DETAILED ACTION
Claims 1 – 20 are pending.
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 .
Information Disclosure Statement
The information disclosure statements (IDS) submitted on 20 December 2024, 23 December 2024, 19 February 2025, 28 February 2025, 12 March 2025, 20 May 2025, 04 September 2025, 12 November 2025, 18 February 2026, 17 May 2026 and 21 July 2026 are in compliance with the provisions of 37 CFR 1.97. Accordingly, the information disclosure statement is being considered by the Examiner.
EXAMINER’S COMMENT
The claims of this application were subject to double patenting rejections over U.S. Application 18/448,020 (now issued as U.S. 12,725,620), and patented U.S. 11,915,712 B2, U.S. 11,410,668 B2, U.S. 11,049,508 B2, U.S. 10,332,535 B2, and U.S. 10,236,007 B2. The Examiner discussed with Attorney ZIEMIAN Robert on Thursday 13 August 2026 to inform him of this. The Applicant filed a terminal disclaimer to address this 19 August 2026. The terminal disclaimer was approved also on 19 August 2026.
As a result of this, the Examiner does not provide a non-statutory double patenting rejection over the claims of the U.S. patents mentioned above.
Claim Objections
Claims 6, 8, 9, 15 and 16 are objected to because of the following informalities:
Claim 6 recites in its second limitation, ‘a delay stage configured for delaying the decoded version of the first decoded signal portion …’ which the Examiner believes should be --encoded-- to fit with the limitation provided by the first limitation of the same claim 6. For the purpose of this Office Action, the Examiner will treat the [[decoded signal portion]] as --encoded signal portion--
Claims 8 and 9 are recite the ‘audio decoder’ while being dependent on claim 1 which is directed to an audio encoder, clearly different. The Examiner suggests amending the claims to instead read --The audio encoder-- just as presented for both claims 6 and 7. The Examiner will treat these claims as each being directed to ‘audio encoder’ for the purpose of this Office Action.
Claims 15 and 16 are presented as being dependent on claim 8, but reading the limitations of these claims leads the Examiner to believe these claims should instead depend on claim 14, not claim 8. Claim 14 is directed to a decoder, and while claim 8 also appears to be directed to a decoder, it has been indicated earlier that that appears to be an error as the subject matter of claim 8 are more in line with an encoder. Claims 15 and 16 also mention ‘the decoded spectral representation’ which has antecedent basis in claim 14, not claim 8, leading them Examiner to believe that they should depend on claim 14, and not claim 8. For the purpose of this Office Action, the Examiner will treat both claims 15 and 6 as being dependent on claim 14.
Appropriate correction is required.
Claim Interpretation
The following is a quotation of 35 U.S.C. 112(f):
(f) Element in Claim for a Combination. – An element in a claim for a combination may be expressed as a means or step for performing a specified function without the recital of structure, material, or acts in support thereof, and such claim shall be construed to cover the corresponding structure, material, or acts described in the specification and equivalents thereof.
The following is a quotation of pre-AIA 35 U.S.C. 112, sixth paragraph:
An element in a claim for a combination may be expressed as a means or step for performing a specified function without the recital of structure, material, or acts in support thereof, and such claim shall be construed to cover the corresponding structure, material, or acts described in the specification and equivalents thereof.
This application includes one or more claim limitations that do not use the word “means,” but are nonetheless being interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph, because the claim limitation(s) uses a generic placeholder that is coupled with functional language without reciting sufficient structure to perform the recited function and the generic placeholder is not preceded by a structural modifier. Such claim limitations are:
‘an encoded signal former configured for forming an encoded audio signal …’ in claim 1; and
‘a combiner configured for combining the decoded first audio signal portion …’ in claim 14.
Because these claim limitations are being interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph, they are being interpreted to cover a microprocessor or a programmable computer from page 49 par 2 as the corresponding structure described in the Specification as performing the claimed function, and equivalents thereof.
If Applicant does not intend to have these limitations interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph, Applicant may: (1) amend the claim limitations to avoid them being interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph (e.g., by reciting sufficient structure to perform the claimed function); or (2) present a sufficient showing that the claim limitations recite sufficient structure to perform the claimed function so as to avoid them being interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph.
This application includes one or more claim limitations that use the word “means” or “step” or a term used as a substitute for “means” that is a generic placeholder (also called a nonce term or a non-structural term having no specific structural meaning) but are nonetheless not being interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph because the claim limitations recite sufficient structure, materials, or acts to entirely perform the recited function. Such claim limitations are:
“a first encoding processor configured for encoding …” in claim 1;
“a second encoding processor configured for encoding …” in claim 1;
“a cross-processor configured for calculating …” in claim 1;
“a controller configured for analyzing the audio signal …” in claim 1;
“a sampling rate converter configured for converting …” in claim 2;
“a time domain low band encoder configured for time domain …” in claim 2;
“a time domain bandwidth extensions encoder configured for …” in claim 2;
“a preprocessor configured for preprocessing …” in claim 3;
“… a prediction analyzer configured for determining …” in claim 3;
“a spectral decoder configured for calculating a …” in claim 6;
“a pre-emphasis filter configured for filtering …” in claim 9;
“a first decoding processor configured for decoding …” in claim 14; and
“a second decoding processor configured for decoding …” in claim 14.
