DETAILED ACTION
Introduction
This office action is in response to Applicant’s submission filed on December 12, 2024.
Claims 1-20 are pending in the application. As such, claims 1-20 have been examined.
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 .
Drawings
The drawings were received on December 12, 2024. These drawings have been accepted and considered by the Examiner.
Claim Rejections - 35 USC § 101
35 U.S.C. 101 reads as follows:
Whoever invents or discovers any new and useful process, machine, manufacture, or composition of matter, or any new and useful improvement thereof, may obtain a patent therefor, subject to the conditions and requirements of this title.
Claim 20 is rejected under 35 U.S.C. 101 because the claimed invention is directed to non-statutory subject matter. The claim(s) does/do not fall within at least one of the four categories of patent eligible subject matter.
Claim 20 is drawn to a "computer program product" which is not recited, nor clarified, in the specification and as such is non-statutory subject matter. See MPEP § 2106.1V.B.1.a. Software not claimed as embodied in computer readable media are descriptive material per se and are not statutory because they are not capable of causing functional change in the computer. See, e.g., Warmerdam, 33 F.3d at 1361, 31 USPQ2d at 1760 (claim to a data structure per se held nonstatutory). Such claimed data structures do not define any structural and functional interrelationships between the data structure and other claimed aspects of the invention, which permit the data structure's functionality to be realized. In contrast, a claimed computer readable medium encoded with a data structure defines structural and functional interrelationships between the data structure and the computer software and hardware components which permit the data structure's functionality to be realized, and is thus statutory. Similarly, computer programs claimed as computer listings per se, i.e., the descriptions or expressions of the programs are not physical "things." They are neither computer components nonstatutory processes, as they are not "acts" being performed. Such claimed computer programs do not define any structural and functional interrelationships between the computer program and other claimed elements of a computer, which permit the computer program's functionality to be realized.
Claim Rejections - 35 USC § 103
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 35 U.S.C. 103 which forms the basis for all obviousness rejections set forth in this Office action:
A patent for a claimed invention may not be obtained, notwithstanding that the claimed invention is not identically disclosed as set forth in section 102, if the differences between the claimed invention and the prior art are such that the claimed invention as a whole would have been obvious before the effective filing date of the claimed invention to a person having ordinary skill in the art to which the claimed invention pertains. Patentability shall not be negated by the manner in which the invention was made.
Claims 1-2 and 19-20 are rejected under 35 U.S.C. 103 as being unpatentable over Cho (US Patent Pub. No. 20080130914 A1), in view of Gunawan et al. (US Patent No. 10923132 B2), hereinafter Gunawan.
Regarding claim 1, Cho teaches a method of processing audio (Cho in [0054] teaches suppressing noise in audio signals),
the method comprising:
receiving, from a plurality of microphones, an input audio signal (Cho in [0056] teaches using an array of microphones to receive the audio input);
identifying an angle of arrival associated with the input audio signal (Cho in [0012] teaches estimating the direction of arrival (DOA) of the audio signals by a spatial spectrum derived from directly using the signal subspace, and preparing a weighting vector based on the DOA, and obtaining noise reduced audio signals using the weighting vector, and outputting the noise reduced audio signals);
determining a plurality of gains corresponding to a [plurality of bands] of the input audio signal based on a combination of at least: 1) a representation of a covariance of signals associated with microphones of the plurality of microphones on [a per-band basis]; and 2) the angle of arrival (Cho in [0062] teaches using a covariance matrix and using weighting vector to give more weight to analog signals, or maximize gain of analog signals, at incident angles adjacent to the DOA, and to give less weight to analog signals, or minimize gain of analog signals, at incident angles away from the DOA);
and
applying the plurality of gains to the [plurality of bands] of the input audio signal such that at least a portion of the input audio signal is suppressed to form an enhanced audio signal (Cho in [0011] teaches using a transformation unit for converting the digital signals in time domain into digital signals in frequency domain; a noise suppression unit for suppressing noise in the digital signals in frequency domain by multiplying a weighting vector to the digital signals in frequency domain, thereby obtaining noise reduced digital signals in frequency domain; and an inverse transformation unit for converting the noise reduced digital signals in frequency domain into noise reduced digital signals in time domain).
