DETAILED ACTION
Notice of Pre-AIA or AIA Status
The present application, filed on or after March 16, 2013, is being examined under the first inventor to file provisions of the AIA .
Drawings
The drawings are objected to as failing to comply with 37 CFR 1.84(p)(5) because they include the following reference characters not mentioned in the description:
“614” in Figure 9
“202D” in Figure 10.
Corrected drawing sheets in compliance with 37 CFR 1.121(d), or amendment to the specification to add the reference character(s) in the description in compliance with 37 CFR 1.121(b) are required in reply to the Office action to avoid abandonment of the application. Any amended replacement drawing sheet should include all of the figures appearing on the immediate prior version of the sheet, even if only one figure is being amended. Each drawing sheet submitted after the filing date of an application must be labeled in the top margin as either “Replacement Sheet” or “New Sheet” pursuant to 37 CFR 1.121(d). If the changes are not accepted by the examiner, the applicant will be notified and informed of any required corrective action in the next Office action. The objection to the drawings will not be held in abeyance.
Specification
The disclosure is objected to because of the following informalities:
On page 10, line 2, “algorithms” should read “algorithms.”.
On page 10, lines 16-27, “should adjusted to operate” should read “should be adjusted to operate”.
On page 20, line 22, “different instance provide different modes” should read “different instances provide different modes”.
On page 22, line 25, “apply a window function the frames” should read “apply a window function to the frames”.
Appropriate correction is required.
Claim Rejections - 35 USC § 112
The following is a quotation of 35 U.S.C. 112(b):
(b) CONCLUSION.—The specification shall conclude with one or more claims particularly pointing out and distinctly claiming the subject matter which the inventor or a joint inventor regards as the invention.
The following is a quotation of 35 U.S.C. 112 (pre-AIA ), second paragraph:
The specification shall conclude with one or more claims particularly pointing out and distinctly claiming the subject matter which the applicant regards as his invention.
Claims 4, 5 and 15 are rejected under 35 U.S.C. 112(b) or 35 U.S.C. 112 (pre-AIA ), second paragraph, as being indefinite for failing to particularly point out and distinctly claim the subject matter which the inventor or a joint inventor (or for applications subject to pre-AIA 35 U.S.C. 112, the applicant), regards as the invention.
Claim 4 recites the limitation "adjust the speech enhancement processing to operate with smaller latency if the determined at least one quality value indicates at least one of: that the latency associated with the obtained one or more audio signals is higher; or that the noise levels in the obtained one or more audio signals is lower" in lines 2-6. This limitation is indefinite because it is not clear what "the latency associated with the obtained one or more audio signals" is compared with to determine that it is "higher" and it is not clear what "the noise levels in the obtained one or more audio signals" is compared with to determine that it is "lower". The specification recites, on page 11, lines 1-5, “The speech enhancement processing can be adjusted to operate with smaller latency if the determined quality value indicates that the latency associated with the obtained one or more audio signals is higher, or that the noise levels in the obtained one or more audio signals is lower. The latency and/or the noise levels can be determined to be higher or lower compared to static threshold.”. This rejection can be overcome by changing the limitation to "adjust the speech enhancement processing to operate with smaller latency if the determined at least one quality value indicates at least one of: that the latency associated with the obtained one or more audio signals is higher compared to a static threshold; or that the noise levels in the obtained one or more audio signals is lower compared to a static threshold".
Claim 5 recites the limitation "adjust the speech enhancement processing to operate with larger latency if the determined at least one quality value indicates at least one of: that the latency associated with the obtained one or more audio signals is lower; or that the noise levels associated with the obtained one or more audio signals is higher" in lines 2-6. This limitation is indefinite because it is not clear what "the latency associated with the obtained one or more audio signals" is compared with to determine that it is "lower" and it is not clear what "the noise levels in the obtained one or more audio signals" is compared with to determine that it is "higher". The specification recites, on page 11, lines 10-14, “The speech enhancement processing can be adjusted to operate with larger latency if the determined quality value indicates that the latency associated with the obtained one or more audio signals is lower, or that the noise levels associated with the obtained one or more audio signals is higher. The latency and/or the noise levels can be determined to be lower or higher compared to static threshold.”. This rejection can be overcome by changing the limitation to "adjust the speech enhancement processing to operate with larger latency if the determined at least one quality value indicates at least one of: that the latency associated with the obtained one or more audio signals is lower compared to a static threshold; or that the noise levels associated with the obtained one or more audio signals is higher compared to a static threshold".