These limitations are presented as processors or a controller, or as being an embodiment of a processor, which would clearly present a known structure, or are covered under a limitation presented as a processor.
Because these claim limitations are not being interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph, they are not being interpreted to cover only the corresponding structure, material, or acts described in the specification as performing the claimed function, and equivalents thereof.
If Applicant intends to have these limitations interpreted under 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph, Applicant may: (1) amend the claim limitations to remove the structure, materials, or acts that performs the claimed function; or (2) present a sufficient showing that the claim limitations do not recite sufficient structure, materials, or acts to perform the claimed function.
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 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, 11, 14, 17, 18, 19 and 20 are rejected under 35 U.S.C. 102(a)(1) as being anticipated by Makinen (US 2005/0256701 A1).
For claim 1, Makinen discloses an audio encoder for encoding an audio signal (Makinen: [0023] — a hardware electronic device suitable to performing the functions of an encoder), comprising:
a first encoding processor configured for encoding a first audio signal portion in a frequency domain (Makinen: [0043] — a frame (as a portion of the audio signal) is encoded by a TCX (transform coding) model),
a second encoding processor configured for encoding a second different audio signal portion in a time domain (Makinen: [0043] — encoding by ACELP (which is also a time-domain coding));
a cross-processor configured for calculating, from an encoded spectral representation of the first audio signal portion, initialization data of the second encoding processor, so that the second encoding processor is initialized to encode the second different audio signal portion immediately following the first audio signal portion in time in the audio signal (Makinen: [0006] — an AMR-WB codec which utilises both ACELP and TCX coding models (indicating a cross processing of both frequency and time domain ); FIG. 3 — an encoding of a TCX frame which immediately follows an ACELP encoding (indicating cross-processing); Fig. 2, [0041]-[0047] — whereby the signal is divided into different frames and a classification is made for the frames to determine whether the frames are music-like or speech-like, considering each superframe after the other and encoding each based on the classification);
a controller configured for analyzing the audio signal and configured for determining, which portion of the audio signal is the first audio signal portion encoded in the frequency domain and which portion of the audio signal is the second audio signal portion encoded in the time domain (Makinen: [0045] — an evaluation portion able to determine parts of the audio that are music-like or speech-like (and thereby able to determine which portion of the signal was encoded in either the frequency domain or the time domain since each of the audio types is encoded based on the audio type it falls under)); and
an encoded signal former configured for forming an encoded audio signal comprising a first encoded signal portion for the first audio signal portion and a second encoded signal portion for the second audio signal portion (Makinen: [0022] — encoding an audio signal by applying both coding models (assigned for both the first and the second portions)).
For claim 11, claim 1 is incorporated and Makinen discloses the audio encoder, wherein the second encoding processor comprises at least one element of the following group of elements: a prediction analysis filter; an adaptive codebook stage; an innovative codebook stage; an estimator configured for estimating an innovative codebook entry; an ACELP/gain coding stage (Makinen: [0039] — an ACELP coding model); a prediction synthesis filtering stage; a de-emphasis stage; and a bass post-filter analysis stage.
For claim 14, it is analysed and rejected by the same reasons set forth in the rejection of claim 1 above given that the limitations of this claim constitute a decoding process which is a reverse of the encoding process provided by the reference claim with both instant claims having similar limitations. The reference of Makinen in [0024] provides the decoding stage as a direct reverse of the consecutive encoding sections, and the presence of a hardware device capable of performing the functions of a decoder in [0070]. Claim 14 is hereby rejected under the same reasons set forth in claim 1, claim 14 being its decoding process.
As for claim 17, method claim 17 and system claim 1 are related as method detailing procedures for using the claimed system, with each claimed element’s function corresponding to the claimed system parts. Accordingly, claim 17 is similarly rejected under the same rationale as applied above with respect to system claim 1.
As for claim 18, method claim 18 and system claim 14 are related as method detailing procedures for using the claimed system, with each claimed element’s function corresponding to the claimed system parts. Accordingly, claim 18 is similarly rejected under the same rationale as applied above with respect to system claim 14.
As for claim 19, computer program product claim 19 and system claim 1 are related as computer program product storing executable instructions required for the operation of the system as programmable steps on a computer. Makinen provides teaching for the storage of software code necessary for performing the techniques in [0025]. Accordingly, claim 19 is similarly rejected under the same rationale as applied above with respect to system claim 1.
As for claim 20, computer program product claim 20 and system claim 14 are related as computer program product storing executable instructions required for the operation of the system as programmable steps on a computer. Makinen provides teaching for the storage of software code necessary for performing the techniques in [0025]. Accordingly, claim 20 is similarly rejected under the same rationale as applied above with respect to system claim 14.
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.
Claim 2 is rejected under 35 U.S.C. 103 as being unpatentable over Makinen (US 2005/0256701 A1) as applied to claim 1, in view of Moriss (US 2010/0076755 A1: hereafter — Morii) and further in view of Kim et al. (US 2012/0209600 A1: hereafter — Kim).
For claim 2, claim 1 is incorporated and Makinen discloses the audio encoder, wherein the audio signal comprises a high band and a low band (Makinen: [0045] — input signal having lower and higher frequency bands).