Cho does not teach, however Gunawan teaches
determining a plurality of gains corresponding to a plurality of bands of the input audio signal based on a combination of at least: 1) a representation of a covariance of signals associated with microphones of the plurality of microphones on a per-band basis; and 2) the [angle of arrival] (Gunawan in [col 6 ln 55-65] teaches adjusting the gain of each sub band, and in [col 6 ln 39-54] teaches calculating covariance matrices for each of the bands where resulting measurement is a value between 0 and 1, with a value of 1 representing that the sound signal is from a purely diffuse sound source, and a value of 0 indicating that the sound is from a purely directional sound source).
Gunawan is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho further in view of Gunawan to allow for adjusting the gain of each sub band. Motivation to do so would allow for suppressing the diffuse background sound in order to improve intelligibility of the foreground noise source (Gunawan [col 7 ln 49-58]).
Regarding claim 2, Cho, as modified above, teaches the method of claim 1.
Cho, as modified above, teaches the angle of arrival, and the microphones.
Cho, as modified above, does not teach, however Gunawan teaches
wherein identifying the angle of arrival comprises
converting the signals received associated with microphones of the plurality of microphones to a spatial representation (Gunawan in [col 4 ln 32-37] teaches the data collected from the audio scene contains spatial information, such as the direction of the sound source should the signal be directional, or the diffusivity of the sound signal, if the sound is diffuse),
and
wherein the input audio signal corresponds to the spatial representation (Gunawan in [col 4 ln 32-37] teaches the data collected from the audio scene contains spatial information, such as the direction of the sound source should the signal be directional, or the diffusivity of the sound signal, if the sound is diffuse).
Gunawan is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Gunawan to allow for using spatial information. Motivation to do so would allow for suppressing the diffuse background sound in order to improve intelligibility of the foreground noise source (Gunawan [col 7 ln 49-58]).
Regarding claim 19, Cho, as modified above, teaches the method of claim 1.
Cho further teaches
A system including one or more processors (Cho in [0064] teaches using a processor which performs operations)
configured to perform operations of claim 1 (Cho in [0064] teaches using a processor which performs operations).
Regarding claim 20, Cho, as modified above, teaches the method of claim 1.
Cho further teaches
configured to cause one or more processors to perform operations of claim 1 (Cho in [0064] teaches using a processor which performs operations).
Cho, as modified above, does not teach, however Gunawan teaches
A computer program product (Gunawan in [col 9 ln 15-52] teaches using a computer program product).
Gunawan is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Gunawan to allow for using spatial information. Motivation to do so would allow for suppressing the diffuse background sound in order to improve intelligibility of the foreground noise source (Gunawan [col 7 ln 49-58]).
Claims 3-6 and 8 are rejected under 35 U.S.C. 103 as being unpatentable over Cho, in view of Gunawan, in view of Bondalapati et al. (US Patent Pub. No. 20220205669 A1), hereinafter Bondalapati.
Regarding claim 3, Cho, as modified above, teaches the method of claim 1.
Cho, as modified above, teaches the plurality of gains.
Cho, as modified above, does not teach, however Bondalapati teaches
wherein determining the plurality of gains comprises:
identifying one or more objects of the input audio signal (Bondalapati in [0093] teaches identifying the voices in the audio signal);
and
clustering the one or more objects of the input audio signal as being within one of a plurality of clusters (Bondalapati in [0093] teaches computing an audio signature for each audio signal and determine a direction of arrival for each audio signal, and then populates entries in the voice data log for the audio signal, and then identifying one or more clusters for the entries based on the voices identified in the audio signal and a direction of arrival that is associated with each voice),
wherein [the plurality of gains] associated with a current time frame of the input audio signal are determined based on a proximity of the current time frame of the input audio signal to objects within the clustering of the one or more objects (Bondalapati in [0093] teaches recording sound samples over a period of time, and computing an audio signature for each audio signal and determine a direction of arrival for each audio signal, and then populates entries in the voice data log for the audio signal, and then identifying one or more clusters for the entries based on the voices identified in the audio signal and a direction of arrival that is associated with each voice).
