, wherein the signal adjuster is adapted to generate at least one additional subband signal by conducting spectral band replication to generate at least one additional subband signal ([col. 1 lines 15-26] Liljeryd explicitly ties its high-frequency reconstruction technique to spectral band replication: “The proposed invention offers a low-complexity, intermediate quality HFR method and relates to the PCT patent Spectral Band Replication (SBR) [WO 98/57436]”; [col. 1 line 52 to col. 2 line 3] Liljeryd explains the SBR/HFR principle: “The general idea with HFR techniques is to replace the original high frequency information with information created from the available lowband and subsequently apply spectral envelope adjustment to this information”; [col. 4 lines 8-14] Liljeryd further teaches the operative subband process: “The subband signals are then repatched, through reconnection of analysis- and synthesis subband channels, to achieve spectral translation or folding or a combination thereof”; [col. 4 lines 57-67] Liljeryd further teaches that “the number of subband channels may be increased after the analysis filtering,” and that, when “the L subband signals are repatched to the highband channels,” the bandwidth is increased; [col. 5 lines 26-43] in the Fig. 3 example, Liljeryd states that the “the 16 subbands are extended to 23 subbands,” and then “the 23 subbands are extended to 28 subbands”; see Fig. 3]; [col. 6 lines 41-59] Liljeryd also identifies the claimed signal adjuster as the “integrated translation or folding and envelope adjusting filterbank unit 105,” which “translates or folds the lowband signal … to form a wideband signal and applies the transmitted spectral envelope”).
Claim 5,
Liljeryd further teaches the apparatus according to claim 3, wherein the signal adjuster is adapted to generate a zero signal as additional subband signal ([col. 6 lines 51-59] Liljeryd identifies the relevant signal-adjusting unit as the “integrated translation or folding and envelope adjusting filterbank unit 105”; Liljeryd states that “the resulting spectral envelope information is fed together with the subband samples from the analysis filterbank to the integrated translation or folding and envelope adjusting filterbank unit 105”; see Fig. 1; [col. 5 lines 7-25] Liljeryd explicitly teaches zero-valued additional/highband channels: “Since, in this case, the L(Q-1) highband filters are unused (fed with zeros), the audio bandwidth will not change--the filter bank will merely reconstruct an upsampled version of {circumflex over (x)}(n)”; [col. 5 lines 26-43] Liljeryd also gives a specific synthesis-filterbank example: “The subband signals could also be synthesized using a 32-channel filterbank, where the four uppermost channels are fed with zeros, illustrated by the dashed lines in the figure, producing an output signal with sampling frequency 2 f.sub.s”; see Fig. 3]; [col. 6 lines 8-17] Additionally, in the guard-band embodiment, Liljeryd states: “In filterbank based translation or folding, guard-bands could be inserted and may preferably consist of one or several subband channels set to zero”).
Claim 6 is/are rejected under pre-AIA 35 U.S.C. 103(a) as being unpatentable over Liljeryd et al. (US 7,483,758) in view of Brennan et al. (US 6,236,731) and further in view of Schuijers et al. (US 2007/0038439).
Claim 6,
Liljeryd and Brennan teach all the limitations in claim 1. The difference between the prior art and the claimed invention is that Liljeryd nor Brennan explicitly teach wherein the analysis filter bank is a QMF analysis filter bank and wherein the synthesis filter bank is a QMF synthesis filter bank.
Schuijers teaches wherein the analysis filter bank is a QMF analysis filter bank and wherein the synthesis filter bank is a QMF synthesis filter bank ([0011] Schuijers teaches that a “QMF bank is used” and that Fig. 2 shows a “complex QMF analysis and synthesis filter bank”; [0030] Schuijers also explicitly identifies “QMF analysis filter bank 30” and “QMF synthesis filter bank 31”).
Therefore, it would have been obvious to one of ordinary skill in the art, before the effective filing date of the claimed invention, to modify the teachings of Liljeryd with teachings of Schuijers by modifying the spectral translation/folding in the subband domain as taught by Liljeryd to include wherein the analysis filter bank is a QMF analysis filter bank and wherein the synthesis filter bank is a QMF synthesis filter bank as taught by Schuijers for the benefit of obtaining the best audio quality where the delays in the plurality of delay units are monotonically increasing from high frequency to low frequency to efficiently processing the audio signal (Schuijers [0009]).
Claim 7 is/are rejected under pre-AIA 35 U.S.C. 103(a) as being unpatentable over Liljeryd et al. (US 7,483,758) in view of Brennan et al. (US 6,236,731) and further in view of Reznik et al. (US 2009/0099844).
Claim 7,
Liljeryd and Brennan teach all the limitations in claim 1. The difference between the prior art and the claimed invention is that Liljeryd nor Brennan explicitly teach wherein the analysis filter bank is an MDCT analysis filter bank and wherein the synthesis filter bank is an MDCT synthesis filter bank.