The reference of Makinen provides teaching for the presence of a second encoder with the encoding of a signal portion in time domain. This however differs from the claimed invention in that the claimed invention further provides teaching for performing sampling rate conversion.
This isn’t new to the art as the reference of Morii is now introduced to teach this as:
wherein the second encoding processor comprises:
a sampling rate converter configured for converting the second audio signal portion to a lower sampling rate representation having a second sampling rate, the second sampling rate of the lower sampling rate representation being lower than a first sampling rate of the audio signal, wherein the lower sampling rate representation does not comprise the high band of the audio signal (Morii: [0034] — a sample rate conversion that down-samples the frequency of an input signal from 16 kHz to 8kHz, so that the upper limit of the down-sampled signal is 4 kHz (a down-sampling being known to reduce the sampling rate of an audio so that no signal component has a frequency higher than half of its sampling rate, leading to blocking high frequency components or the high bands)).
The reference of Makinen teaches the presence of a second encoder with the encoding of a signal portion in time domain but differs from the claimed invention in that the claimed invention further teaches of converting the sampling rate of the audio portion to a lower sampling rate which is lower than a first sampling rate of the audio signal. This isn’t new to the art as the reference of Morii is seen to teach above.
Hence, before the effective filing date of the claimed invention, one of ordinary skill in the art would have found it obvious to improve upon the teaching for Makinen which provides a second encoder with the encoding of a signal portion in time domain, by incorporating the known teaching of Morii which converts the sampling rate of the audio signal to a lower sampling rate, to thereby come up with the claimed invention. The combination of both prior art elements would have provided the predictable result of lowering the sampling rate of an audio signal which leads to an operating system that consumes fewer resources, while also removing high-band frequency components that may not be useful for the intended processes. See KSR Int’l Co. v. Teleflex Inc., 550 U.S. 398, 415-421, 82 USPQ2d 1385, 1395-97 (2007).
The combination of Makinen in view of Mani provides teaching for a sampling rate conversion, but differs from the claimed invention in that the claimed invention further provides teaching for time encoding the low band and parametrically encoding the high band.
The reference of Kim is now introduced to teach this as:
a time domain low band encoder configured for time domain encoding the lower sampling rate representation (Kim: [0243] — separation of the high and low frequency bands for different types of encoders, and ‘the audio signal limited in the low band may be encoded through a core encoding of AAC’; FIG. 49 Parts 4942, 4944; [0279] — low core band encoder in time domain; [0280] — utilising parameter stereo based encoder to perform encoding); and
a time domain bandwidth extension encoder configured for parametrically encoding the high band (Kim: [0243] — separation of the high and low frequency bands for different types of encoders and ‘an audio signal corresponding to the high band may be represented as additional data for SBE and may be transferred to a decoding apparatus’; FIG. 49 Parts 4930, 4950; [0279]-[0280] — time domain bandwidth extension encoder and utilising parameter stereo based encoder to perform encoding).
The combination of Makinen in view of Mani provides teaching for the indication that the audio signal comprises low band and high band portions and the time domain encoding of the audio signals. It differs from the claimed invention in that the claimed invention further provides teaching for the separate encoding of lower sampling rate representations and high band portions. This is however not new to the art as the reference of Kim is provided to teach above.
Hence, before the effective filing date of the claimed invention, one of ordinary skill in the art would have found it obvious to improve upon the teaching for the combination of Makinen in view of Mani which indicates that the audio signal comprises low band and high band portions and also the time domain encoding of the audio signals, by incorporating the known teaching of Kim which provides the separate encoding of lower sampling rate representations and high band portions, to thereby come up with the claimed invention. The combination of both prior art elements would have provided the predictable result of adaptively encoding different forms of audio signals according to the encoding form best suited to each. See KSR Int’l Co. v. Teleflex Inc., 550 U.S. 398, 415-421, 82 USPQ2d 1385, 1395-97 (2007).
Claim 3 is rejected under 35 U.S.C. 103 as being unpatentable over Makinen (US 2005/0256701 A1) as applied to claim 1 in view of Kim (US 2012/0209600 A1).
For claim 3, claim 1 is incorporated and Makinen discloses the audio encoder, further comprising:
a preprocessor configured for preprocessing the first audio signal portion and the second different audio signal portion (Makinen: [0043] — calculating linear prediction coefficients for each audio signal frame whereby each is to be encoded either as ACELP coding or a TCX model (indicating a preparation of the frames of the audio signal for either time encoding or frequency encoding, further indicating the preparation of each of the first audio signal portions or the second audio signal portions, also the calculation of the linear prediction coefficients being a preprocessing function); [0040] — an encoding portion being realised by being run on a processing component (indicating the presence of a preprocessor serving the preprocessing functions)),
wherein the preprocessor comprises a prediction analyzer configured for determining prediction coefficients (Makinen: [0040] — an encoding portion being realised by being run on a processing component (indicating the presence of a preprocessor serving the preprocessing functions); [0043] — determining linear prediction coefficients of the audio signal (this being a prediction analysis to determine prediction coefficients)).
The reference of Makinen however fails to teach of introducing an encoded version of the prediction coefficients into the encoded audio signal.