Claim 15 recites the limitation "the speech enhancement processing is adjusted to operate with larger latency if the determined at least one quality value indicates at least one of: that the latency associated with the obtained one or more audio signals is lower; or that the noise levels associated with the obtained one or more audio signals is higher" in lines 1-4. This limitation is indefinite because it is not clear what "the latency associated with the obtained one or more audio signals" is compared with to determine that it is "lower" and it is not clear what "the noise levels in the obtained one or more audio signals" is compared with to determine that it is "higher". The specification recites, on page 11, lines 10-14, “The speech enhancement processing can be adjusted to operate with larger latency if the determined quality value indicates that the latency associated with the obtained one or more audio signals is lower, or that the noise levels associated with the obtained one or more audio signals is higher. The latency and/or the noise levels can be determined to be lower or higher compared to static threshold.”. This rejection can be overcome by changing the limitation to "the speech enhancement processing is adjusted to operate with larger latency if the determined at least one quality value indicates at least one of: that the latency associated with the obtained one or more audio signals is lower compared to a static threshold; or that the noise levels associated with the obtained one or more audio signals is higher compared to a static threshold".
Claim Rejections - 35 USC § 102
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.
(a)(2) the claimed invention was described in a patent issued under section 151, or in an application for patent published or deemed published under section 122(b), in which the patent or application, as the case may be, names another inventor and was effectively filed before the effective filing date of the claimed invention.
Claims 1 – 2, 4 – 7, 11, 13, 15 – 17 and 20 are rejected under 35 U.S.C. 102(a)(1) and 102(a)(2) as being anticipated by Dolson (US Patent No. 10,313,416).
Regarding claim 1, Dolson discloses an apparatus for speech enhancement processing, the apparatus comprising:
at least one processor (Column 3, lines 20-23, "Various apparatus embodiments involve implementation with circuitry, as may be carried out with a computer or other type processor, and as may be implemented separately or together in a combined circuit.");
and at least one memory storing instructions that, when executed with the at least one processor (Column 7, lines 20-28, "In certain embodiments, such illustrated items represent one or more computer circuitry (e.g., microcomputer or other CPU) which is understood to include memory circuitry that stores code (program to be executed as a set/sets of instructions) for performing a basic algorithm (e.g., noise suppression) or the more complex process/algorithm as described above for processing audio signals with the steps, functions, operations, activities, etc."), cause the apparatus at least to:
obtain one or more audio signals during audio communication (Column 3, lines 46-56, "Consistent with the above, the first, second and third circuits may be implemented in a common circuit such as a computer, and utilized with an input port for receiving an audio signal. Such circuitry may, for example, be employed with mobile telephone circuitry for communicating voice or other audio signals, such as for live conversation and/or live or recorded audio media streaming. In these or other applications, the circuitry may be employed on a transmitter, a receiver, with circuitry that processes/communicates audio signals between such a transmitter and receiver, or at a combination of two or more of such locations."; Telephone circuitry for communicating voice or other audio signals reads on obtaining one or more audio signals during audio communication.);
determine at least one quality value for at least one of the obtained one or more audio signals (Column 3, lines 27-30, "As such, one or more embodiments involve such a first circuit that assesses a quality characteristic of an audio signal exhibiting a signal quality and having time-sequenced frames."; Assessing a quality characteristic of an audio signal reads on determining at least one quality value for at least one of the obtained one or more audio signals.);
and enable adjustment of speech enhancement processing used for at least one of the one or more obtained audio signals wherein the adjustment is based, at least in part, on the at least one quality value (Column 3, lines 27-33, "As such, one or more embodiments involve such a first circuit that assesses a quality characteristic of an audio signal exhibiting a signal quality and having time-sequenced frames. The first circuit provides an output indicative of the signal quality, and a second circuit dynamically adjusts an amount of latency in the audio signal, based on the output indicative of the signal quality."; Dynamically adjusting an amount of latency in the audio signal based on the output indicative of the signal quality reads on enabling an adjustment of speech enhancement processing used for at least one of the one or more obtained audio signals wherein the adjustment is based on the at least one quality value.).