Reznik teaches wherein the analysis filter bank is an MDCT analysis filter bank and wherein the synthesis filter bank is an MDCT synthesis filter bank ([0011] Reznik states: “One feature provides a way to implement AAC-ELD or both AAC and AAC-ELD algorithms using the same core MDCT analysis filterbank and core IMDCT synthesis filterbank”; [0030] Reznik further states: “An MDCT Analysis Filterbank 106 … operates to decompose the time-domain input audio signal 104 into a plurality of sub-band signals and convert the signals to the frequency-domain”; [0031] Reznik also states: “An IMDCT Synthesis Filterbank 210 … operates to convert the frequency-domain signal 104 back to a time-domain audio signal 212”; [0061] [0082] In another synthesis embodiment, Reznik states: “The synthesis filterbank in AAC is simply an IMDCT filterbank 702” and that the algorithm includes “Apply the IMDCT transform … to obtain output samples”).
Therefore, it would have been obvious to one of ordinary skill in the art, before the effective filing date of the claimed invention, to modify the teachings of Liljeryd with teachings of Reznik by modifying the spectral translation/folding in the subband domain as taught by Liljeryd to include wherein the analysis filter bank is an MDCT analysis filter bank and wherein the synthesis filter bank is an MDCT synthesis filter bank as taught by Reznik for the benefit of simplifying a way to implement AAC-ELD or both AAC and AAC-ELD codec algorithms on the same DSP core (Reznik [0007]).
Claim 8 is/are rejected under pre-AIA 35 U.S.C. 103(a) as being unpatentable over Liljeryd et al. (US 7,483,758) in view of Brennan et al. (US 6,236,731) and further in view of ETSI (“Digital cellular telecommunications system (Phase 2+)…”; Jan. 2009; ETSI TS 126 402).
Claim 8,
Liljeryd and Brennan teach all the limitations in claim 1. The difference between the prior art and the claimed invention is that Liljeryd nor Brennan explicitly teach an additional resampler being adapted to receive a synthesis filter bank output signal comprising a first synthesis sampling rate, and wherein the additional resampler resamples the synthesis filter bank output signal to generate a resampled output signal comprising a second synthesis sampling rate.
ETSI teaches an additional resampler being adapted to receive a synthesis filter bank output signal comprising a first synthesis sampling rate ([7.] ETSI identifies an “output resampler tool” and states that a “downsampler tool is required” to convert audio output from a source sampling frequency to a target sampling frequency”; [Fig. 1] Fig. 1 places a Spline Resampler after the Synthesis QMF Bank and before the postfilter/output; [7.2] Section 7.2 further states that the resampler receives the “discrete time output x(l) from synthesis QMF bank”), and
wherein the additional resampler resamples the synthesis filter bank output signal to generate a resampled output signal comprising a second synthesis sampling rate ([7.2] ETSI teaches conversion from “source sampling frequency” to “target sampling frequency”; [7.3] in the spline-resampler section, ETSI defines the discrete-time resampler output as y(n) = s(n/Ft) and defines the resampling relationship using Fsout/Ft; The postfilter then produces the final output from the resampler output y(n)).
Therefore, it would have been obvious to one of ordinary skill in the art, before the effective filing date of the claimed invention, to modify the teachings of Liljeryd with teachings of ETSI by modifying the spectral translation/folding in the subband domain as taught by Liljeryd to include an additional resampler being adapted to receive a synthesis filter bank output signal comprising a first synthesis sampling rate, and wherein the additional resampler resamples the synthesis filter bank output signal to generate a resampled output signal comprising a second synthesis sampling rate as taught by ETSI for the benefit of converting the audio output from the source sampling frequency out to the target sampling frequency out (ETSI [7.]).
Claim 9 is/are rejected under pre-AIA 35 U.S.C. 103(a) as being unpatentable over Liljeryd et al. (US 7,483,758) in view of Brennan et al. (US 6,236,731) and further in view of Sande et al. (US 2005/0018796).
Claim 9,
Liljeryd and Brennan teach all the limitations in claim 1. The difference between the prior art and the claimed invention is that Liljeryd nor Brennan explicitly teach wherein the apparatus is adapted to feed a synthesis filter bank output signal comprising a first synthesis sampling rate into an analysis filter bank as an analysis filter bank input signal.