This is however not new to the art as the reference of Kim is now introduced to teach this as:
wherein the encoded signal former is configured for introducing an encoded version of the prediction coefficients into the encoded audio signal (Kim: FIG. 1 — a Bitstream Multiplexer which is able to introduce prediction coefficients into an encoded audio signal; [0289] — describes the bitstream demultiplexer which is a reverse process of the multiplexer).
The reference of Makinen provides teaching for obtaining prediction coefficients for audio signal portions to either be processed in the time domain or the frequency domain, but differs from the claimed invention in that the claimed invention now further provides teaching for introducing an encoded version of the prediction coefficient into the encoded audio signal. This isn’t new to the art as the reference of Kim is seen to teach above.
Hence, before the effective filing date of the claimed invention, one of ordinary skill in the art would have found it obvious to combine the teaching of Makinen which obtains prediction coefficients for audio signal portions, with the known teaching of Kim which further provides the introduction of an encoded version of the prediction coefficients into an encoded audio signal, to thereby come up with the claimed invention. The combination of both prior art elements would have provided the predictable result of transmitting all intended signal prediction coefficients as one combined signal, instead of an individual transmission of the calculated prediction coefficients, leading to a more organised storage and transmission of the encoding parameters. See KSR Int’l Co. v. Teleflex Inc., 550 U.S. 398, 415-421, 82 USPQ2d 1385, 1395-97 (2007).
Claim 4 is rejected under 35 U.S.C. 103 as being unpatentable over Makinen (US 2005/0256701 A1) as applied to claim 1 in view of Zhang, Fan, et al. (“Adaptive prediction order scheme for AMR-WB+.” 2010 10th International Symposium on Communications and Information Technologies. IEEE, 2010: hereafter — Zhang).
For claim 4, claim 1 is incorporated and the reference of Makinen discloses the audio encoder, comprising:
a preprocessor configured for preprocessing the first audio signal portion and the second different audio signal portion (Makinen: [0043] — calculating linear prediction coefficients for each audio signal frame whereby each is to be encoded either as ACELP coding or a TCX model (indicating a preparation of the frames of the audio signal for either time encoding or frequency encoding, further indicating the preparation of each of the first audio signal portions or the second audio signal portions, also the calculation of the linear prediction coefficients being a preprocessing function); [0040] — an encoding portion being realised by being run on a processing component (indicating the presence of a preprocessor serving the preprocessing functions)).
The reference of Makinen however fails to teach of a resampler resampling the audio signal.
This is however not new to the art as the reference of Zhang is now introduced to teach this as:
wherein the preprocessor comprises a resampler configured for resampling the audio signal to a sampling rate of the second encoding processor to obtain a resampled audio signal (Zhang: page 2 col 1 par 3 — resampling an input audio sequence to an internal sampling frequency (producing a resampled signal)Sperschneider: FIG. 1 Part 28 — taking this downsampling stage as the pre-processor containing the resampler; [0004] — downsampling the sampling rate of the input stream to the sampling rate of the CELP coder); and
wherein the preprocessor comprises a prediction analyzer configured to determine prediction coefficients using the resampled audio signal Zhang: page 2 col 1 par 3 — performing ACELP on the low-frequency signal after the resampling of the signal (ACELP here would also indicate the obtaining of linear prediction coefficients as ACELP is a linear predictive coding)).
The reference of Makinen provides teaching for a preprocessing operation that calculates prediction coefficients for preparing frames of an audio signal for either a time domain or frequency domain encoding, setting both a first signal portion and a second signal portion. This differs from the claimed invention in that the claimed invention further provides resampling the audio signal to a sampling rate of the second encoding processor to obtain a resampled audio signal, and then computing prediction coefficients for the resampled signal. This isn’t new to the art as the reference of Zhang is seen to teach above.
Hence, before the effective filing date of the claimed invention, one of ordinary skill in the art would have found it obvious to combine the teaching of Makinen which obtains prediction coefficients for audio signal portions, with the known teaching of Zhang which resamples an audio signal and obtains the prediction coefficients of the resampled signal, to thereby come up with the claimed invention. The combination of both prior art elements would have provided the predictable result of performing audio signal encoding input signals that have been sampled to a predetermined sampling frequency, resulting in uniformity while computing the prediction coefficients of all the audio signal portions. See KSR Int’l Co. v. Teleflex Inc., 550 U.S. 398, 415-421, 82 USPQ2d 1385, 1395-97 (2007).
Claim 5 is rejected under 35 U.S.C. 103 as being unpatentable over Makinen (US 2005/0256701 A1) as applied to claim 1 in view of Yu et al. (US 2011/0010168 A1: hereafter — Yu).
For claim 5, claim 1 is incorporated and the combination of Makinen provides teaching for a cross-processor suitable for encoding a second audio signal portion following a first audio signal portion. This reference however fails to teach of the cross-processing comprising long term prediction analysis so as to determine long-term parameters.
This is however not new to the art as the reference of Yu is now introduced to teach as:
the audio encoder, wherein the cross-processor comprises a long term prediction analysis stage configured for determining one or more long term prediction parameters for the first audio signal portion (Yu: [0010], [0045], [0049] — long term prediction to obtain long term prediction parameters of an audio signal).
The reference of Makinen provides teaching for a cross-processor suitable for encoding a second audio signal portion following a first audio signal portion but fails to teach of determining one or more long term prediction parameters. This isn’t new to the art as the reference of Yu, as seen above, is provided to teach this.