Regarding claim 2, Dolson discloses the apparatus as claimed in claim 1.
Dolson further discloses:
wherein the determined at least one quality value is based on at least one of: latency associated with the obtained one or more audio signals; noise levels in the obtained one or more audio signals; or coding/decoding bit rates associated with the obtained one or more audio signals (Column 6, lines 29-36, "In an exemplary instantiation, the dynamic degradation detector 320 can be implemented with a real-time algorithm that produces an estimate of wind-noise severity for each successive signal time-window. For example, the relative amount of low-frequency energy in the signal spectrum can be used as a measure of wind-noise intensity in the current time-window, and a threshold can be established to distinguish between a low-noise state and a high-noise state."; Measuring the intensity of wind-noise in an audio signal reads on the determining at least one quality value being based on noise levels in the obtained one or more audio signals.).
Regarding claim 4, as best understood based on the 35 U.S.C. 112(b) issues identified above, Dolson discloses the apparatus as claimed in claim 1.
Dolson further discloses:
wherein the instructions, when executed with the at least one processor, cause the apparatus to adjust the speech enhancement processing to operate with smaller latency if the determined at least one quality value indicates at least one of: that the latency associated with the obtained one or more audio signals is higher; or that the noise levels in the obtained one or more audio signals is lower (Column 4, lines 5-20, "In some embodiments, the amount of latency in an audio signal is dynamically adjusted based on an expected increase in quality of the signal gained relative to an increase in the latency. The amount of latency in the audio signal may be increased and decreased over time, in response to respective increases and decreases in quality of the signal. For instance, the amount of latency in the audio signal can be increased in response to the signal degrading beyond a predefined amount of degradation, and reduced in response to the signal recovering such that the amount of degradation of the signal is within the predefined amount. In some embodiments, a low-latency mode is effected in response to an audio signal exhibiting a high level of signal quality, and an increased-latency mode may be effected in response to the audio signal exhibiting a low level of signal quality that is lower than the high level of signal quality."; Reducing the amount of latency in the audio signal in response to the amount of degradation of the signal being within the predefined amount reads on adjusting the speech enhancement processing to operate with smaller latency if the determined at least one quality value indicates that the noise levels in the obtained one or more audio signals is lower.).
Regarding claim 5, as best understood based on the 35 U.S.C. 112(b) issues identified above, Dolson discloses the apparatus as claimed in claim 1.
Dolson further discloses:
wherein the instructions, when executed with the at least one processor, cause the apparatus to adjust the speech enhancement processing to operate with larger latency if the determined at least one quality value indicates at least one of: that the latency associated with the obtained one or more audio signals is lower; or that the noise levels associated with the obtained one or more audio signals is higher (Column 4, lines 5-20, "In some embodiments, the amount of latency in an audio signal is dynamically adjusted based on an expected increase in quality of the signal gained relative to an increase in the latency. The amount of latency in the audio signal may be increased and decreased over time, in response to respective increases and decreases in quality of the signal. For instance, the amount of latency in the audio signal can be increased in response to the signal degrading beyond a predefined amount of degradation, and reduced in response to the signal recovering such that the amount of degradation of the signal is within the predefined amount. In some embodiments, a low-latency mode is effected in response to an audio signal exhibiting a high level of signal quality, and an increased-latency mode may be effected in response to the audio signal exhibiting a low level of signal quality that is lower than the high level of signal quality."; Increasing the amount of latency in the audio signal in response to the signal degrading beyond a predefined amount of degradation reads on adjusting the speech enhancement processing to operate with larger latency if the determined at least one quality value indicates that the noise levels associated with the obtained one or more audio signals is higher.).
Regarding claim 6, Dolson discloses the apparatus as claimed in claim 1.