Bondalapati is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Bondalapati to allow for using clustering of voices based on DoA. Motivation to do so would allow for a tracking system to more efficiently manage and operate another system based on the number of people that are present within a space (Bondalapati [0003]).
Regarding claim 4, Cho, as modified above, teaches the method of claim 3.
Cho, as modified above, teaches the one or more objects.
Cho, as modified above, does not teach, however Bondalapati teaches
wherein identifying the one or more objects of the input audio signal is based on a current input and a historical input (Bondalapati in [0093] teaches recording sound samples over a period of time, and computing an audio signature for each audio signal and determine a direction of arrival for each audio signal, and then populates entries in the voice data log for the audio signal, and then identifying one or more clusters for the entries based on the voices identified in the audio signal and a direction of arrival that is associated with each voice, and populates entries in the voice data log).
Bondalapati is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Bondalapati to allow for using clustering of voices based on DoA and history. Motivation to do so would allow for a tracking system to more efficiently manage and operate another system based on the number of people that are present within a space (Bondalapati [0003]).
Regarding claim 5, Cho, as modified above, teaches the method of claim 3.
Cho, as modified above, teaches the one or more objects.
Cho, as modified above, does not teach, however Bondalapati teaches
wherein clustering the one or more objects of the input audio signal is responsive to determining the one or more audio objects have been present for more than a threshold number of frames of the input audio signal (Bondalapati in [0068] teaches determining whether the presence value meets a presence threshold value, amount of time).
Bondalapati is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Bondalapati to allow for using clustering of voices based on DoA and a time threshold. Motivation to do so would allow for a tracking system to more efficiently manage and operate another system based on the number of people that are present within a space (Bondalapati [0003]).
Regarding claim 6, Cho, as modified above, teaches the method of claim 3.
Cho, as modified above, teaches the one or more objects.
Cho, as modified above, does not teach, however Bondalapati teaches
wherein clustering a given object of the one or more objects of the input audio signal comprises one of: 1) updating an existing object in a cluster; 2) creating a new object in the cluster corresponding to the given object; or 3) replacing the existing object in the cluster with the given object (Bondalapati in [0053] teaches when multiple clusters share the same direction of arrival location, this indicates that these clusters share the same sound source. Typically, when the same sound source generates multiple voices this indicates that the sound source in an electronic device (e.g. a television or radio) and that the sound source is not a person. Here, the system identifies the clusters that share a common sound source and removes the clusters from consideration. This process allows the system to filter out the electronic devices which are not actual people that are present within the space).
Bondalapati is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Bondalapati to allow for updating the clustering. Motivation to do so would allow for a tracking system to more efficiently manage and operate another system based on the number of people that are present within a space (Bondalapati [0003]).
Regarding claim 8, Cho, as modified above, teaches the method of claim 3.
Cho, as modified above, teaches the clustering, and the plurality of bands.
Cho, as modified above, does not teach, however Gunawan teaches
wherein [the clustering] is on a broadband basis with respect to the plurality of bands (Gunawan in [col 6 ln 55-65] teaches using a broadband diffusivity measure).
Gunawan is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Gunawan to allow for using broadband information. Motivation to do so would allow for suppressing the diffuse background sound in order to improve intelligibility of the foreground noise source (Gunawan [col 7 ln 49-58]).
Claim 7 is rejected under 35 U.S.C. 103 as being unpatentable over Cho, in view of Gunawan, in view of Bondalapati, in view of Marash et al. (US Patent Pub. No. 20130136273 A1), hereinafter Marash.
Regarding claim 7, Cho, as modified above, teaches the method of claim 6.
Cho, as modified above, teaches the clustering.
Cho, as modified above, does not teach, however Marash teaches
wherein the existing object that is replaced is the existing object with a lowest activity level of the cluster (Marash in [0033] teaches selecting for inclusion in a cluster the signals with the highest power level [here lowest activity level maps to the items not selected (replacement) for inclusion in the cluster]).