Hence, before the effective filing date of the claimed invention, one of ordinary skill in the art would have found it obvious to improve upon the teaching of Makinen which provides a cross-processing function, by incorporating the teaching of Yu which provides determining long-term prediction parameters for an audio signal, to thereby come up with the claimed invention. The combination of both prior art elements would have provided the predictable result of the presence of long-term prediction being suitable for coding signals with strong harmonic energies (Yu: [0045]) so as to reduce the overall signal energy and also lower the bit rates required for encoding. See KSR Int’l Co. v. Teleflex Inc., 550 U.S. 398, 415-421, 82 USPQ2d 1385, 1395-97 (2007).
Claims 6, 8, 9, 12 and 16 are rejected under 35 U.S.C. 103 as being unpatentable over Makinen (US 2005/0256701 A1) as applied to claims 1 and 14 in view of Mittal et al. (US 2013/0030798 A1: hereafter — Mittal).
For claim 6, claim 1 is incorporated and the reference of Makinen provides teaching for performing cross-processing of a time domain and frequency domain encoding, and also for
the audio encoder, wherein the cross-processor comprises:
a spectral decoder configured for calculating a decoded version of the first encoded signal portion (Makinen: [0070] — performing TCX decoding (which is a spectral decoding)).
The reference of Makinen however differs from the claimed invention in that the claimed invention further provides teaching for feeding a delayed version of a decoded signal portion into a de-emphasis stage for initialisation.
This isn’t new to the art as the reference of Mittal is now introduced to teach this as:
the audio encoder, wherein the cross-processor comprises:
a spectral decoder configured for calculating a decoded version of the first encoded signal portion (Mittal: [0025], [0034] — reconstruction of audio from previously coded audio); and
a delay stage configured for delaying the decoded version of the first decoded signal portion to obtain a delayed version and for feeding the delayed version into a de-emphasis stage of the second encoding processor for initialization (Mittal: [0025] — back-propagation of reconstructed (decoded) audio from previously coded generic audio frame; [0026]–[0033] — the subsequent decoded audio frame m+1 (indicating a delay) is sent to a state generator where parameters for a speech coder are determined, this being performed through the use of a de-emphasis filter state memory, and then, the parameters of the de-emphasis are used as initialisation states for encoding a subsequent speech frame).
Hence, before the effective filing date of the claimed invention, one of ordinary skill in the art would have found it obvious to improve upon the teaching of Makinen which provides a cross-processing of both time-domain and frequency-domain encoding including a spectral decoder, by making use of the known technique of Mittal which provides feeding a delayed version of a decoded signal portion into a de-emphasis stage for initialisation, to thereby come up with the claimed invention. The combination of both prior art elements would have provided the predictable result of the parameters from the de-emphasis stage being suitable for improving the switching between frequency and time domain encodings. See KSR Int’l Co. v. Teleflex Inc., 550 U.S. 398, 415-421, 82 USPQ2d 1385, 1395-97 (2007).
For claim 8, claim 1 is incorporated and the reference of Makinen provides teaching for performing cross-processing of a time domain and frequency domain encoding, and also for
the audio [[decoder]]encoder1, wherein the cross-processor comprises:
a spectral decoder configured for calculating a decoded version of the first encoded signal portion (Makinen: [0070] — performing TCX decoding (which is a spectral decoding)).
The reference of Makinen however differs from the claimed invention in that the claimed invention further provides teaching for feeding a filter residual signal from an analysis filter stage filtering of a decoded signal version, into an adaptive codebook determiner for the purpose of initialisation.
This isn’t new to the art as the reference of Mittal is now introduced to teach this as:
an analysis filtering stage configured for filtering the decoded version of the first encoded signal portion [[or a pre-emphasized version derived by a pe-emphasis stage from the decoded version of the first encoded signal portion]] to obtain a filter residual signal and configured for feeding the filter residual signal into an adaptive codebook determiner of the second encoding processor for initialization (Mittal: [0026]–[0033] — the subsequent decoded audio frame m+1 (indicating a delay) is sent to a state generator where parameters for a speech coder are determined, this being performed through the use of a pre-emphasis filter state memory, and then, the parameters of the pre-emphasis are used as initialisation states for encoding a subsequent speech frame; [0046] — signal being passed through an analysis filter in order to generate the residual signal; [0067] — a generated residual signal being used for the generation of the adaptive codebook state).
Hence, before the effective filing date of the claimed invention, one of ordinary skill in the art would have found it obvious to improve upon the teaching of Makinen which provides a cross-processing of both time-domain and frequency-domain encoding including a spectral decoder, by applying the known technique of Mittal which provides feeding a filter residual signal from an analysis filter stage filtering of a decoded signal version, into an adaptive codebook determiner for the purpose of initialisation, to thereby come up with the claimed invention. The combination of both prior art elements would have provided the predictable result that the adaptive codebook initialisation from the residual signal gives the encoder a better representation and prediction reference for each current speech frame, so that later reconstruction can be effectively performed. See KSR Int’l Co. v. Teleflex Inc., 550 U.S. 398, 415-421, 82 USPQ2d 1385, 1395-97 (2007).