Dolson further discloses:
wherein the instructions, when executed with the at least one processor, cause the apparatus to adjust speech enhancement processing based on selecting at least one of a plurality of available modes for use in speech enhancement processing (Column 5, line 57 - Column 6, line 8, "FIG. 3 shows an apparatus for dynamic latency control, as may be implemented in accordance with one or more embodiments. The apparatus 300 includes a signal input circuit 310, dynamic degradation detector 320, signal enhancer 330, dynamic latency control circuit 340, and signal output circuit 350. The signal input circuit 310 receives a signal having a number of frames, with a few frames shown by way of example. The dynamic degradation detector 320 indicates when a signal is sufficiently compromised that the benefit of increased latency is worth the cost (e.g., signal latency can be tolerated to achieve signal quality improvement), and vice versa. The signal enhancer 330 switches between operating in a low-latency mode and an increased-latency mode. The dynamic latency control circuit 340 dynamically increases and/or decreases the signal latency, which can be effected without introducing a significant perceptual disturbance. This may be carried out, for example, by causing a signal output circuit 350 to inject additional frames to, or remove frames from, the signal."; Switching between operating in a low-latency mode and an increased-latency mode reads on adjusting speech enhancement processing based on selecting at least one of a plurality of available modes for use in speech enhancement processing.).
Regarding claim 7, Dolson discloses the apparatus as claimed in claim 6.
Dolson further discloses:
wherein the instructions, when executed with the at least one processor, further cause the apparatus to select a window function for performing one or more transforms of the one or more audio signals, wherein the window function is selected based, at least in part, on the selected mode (Column 6, lines 44-67, "The signal enhancer 330 may be used to alter the signal spectrum corresponding to each successive signal time-window by calculating and applying a unique gain to each frequency bin in the spectrum. This calculation can make use of information from signal spectra corresponding to time-windows both before and after the current time-window. In an exemplary instantiation, the number of look-ahead time-windows that the signal enhancer can access is allowed to toggle between a low-latency condition (e.g., one time-window of look-ahead) and an increased-latency condition (e.g., three time-windows of look-ahead), utilizing dynamic latency control as noted herein. Since wind noise tends to occur in short “bursts” of predominantly low-frequency energy, and since low-frequency energy in speech tends to be associated with sustained harmonics extending over many successive time-windows, the signal enhancer 330 can use the increased-latency condition to distinguish between the unwanted wind noise and the desired speech. For example, for the increased-latency condition, an improved gain suppression factor can be calculated for each spectral frequency band by assessing the variability of the spectral energy for the band in question over seven successive time-windows, three in the past and three in the future. An analysis may be implemented such that greater observed variability is treated with greater suppression."; Using information from signal spectra corresponding to time-windows both before and after the current time-window to calculate a gain suppression factor for each spectral frequency band reads on selecting a window function for performing one or more transforms of the one or more audio signals, and changing the number of look-ahead time-windows that the signal enhancer can access from one time-window of look-ahead for a low-latency condition to three time-windows of look-ahead for an increased-latency condition reads on the window function being selected based, at least in part, on the selected mode.).
Regarding claim 11, Dolson discloses the apparatus as claimed in claim 1.
Dolson further discloses:
wherein the speech enhancement processing comprises at least one of: speech denoising; automatic gain control; or bandwidth extension (Column 3, lines 5-19, "Various embodiments are directed toward addressing challenges involving speech enhancement algorithms for voice communication, which are tightly constrained in the amount of latency that they can introduce into a signal path. Signal degradation, such as may result from intermittent wind noise caused by turbulent airflow over a voice-input microphone, can degrade the voice communication experience. This degradation can be effectively minimized for a particular portion of audio by providing a noise-suppression algorithm with signal look-ahead information obtained by looking at future portions of audio. Dynamic adjustment of such look-ahead is facilitated in an adaptively manner such that noise-suppression is enhanced, while maintaining an overall audio listening experience that may be maintained in a generally consistent manner with low-latency constraints."; A speech enhancement algorithm for voice communication that includes a noise-suppression algorithm reads on the speech enhancement processing comprising speech denoising.).
Regarding claim 13, arguments analogous to claim 1 are applicable.
Regarding claim 15, arguments analogous to claim 5 are applicable.
Regarding claim 16, arguments analogous to claim 6 are applicable.
Regarding claim 17, arguments analogous to claim 7 are applicable.
Regarding claim 20, arguments analogous to claim 11 are applicable.
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.
Claims 3, 10 and 19 are rejected under 35 U.S.C. 103 as being unpatentable over Dolson in view of Nyayate et al. (US Patent No. 12,192,720), hereinafter Nyayate.