Sande teaches wherein the apparatus is adapted to feed a synthesis filter bank output signal comprising a first synthesis sampling rate into an analysis filter bank as an analysis filter bank input signal ([Abstract] Sande states: “An efficient method and structure for combining an M-channel synthesis filter bank followed by an L-channel analysis filter bank”; [0019] Sande further states: “The present invention provides a processing system having a synthesis filter bank and an analysis filter bank where the output of the synthesis filter bank is provided as an input to the analysis filter bank”; [0029] Sande also states: “The analysis filter bank 64 receives the output signal X(n) of the synthesis filter bank 58”; [0051] Sande further teaches the sampling-rate aspect: “the rate of the intermediate synthesized output is M fclock and the synthesized output is input to the analysis filter bank 20”).
Therefore, it would have been obvious to one of ordinary skill in the art, before the effective filing date of the claimed invention, to modify the teachings of Liljeryd with teachings of Sande by modifying the spectral translation/folding in the subband domain as taught by Liljeryd to include wherein the apparatus is adapted to feed a synthesis filter bank output signal comprising a first synthesis sampling rate into an analysis filter bank as an analysis filter bank input signal as taught by Sande for the benefit of decreasing circuit complexity (Sande [0006]).
Claims 10 and 11 is/are rejected under pre-AIA 35 U.S.C. 103(a) as being unpatentable over Liljeryd et al. (US 7,483,758) in view of Brennan et al. (US 6,236,731) and further in view of Ott (US 6,084,916).
Claim 10,
Liljeryd and Brennan teach all the limitations in claim 1. The difference between the prior art and the claimed invention is that Liljeryd nor Brennan explicitly teach wherein the controller is adapted to determine the first number of analysis filter bank channels or the second number of synthesis filter bank channels based on a tolerable error.
Ott teaches wherein the controller is adapted to determine the first number of analysis filter bank channels or the second number of synthesis filter bank channels based on a tolerable error ([col. 5 line 37 to col. 6 line 2] Ott teaches that a sampling rate may vary within a specified tolerance: “FR can vary, as long as it remains within the specified frequency tolerance”; Ott further states: “ DL represents a residual frequency error that is tolerated by the specifications of a given system. Given this extra degree of freedom, it is now possible to compute a solution for M, N, and D”; Ott provides the tolerance relationship: “absolute value of ((FT multiplied by N/FR multiplied by M) -1).”; Ott then states: “DL represents a frequency ‘window’ that has to be ‘hit’ by selecting proper values for M and N,” followed by an algorithm that computes M, computes D, and determines that “if the absolute value of D is, a solution has been found, else increase N”; [Claim 3] Ott also claims that the second converted sampling rate is adjusted “within a tolerance window” so that “lower values of M and N are utilized”).
Therefore, it would have been obvious to one of ordinary skill in the art, before the effective filing date of the claimed invention, to modify the teachings of Liljeryd with teachings of Ott by modifying the spectral translation/folding in the subband domain as taught by Liljeryd to include wherein the controller is adapted to determine the first number of analysis filter bank channels or the second number of synthesis filter bank channels based on a tolerable error as taught by Ott for the benefit of reducing signal distortions caused by asynchronous sample rates between digital information systems (Ott [Technical Field]).
Claim 11,
Ott further teaches the apparatus according to claim 10, wherein the controller comprises an error comparator for comparing the actual error with a tolerable error ([col. 5 line 48 to col. 6 line 2] Ott teaches a tolerated error value: “DL represents a residual frequency error that is tolerated by the specifications of a given system”; Ott then defines the tolerated-error relationship: “absolute value of((FT multiplied by N/FR multiplied by M)-1). ltoreq.DL. Equation 4”; Ott further states that “DL represents a frequency "window" that has to be "hit" by selecting proper values for M and N” and gives an algorithm including “compute D=(FT multiplied by N)/(FR multiplied by M)-1” and “if the absolute value of D is .ltoreq.DL, a solution has been found, else increase N by 1”).
Conclusion
The prior art made of record and not relied upon is considered pertinent to applicant's disclosure.
Oshikiri (US 7,756,711) – A coding apparatus capable of reducing a circuit scale and also reducing the amount of coding processing calculation is disclosed. In this apparatus, frequency domain conversion section (103) performs a frequency analysis of the signal sampled at a sampling rate Fx with an analysis length of 2Na and calculates first spectrum S1(k) (0.ltoreq.k<Na). Band extension section (104) extends the effective frequency band of first spectrum S1(k) to 0.ltoreq.k<Nb so that a new spectrum can be assigned to the extended area following to the frequency k=Na of first spectrum S1(k). Extended spectrum assignment section (105) assigns extended spectrum S1'(k) (Na.ltoreq.k<Nb) input to the extended frequency band from outside. Spectral information specification section (106) outputs information necessary to specify extended spectrum S1'(k) out of the spectrum given from extended spectrum assignment section (105) as a code.
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SHREYANS A. PATEL
Primary Examiner
Art Unit 2653
/SHREYANS A PATEL/ Examiner, Art Unit 2659
Prosecution Insights