For claim 9, claim 1 is incorporated and the reference of Makinen provides teaching for performing cross-processing of a time domain and frequency domain encoding, and also for
the audio [[decoder]]encoder2, wherein the cross-processor comprises:
a spectral decoder configured for calculating a decoded version of the first encoded signal portion (Makinen: [0070] — performing TCX decoding (which is a spectral decoding)).
The reference of Makinen however differs from the claimed invention in that the claimed invention further provides teaching for feeding a pre-emphasis version or a delayed pre-emphasis version to a synthesis filtering stage for initialisation.
This isn’t new to the art as the reference of Mittal is now introduced to teach this as:
a pre-emphasis filter configured for filtering the decoded version of the first encoded signal portion to obtain a pre-emphasized version and configured for feeding the pre-emphasized version or a delayed pre-emphasized version to a synthesis filtering stage of the second encoding processor for initialization (Mittal: [0025] — back-propagation of reconstructed (decoded) audio from previously coded generic audio frame; [0026]–[0033] — the subsequent decoded audio frame m+1 (indicating a delay) is sent to a state generator where parameters for a speech coder are determined, this being performed through the use of a pre-emphasis filter state memory, and then, the parameters of the pre-emphasis are used as initialisation states for encoding a subsequent speech frame; FIG. 6 — the delayed pre-emphasised version gets fed to an LPC synthesis filter, all of these being enacted at a speech encoder).
Hence, before the effective filing date of the claimed invention, one of ordinary skill in the art would have found it obvious to improve upon the teaching of Makinen which provides a cross-processing of both time-domain and frequency-domain encoding including a spectral decoder, by making use of the known technique of Mittal which provides feeding a pre-emphasised version of the decoded version of the encoded audio signal, into a synthesis filtering stage of an encoding processor for initialization, to thereby come up with the claimed invention. The combination of both prior art elements would have provided the predictable result that the pre-emphasis results in establishing the state of a time-domain synthesis filter, so as to make the transition between both the frequency encoding and the time encoding much smoother. See KSR Int’l Co. v. Teleflex Inc., 550 U.S. 398, 415-421, 82 USPQ2d 1385, 1395-97 (2007).
For claim 12, claim 1 is incorporated and the reference of Makinen discloses a hybrid frequency and time encoding of an audio signal. This however differs from the claimed invention in that the claimed invention further provides down-sampling of the audio signal and selecting a low-band portion so as to obtain initialisation data.
This isn’t new to the art as the reference of Mittal is now introduced to teach as:
the audio encoder, wherein the cross-processor is configured to use a frequency-time transform additionally performing a downsampling from the first sampling rate to the second sampling rate using selecting a low band portion of the frequency domain representation with a reduced transform size to obtain the initialization data of the second encoding processor (Mittal: FIG. 6, [0062]–[0063] — down-sampling of the audio (from a first sampling rate to a second sampling rate, down-sampling also resulting in reduced sample or transform size), leading to the use of a low-pass filter (to obtain a low-band portion of the frequency domain representation with reduced transform size); [0026]–[0033] — the values of the down-sampling filter are used at the speech encoder for initialisation; [0052] — down-sampling from 32 kHz and 48 kHz sampling to 12.8 kHz (from a first sampling rate to a second sampling rate)).
Hence, before the effective filing date of the claimed invention, one of ordinary skill in the art would have found it obvious to improve upon the audio encoding process as provided by the reference of Makinen, by incorporating the known teaching of Mittal which provides down-sampling of the audio signal from a first sampling rate to a second sampling rate through the use of a low-pass filter resulting in the selection of a low band frequency region, so as to obtain initialisation data, to thereby come up with the claimed invention. The combination of both prior art elements would have provided the predictable result of reducing the number of samples that would be encode through down-sampling while focusing on only relevant frequency bands. See KSR Int’l Co. v. Teleflex Inc., 550 U.S. 398, 415-421, 82 USPQ2d 1385, 1395-97 (2007).
For claim 163, claim 14 is incorporated and the reference of Makinen provides teaching for obtaining a decoded spectral representation of a first encoded audio signal. This reference however fails to teach of the down-sampling of the audio signal and selecting a low-band portion to obtain initialisation data.
This however isn’t new to the art as the reference of Mittal is now introduced to teach this as:
the audio decoder, wherein the cross-processor is configured to use a frequency-time transform additionally performing a downsampling from the first sampling rate to the second sampling rate using selecting a low band portion of the decoded spectral representation with a reduced transform size to obtain the initialization data of the second decoding processor (Mittal: FIG. 6, [0062]–[0063] — down-sampling of the audio (from a first sampling rate to a second sampling rate, down-sampling also resulting in reduced sample or transform size), leading to the use of a low-pass filter (to obtain a low-band portion of the frequency domain representation with reduced transform size); [0026]–[0033] — the values of the down-sampling filter are used at the speech encoder for initialisation; [0052] — down-sampling from 32 kHz and 48 kHz sampling to 12.8 kHz (from a first sampling rate to a second sampling rate)).
The same motivation for combination which introduced the Mittal reference as applied to claim 12 above is applicable here still.
Claim 7 is rejected under 35 U.S.C. 103 as being unpatentable over Makinen (US 2005/0256701 A1) as applied to claim 1 in view of Ehara (US 2005/0060143 A1).