Marash is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Marash to allow for not using signals with the lower power level. Motivation to do so would allow for selecting from a plurality of microphone clusters distributed in a room a microphone cluster that senses the smallest amount of reverberation and for estimating the reverberation level sensed at a microphone cluster at a given time in an economical and practical manner for real-time dynamic monitoring (Marash [0016]).
Claims 9-10 and 12 are rejected under 35 U.S.C. 103 as being unpatentable over Cho, in view of Gunawan, in view of Bondalapati, in view of Wijetillake et al. (US Patent Pub. No. 20220080198 A1), hereinafter Wijetillake.
Regarding claim 9, Cho, as modified above, teaches the method of claim 3.
Cho, as modified above, teaches the clustering, and the one or more objects.
Cho, as modified above, does not teach, however Wijetillake teaches
wherein clustering the one or more objects comprises determining a plurality of similarity metrics of the input audio signal to each cluster (Wijetillake in [0052] teaches processing the audio signal into a plurality of band limited audio signals, and using a mapping unit configured to map each of coding parameter group of the sequence of coding parameter groups to each of the plurality of band limited audio signals based on an acoustic parameter of the multiple acoustic parameters).
Wijetillake is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Wijetillake to allow for using multiple acoustic parameters. Motivation to do so would allow for a cochlear implant system that improves the patient's ability to perceive perceptual information coded into stimulation pulses provided by the cochlear implant system to the auditory nerve fibers of the patient (Wijetillake [0008]).
Regarding claim 10, Cho, as modified above, teaches the method of claim 9.
Cho, as modified above, teaches the clustering, and the one or more objects.
Cho, as modified above, does not teach, however Wijetillake teaches
wherein the plurality of similarity metrics correspond to the plurality of bands (Wijetillake in [0052] teaches processing the audio signal into a plurality of band limited audio signals, and using a mapping unit configured to map each of coding parameter group of the sequence of coding parameter groups to each of the plurality of band limited audio signals based on an acoustic parameter of the multiple acoustic parameters).
Wijetillake is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Wijetillake to allow for using multiple acoustic parameters. Motivation to do so would allow for a cochlear implant system that improves the patient's ability to perceive perceptual information coded into stimulation pulses provided by the cochlear implant system to the auditory nerve fibers of the patient (Wijetillake [0008]).
Regarding claim 12, Cho, as modified above, teaches the method of claim 9.
Cho, as modified above, teaches the plurality of gains.
Cho, as modified above, does not teach, however Wijetillake teaches
wherein the [plurality of gains] are determined using the plurality of similarity metrics (Wijetillake in [0052] teaches processing the audio signal into a plurality of band limited audio signals, and using a mapping unit configured to map each of coding parameter group of the sequence of coding parameter groups to each of the plurality of band limited audio signals based on an acoustic parameter of the multiple acoustic parameters).
Wijetillake is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Wijetillake to allow for using multiple acoustic parameters. Motivation to do so would allow for a cochlear implant system that improves the patient's ability to perceive perceptual information coded into stimulation pulses provided by the cochlear implant system to the auditory nerve fibers of the patient (Wijetillake [0008]).
Claim 11 is rejected under 35 U.S.C. 103 as being unpatentable over Cho, in view of Gunawan, in view of Bondalapati, in view of Wijetillake, in view of Marash.
Regarding claim 11, Cho, as modified above, teaches the method of claim 9.
Cho, as modified above, teaches the clustering.
Cho, as modified above, does not teach, however Marash teaches
wherein determining a similarity metric for a given cluster is based on a most active object within the given cluster (Marash in [0033] teaches selecting for inclusion in a cluster the signals with the highest power level [here most active maps to highest power]).
Marash is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Marash to allow for using signals with the highest power level. Motivation to do so would allow for selecting from a plurality of microphone clusters distributed in a room a microphone cluster that senses the smallest amount of reverberation and for estimating the reverberation level sensed at a microphone cluster at a given time in an economical and practical manner for real-time dynamic monitoring (Marash [0016]).
Claim 13 is rejected under 35 U.S.C. 103 as being unpatentable over Cho, in view of Gunawan, in view of Bondalapati, in view of Hui et al. (US Patent Pub. No. 20140140552 A1), hereinafter Hui.