Regarding claim 3, Dolson discloses the apparatus as claimed in claim 1, but does not specifically disclose: wherein the instructions, when executed with the at least one processor, cause the apparatus to determine the at least one quality value using a machine learning model.
Nyayate teaches:
wherein the instructions, when executed with the at least one processor, cause the apparatus to determine the at least one quality value using a machine learning model (Column 3, line 55 - Column 4, line 2, "In at least one embodiment, an audio application 112 executing on client device 106 can cause a digital audio signal to be provided as input to an audio denoiser pipeline 114. In at least one embodiment, this input audio signal can be provided as input to a feature extractor 116 which can extract various types of features from input audio. In at least one embodiment, an output of this feature extractor can be a set of features in a format such as an audio spectrogram, or mel spectrogram. In at least one embodiment, this audio spectrogram can be provided as input to a noise model 118, such as may correspond to one or more neural networks trained to predict a presence in input audio of various types of noise. In at least one embodiment, a noise signal, or audio mask, can be output from noise model 118 and provided as input to a post-processing module 120."; A neural network trained to predict a presence in input audio of various types of noise and output a noise signal reads on determine the at least one quality value using a machine learning model.).
Nyayate is considered to be analogous to the claimed invention because it is in the same field of audio 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 Dolson to incorporate the teachings of Nyayate to use a neural network trained to predict a presence in input audio of various types of noise and output a noise signal. Doing so would allow for improving a quality of speech contained in a digital audio signal before transmitting that speech to a client device (Nyayate; Column 3, lines 44-54).
Regarding claim 10, Dolson discloses the apparatus as claimed in claim 1, but does not specifically disclose: wherein the obtained one or more audio signals comprise at least one of; one or more mono audio signals, one or more stereo audio signals; one or more multichannel audio signals; or one or more spatial audio signals.
Nyayate teaches:
wherein the obtained one or more audio signals comprise at least one of; one or more mono audio signals, one or more stereo audio signals; one or more multichannel audio signals; or one or more spatial audio signals (Column 3, lines 8-10, "In at least one embodiment, audio data can be processed to determine and remove noise using a system 100 such as that illustrated in FIG. 1."; Column 4, lines 41-46, "In at least one embodiment, a feature extractor 116 of audio denoiser pipeline 114 extracts mel frequency coefficients from a continuous audio stream. In at least one embodiment, feature extractor 116 accepts a stream of mono channel noisy audio data, such as noisy speech data, at a sampling rate such as 48 khz."; Accepting a stream of mono channel noisy audio data reads on the obtained one or more audio signals comprising one or more mono audio signals.).
Nyayate is considered to be analogous to the claimed invention because it is in the same field of audio 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 Dolson to incorporate the teachings of Nyayate to accept a stream of mono channel noisy audio data to determine and remove noise. Doing so would allow for improving a quality of speech contained in a digital audio signal before transmitting that speech to a client device (Nyayate; Column 3, lines 44-54).
Regarding claim 19, arguments analogous to claim 10 are applicable.
Claims 9 and 18 are rejected under 35 U.S.C. 103 as being unpatentable over Dolson in view of Dunne et al. (US Patent No. 7,158,572), hereinafter Dunne.
Regarding claim 9, Dolson discloses the apparatus as claimed in claim 1.
Dolson further discloses:
wherein the first speech enhancement processing and the second speech enhancement processing have different latencies (Column 4, lines 5-20, "In some embodiments, the amount of latency in an audio signal is dynamically adjusted based on an expected increase in quality of the signal gained relative to an increase in the latency. The amount of latency in the audio signal may be increased and decreased over time, in response to respective increases and decreases in quality of the signal. For instance, the amount of latency in the audio signal can be increased in response to the signal degrading beyond a predefined amount of degradation, and reduced in response to the signal recovering such that the amount of degradation of the signal is within the predefined amount. In some embodiments, a low-latency mode is effected in response to an audio signal exhibiting a high level of signal quality, and an increased-latency mode may be effected in response to the audio signal exhibiting a low level of signal quality that is lower than the high level of signal quality."; A low-latency mode and an increased-latency mode reads on the first speech enhancement processing and the second speech enhancement processing have different latencies.).