For claim 7, claim 1 is incorporated and the reference of Makinen provides teaching for performing cross-processing of a time domain and frequency domain encoding, and also for
the audio encoder, wherein the cross-processor comprises:
a spectral decoder configured for calculating a decoded version of the first encoded signal portion (Makinen: [0070] — performing TCX decoding (which is a spectral decoding)).
The reference of Makinen however differs from the claimed invention in that the claimed invention further provides teaching for filtering a decoded signal and feeding that to an innovative codebook.
This isn’t new to the art as the reference of Ehara is now introduced to teach this as:
the audio encoder, wherein the cross-processor comprises:
a weighted prediction coefficient analysis filtering block configured for filtering the decoded version of the first encoded signal portion to obtain a filter output and for feeding the filter output into an innovative codebook determiner of the second encoding processor for initialization (Ehara: [0085] — a target vector being obtained as filtering of the signal being input to it at Part 903 of FIG 12; [0080], FIG. 12 Parts 912 → 903 → 907 — a local decoding section that generates a decoded speech signal at 912 which then gets filtered at 903 and gets sent to the fixed codebook at 907).
Hence, before the effective filing date of the claimed invention, one of ordinary skill in the art would have found it obvious to improve upon the teaching of Makinen which provides a cross-processing of both time-domain and frequency-domain encoding including a spectral decoder, by making use of the known technique of Ehara which passes a filtered decoded signal through to a fixed codebook for initialisation, to thereby come up with the claimed invention. The combination of both prior art elements would have provided the predictable result of improving the efficiency and quality of the encoding process, through filtering out undesired spectral components and applying the codebook to represent the important signal components more accurately. See KSR Int’l Co. v. Teleflex Inc., 550 U.S. 398, 415-421, 82 USPQ2d 1385, 1395-97 (2007).
Claim 10 is rejected under 35 U.S.C. 103 as being unpatentable over Makinen (US 2005/0256701 A1) as applied to claim 1 in view of Reza (US 2013/0138398 A1).
For claim 10, claim 1 is incorporated and the reference of Makinen provides teaching for encoding a first audio signal portion. It however differs from the claimed invention in that the claimed invention further provides teaching for the first audio signal portion having its sampling frequency, the maximum being lower than or equal to half the sampling frequency, and at least one quarter the sampling frequency or higher.
The reference of Reza is now introduced to teach this as:
the audio encoder, wherein the first audio signal portion having associated therewith a sampling frequency, and wherein the maximum frequency is lower than or equal to half of the sampling frequency and at least one quarter of the sampling frequency or higher (Reza: [0180] — a maximum frequency being equal to half the sampling frequency, and after resampling, equal to a quarter of the sampling frequency).
The reference of Makinen provides teaching for encoding a first audio signal portion in a frequency domain, but differs from the claimed invention in that the claimed invention further provides teaching for the maximum frequency being lower than half the sampling frequency, and at least a quarter of the sampling frequency. This isn’t new to the art as the reference of Reza provides above.
Hence, before the effective filing date of the claimed invention, one of ordinary skill in the art would have found it obvious to combine the known teaching of Reza which provides the maximum frequency being lower than half the sampling frequency, and at least a quarter of the sampling frequency, with that of Makinen which provides an audio encoder that obtains first audio signal portions for encoding, given the predictable result of having an audio signal with a maximum frequency satisfying the confines of the Nyquist sampling theorem. See KSR Int’l Co. v. Teleflex Inc., 550 U.S. 398, 415-421, 82 USPQ2d 1385, 1395-97 (2007).
Claim 13 is rejected under 35 U.S.C. 103 as being unpatentable over Makinen (US 2005/0256701 A1) as applied to claim 1 in view of Reza (US 2013/0138398 A1) further in view of Mittal (US 2013/0030798 A1).
For claim 13, claim 1 is incorporated and the reference of Makinen discloses the audio encoder, wherein the first audio signal portion has associated therewith a first sampling rate, wherein the first encoding processor comprises:
a time-frequency converter configured for converting the first audio signal portion into a frequency domain representation comprising spectral lines [[up to a maximum frequency of the first audio signal portion, wherein the maximum frequency is lower than or equal to half of the first sampling rate and at least one quarter of the first sampling rate or higher]] (Makinen: [0043] — the mentioning of transform coding (TCX) indicates an inherent time to frequency converter of an input signal into a frequency domain representation; [0069] — performing a fast Fourier transform (which would generate frequency representations that have spectral lines)).
The reference of Makinen however fails to teach the further limitations of this claim regarding the first audio signal portion having its sampling frequency, the maximum being lower than or equal to half the sampling frequency, and at least one quarter the sampling frequency or higher.
This is however not new to the art as the reference of Reza is now introduced to teach this as:
the audio encoder, wherein the first audio signal portion has associated therewith a first sampling rate (Reza: [0180] — a signal being associated with a sampling frequency), wherein the first encoding processor comprises:
a time-frequency converter configured for converting the first audio signal portion into a frequency domain representation comprising spectral lines up to a maximum frequency of the first audio signal portion, wherein the maximum frequency is lower than or equal to half of the first sampling rate and at least one quarter of the first sampling rate or higher (Reza: [0180] — a maximum frequency being equal to half the sampling frequency, and after resampling, equal to a quarter of the sampling frequency).