Regarding claim 13, Cho, as modified above, teaches the method of claim 3.
Cho, as modified above, teaches the plurality of clusters.
Cho, as modified above, does not teach, however Hui teaches
wherein the plurality of clusters comprise a within a region of interest cluster and an outside of the region of interest cluster (Hui in [0006] teaches components can further comprise a permission component that permits the acoustic component to receive a first audio signal determined to originate within a beam forming region and prevents the acoustic component from reception of a second audio signal determined to originate outside the beam forming region).
Hui is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Hui to allow for distinguishing audio signals from within vs without a region. Motivation to do so would allow for enhancing particular signals such as warning signals (Hui [0002]).
Claim 14 is rejected under 35 U.S.C. 103 as being unpatentable over Cho, in view of Gunawan, in view of Bondalapati, in view of Hui, in view of Sun (US Patent Pub. No. 20120045069 A1).
Regarding claim 14, Cho, as modified above, teaches the method of claim 13.
Cho, as modified above, teaches the plurality of bands, the plurality of gains the audio signal inside the region of interest, and the audio outside e region of interest.
Cho, as modified above, does not teach, however Sun teaches
further comprising
determining, for each band of the plurality of bands,
a lower bound gain applicable to a portion of the input audio signal [inside the region of interest] (Sun in [0014] teaches a method of enhancing an audio signal comprising the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component at that respective frequency; d) estimating a set of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequency-specific lower bound gain and respective frequency-specific upper bound gain; and e) applying the frequency-specific gains to the primary audio input signal)
and
an upper bound gain applicable to a portion of the input audio [outside e region of interest] (Sun in [0014] teaches a method of enhancing an audio signal comprising the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component at that respective frequency; d) estimating a set of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequency-specific lower bound gain and respective frequency-specific upper bound gain; and e) applying the frequency-specific gains to the primary audio input signal),
wherein [the plurality of gains] are subject to the lower bound gain and the upper bound gain (Sun in [0014] teaches a method of enhancing an audio signal comprising the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component at that respective frequency; d) estimating a set of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequency-specific lower bound gain and respective frequency-specific upper bound gain; and e) applying the frequency-specific gains to the primary audio input signal).
Sun is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Sun to allow for using upper and lower bound gains. Motivation to do so would allow for a dynamic frequency dependent audibility enhancement system with no calibration or divergence of adaptive filter algorithms due to user speech, which takes into account psychoacoustic effects so that a user is able to hear an audio signal as intended without all environmental noise being totally drowned out (Sun [0013]).
Claim 15 is rejected under 35 U.S.C. 103 as being unpatentable over Cho, in view of Gunawan, in view of Sun, in view of Ota et al. (US Patent Pub. No. 20100273522 A1), hereinafter Ota.
Regarding claim 15, Cho, as modified above, teaches the method of claim 1.
Cho, as modified above, teaches the input audio signal, the plurality of bands, and the plurality of gains.
Cho, as modified above, does not teach, however Gunawan teaches
grouping the input audio signal and the [filtered signal] into the plurality of bands (Gunawan in [col 6 ln 55-65] teaches adjusting the gain of each sub band, and in [col 6 ln 39-54] teaches calculating covariance matrices for each of the bands where resulting measurement is a value between 0 and 1, with a value of 1 representing that the sound signal is from a purely diffuse sound source, and a value of 0 indicating that the sound is from a purely directional sound source).
Gunawan is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Gunawan to allow for using a plurality of bands. Motivation to do so would allow for suppressing the diffuse background sound in order to improve intelligibility of the foreground noise source (Gunawan [col 7 ln 49-58]).
Cho, as modified above, does not teach, however Sun teaches
wherein applying the plurality of gains comprises:
determining a plurality of gain bounds (Sun in [0014] teaches a method of enhancing an audio signal comprising the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component at that respective frequency; d) estimating a set of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequency-specific lower bound gain and respective frequency-specific upper bound gain; and e) applying the frequency-specific gains to the primary audio input signal);
clamping the gains to the gain bounds (Sun in [0014] teaches a method of enhancing an audio signal comprising the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component at that respective frequency; d) estimating a set of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequency-specific lower bound gain and respective frequency-specific upper bound gain; and e) applying the frequency-specific gains to the primary audio input signal);
and
applying the clamped gains to the input audio signal (Sun in [0014] teaches a method of enhancing an audio signal comprising the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component at that respective frequency; d) estimating a set of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequency-specific lower bound gain and respective frequency-specific upper bound gain; and e) applying the frequency-specific gains to the primary audio input signal).