The same motivation for combination incorporating the reference of Reza as applied to claim 10 above is applicable here still.
The combination of Makinen in view of Reza provides teaching for a spectral encoder for encoding a frequency domain representation (Makinen: [0039] — applying TCX model for encoding). This combination however fails to teach precisely that a second sampling rate of a second encoding processor is lower than the first sampling rate.
The reference of Mittal is introduced to teach this as:
a spectral encoder configured for encoding the frequency domain representation, and wherein a second sampling rate of the second encoding processor is lower than the first sampling rate (Mittal: FIG. 6, [0062]–[0063] — down-sampling of the audio (from a first sampling rate to a second sampling rate, down-sampling also resulting in reduced sample or transform size), leading to the use of a low-pass filter (to obtain a low-band portion of the frequency domain representation with reduced transform size); [0026]–[0033] — the values of the down-sampling filter are used at the speech encoder for initialisation; [0052] — down-sampling from 32 kHz and 48 kHz sampling to 12.8 kHz (from a first sampling rate to a second sampling rate)).
The same motivation for combination which introduced the Mittal reference as applied to claim 12 above is applicable here still.
Claim 15 is rejected under 35 U.S.C. 103 as being unpatentable over Makinen (US 2005/0256701 A1) as applied to claim 14 in view of Liu et al. (US 2013/0173275 A1: hereafter — Liu).
For claim 154, claim 14 is incorporated and the reference of Makinen provides teaching for obtaining a decoded spectral representation of a first encoded audio signal. This reference however fails to teach of the decoded spectral representation extending until a maximum frequency of a time representation of the decoded audio signal with a zero-spectral value for the frequency.
This however isn’t new to the art as the reference of Liu is now introduced to teach this as:
the audio decoder, wherein the decoded spectral representation extends until a maximum frequency of a time representation of the decoded audio signal, a spectral value for the maximum frequency being zero or different from zero (Liu: [0047] — by (Equation 2), the decoded spectral representations ranges in summation up to the maximum frequency sb_end[i], and having spectral coefficient value equal to zero).
The reference of Makinen provides teaching for obtaining a decoded spectral representation of a first encoded audio signal, but differs from the claimed invention in that the claimed invention further provides teaching for extending the decoded spectral representation until a value of the maximum frequency of the decoded audio signal. This isn’t new to the art as the reference of Liu goes to show above.
Hence, before the effective filing date of the claimed invention, one of ordinary skill in the art would have found it obvious to combine the known teaching of Liu which provides extending the decoded spectral representation until a value of the maximum frequency of the decoded audio signal, with the teaching of the reference of Makinen which provides the generation of a decoded spectral representation of an encoded audio signal, to thereby come up with the claimed invention. The combination of both prior art elements would have provided the predictable result of reconstructing each subband of the decoded signal to extend their spectral representations up to the maximum frequency to ensure that the decoded signal gets reconstructed up to its appropriate energy and magnitude levels. See KSR Int’l Co. v. Teleflex Inc., 550 U.S. 398, 415-421, 82 USPQ2d 1385, 1395-97 (2007).
Conclusion
The prior art made of record and not relied upon is considered pertinent to Applicant’s disclosure.
GOLDSTEIN et al. (WO 2010/030889 A1) provides teaching for adjusting the sampling rate of a digital audio signal to an internal sampling rate, and then having a feature extractor extract prediction coefficients from the resampled signal [0062]– [0063].
Ramo et al. (US 2014/0330415 A1) provides teaching for digitally sampling an audio frame at a sampling rate and then obtaining prediction coefficients from it [0018].
Geiger et al. (US 2012/0253797 A1) provides teaching for filtering audio content, with the result of filtering of the current frame being filtered through a linear prediction analysis filter in the form of a weighting filter depending on linear prediction coefficients [0167].
Moriya et al. (US 2013/0317814 A1) provides teaching for an encoder that includes a linear prediction analysis unit, a perceptual weighting filter as well as a fixed codebook search unit ([0029], FIG. 1).
Any inquiry concerning this communication or earlier communications from the Examiner should be directed to OLUWADAMILOLA M. OGUNBIYI whose telephone number is (571)272-4708. The Examiner can normally be reached Monday – Thursday (8:00 AM – 5:30 PM Eastern Standard Time).
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If attempts to reach the Examiner by telephone are unsuccessful, the Examiner’s Supervisor, PARAS D. SHAH can be reached at (571) 270-1650. The fax phone number for the organization where this application or proceeding is assigned is 571-273-8300.
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/OLUWADAMILOLA M OGUNBIYI/Examiner, Art Unit 2653
1 The Examiner addresses this claim as being directed to an encoder, not a decoder, as established in the objection section above.
2 The Examiner addresses this claim as being directed to an encoder, not a decoder, as established in the objection section above.
3 By the claims listing, claim 16 is presented as being dependent on claim 8. This claim has been objected to as presented in the Objections section. For the purpose of this Office Action, the Examiner considers this claim as being dependent on claim 14 instead of claim 8.
4 By the claims listing, claim 15 is presence as being dependent on claim 8. This claim has been objected to as presented in the Objections section. For the purpose of this Office Action, the Examiner considers this claim as being dependent on claim 14 instead of claim 8.
Prosecution Insights