Sun is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Sun to allow for using upper and lower bound gains. Motivation to do so would allow for a dynamic frequency dependent audibility enhancement system with no calibration or divergence of adaptive filter algorithms due to user speech, which takes into account psychoacoustic effects so that a user is able to hear an audio signal as intended without all environmental noise being totally drowned out (Sun [0013]).
Cho, as modified above, does not teach, however Ota teaches
utilizing a linear filter to filter the input audio signal to generate a filtered signal (Ota in [0037] teaches using a linear prediction filter to filter a signal);
calculating the [plurality of gains] for the [plurality of bands] by taking a difference between a power of the input audio signal and the filtered signal (Ota in [0037] teaches determining a power ratio between a filtered and an unfiltered signal).
Ota is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Ota to allow for using power ratio between a filtered and an unfiltered signal. Motivation to do so would allow for developing a cellular phone in which autonomous control is performed depending on estimated surroundings of a user to improve usability for the user (Ota [0007]).
Claim 16 is rejected under 35 U.S.C. 103 as being unpatentable over Cho, in view of Gunawan, in view of Schreiber (US Patent No. 5311543 A).
Regarding claim 16, Cho, as modified above, teaches the method of claim 1.
Cho, as modified above, teaches the plurality of gains, the input audio signal, and the spatial components.
Cho, as modified above, does not teach, however Schreiber teaches
wherein applying the plurality of gains comprises:
determining a ratio of [spatial components] of the [input audio signal] (Schreiber in [col 1 ln 10-30] teaches using a SNR which refers to the signal-to-noise ratio of input and output signals);
and
applying the [plurality of gains] based at least in part on the ratio of the [spatial components] (Schreiber in [col 1 ln 10-30] teaches using a SNR which refers to the signal-to-noise ratio of input and output signals).
Schreiber is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Schreiber to allow for using a SNR which refers to the signal-to-noise ratio of input and output signals. Motivation to do so would allow for reducing the effect of multipath distortion in the transmission channel without the use of conventional automatic channel equalizers, and it permits the establishment of a series of graded thresholds of CNR, so that, at each higher level of CNR, improved quality is obtained (Schreiber [col 4 ln 25-44]).
Claims 17 and 18 are rejected under 35 U.S.C. 103 as being unpatentable over Cho, in view of Gunawan, in view of Shin et al. (US Patent Pub. No. 20120263317 A1), hereinafter Shin.
Regarding claim 17, Cho, as modified above, teaches the method of claim 1.
Cho, as modified above, teaches the plurality of gains, and the plurality of bands.
Cho, as modified above, does not teach, however Shin teaches
further comprising smoothing the plurality of gains prior to applying the plurality of gains (Shin in [0180] teaches using a gain smoothing factor on all subbands of an audio signal).
Shin is considered to be analogous to the claimed invention because it is in the same field of audio signal processing. Therefore, it would have been obvious to someone of ordinary skill in the art before the effective filing date of the claimed invention to have modified Cho, as modified above, further in view of Shin to allow for using a gain smoothing factor. Motivation to do so would allow for the sub-band gain factors G(i) to decrease relatively quickly (e.g., to a default value of reinforcement factor R(i), such as 1.0) (Shin [0180]).
Regarding claim 18, Cho, as modified above, teaches the method of claim 17.
Cho, as modified above, teaches the enhanced audio signal.
Cho further teaches
further comprising causing the enhanced audio signal to be presented via a loudspeaker or headphones (Cho in [0066] teaches the output unit includes a speaker).
Conclusion
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PAUL MUELLER
Examiner
Art Unit 2657
/PAUL J. MUELLER/Examiner, Art Unit 2657
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