Dolson does not specifically disclose: wherein the instructions, when executed with the at least one processor, cause the apparatus to determine a first quality value for a first obtained audio signal and a second, different quality value for a second obtained audio signal; and wherein the instructions, when executed with the at least one processor, cause the apparatus to apply a first speech enhancement processing to the first obtained audio signal based, at least in part, on the first quality value and to apply a second speech enhancement processing to the second obtained audio signal based, at least in part, on the second quality value.
Dunne teaches:
wherein the instructions, when executed with the at least one processor, cause the apparatus to determine a first quality value for a first obtained audio signal and a second, different quality value for a second obtained audio signal; and wherein the instructions, when executed with the at least one processor, cause the apparatus to apply a first speech enhancement processing to the first obtained audio signal based, at least in part, on the first quality value and to apply a second speech enhancement processing to the second obtained audio signal based, at least in part, on the second quality value (Column 2, lines 20-23, "A first embodiment of the invention is useful in a communication system arranged to receive a communication signal comprising first data compressed at a compression level within a first range of compression levels and second data compressed at a compression level within a second range of compression levels, the first range of compression levels being greater than the second range of compression levels. The communication signal is transmitted on a communication channel. In such an environment, the quality of the communication signal may be enhanced by generating a first mode signal in response to the first data and by generating a second mode signal in response to the second data. The generating may be accomplished by a mode detector."; Column 2, lines 36-59, 'A first analyzer signal is generated in the event that the first data is deemed suitable for a first type of enhancement in response to the first mode signal and the decoded first data. A second analyzer signal is generated in the event that the first data is deemed suitable for a second type of enhancement in response to the first mode signal and the decoded first data. A third analyzer signal is generated in the event that the second data is deemed suitable for a third type of enhancement in response to the second mode signal and second data. A fourth analyzer signal is generated in the event that the second data is deemed suitable for a fourth type of enhancement in response to the second mode signal and second data. The analyzer signals preferably are generated with a signal analyzer. Enhanced decoded first data enhanced with the first type of enhancement is generated in response to the first analyzer signal and the decoded first data. Enhanced first data enhanced with the second type of enhancement is generated in response to the second analyzer signal and the first data. Enhanced second data enhanced with the third type of enhancement is generated in response to the third analyzer signal and the second data. Enhanced second data enhanced with the fourth type of enhancement is generated in response to the fourth analyzer signal and the second data."; The first data being deemed suitable for a first type of enhancement in response to the first mode signal and the decoded first data reads on determining a first quality value for a first obtained audio signal, the second data being deemed suitable for a third type of enhancement in response to the second mode signal and second data reads on determining a second, different quality value for a second obtained audio signal,
generating enhanced first data enhanced with the first type of enhancement in response to the first analyzer signal and the decoded first data reads on applying a first speech enhancement processing to the first obtained audio signal based, at least in part, on the first quality value, and generating enhanced second data enhanced with the third type of enhancement in response to the third analyzer signal and the second data reads on apply a second speech enhancement processing to the second obtained audio signal based, at least in part, on the second quality value.).
Dunne is considered to be analogous to the claimed invention because it is in the same field of audio 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 Dolson to incorporate the teachings of Dunne to determine first data is suitable for a first type of enhancement in response to a first mode signal and the first data, determine second data is suitable for a third type of enhancement in response to a second mode signal and the second data, generate enhanced first data enhanced with the first type of enhancement in response to the first analyzer signal and the first data, and generate enhanced second data enhanced with the third type of enhancement in response to the third analyzer signal and the second data. Doing so would allow for measuring metrics or characteristics of highly compressed data and use the metrics or characteristics to enable or disable audio enhancement (Dunne; Column 5, line 66 - Column 6, line 30).
Regarding claim 18, arguments analogous to claim 9 are applicable.
Conclusion
The prior art made of record and not relied upon is considered pertinent to applicant's disclosure:
Wojcieszak et al. (US Patent No. 11,176,956) teaches a method for enabling application directed latency control for wireless audio streaming.
Weingertner et al. (US Patent No. 10,212,552) teaches a method for controlling speech quality by controlling end-to-end latency and by improving speech quality in case of mobility scenarios.
Lawrence (US Patent No. 9,928,844) teaches a method of audio quality and latency adjustment for audio processing by using audio feedback.
Any inquiry concerning this communication or earlier communications from the examiner should be directed to James Boggs whose telephone number is (571)272-2968. The examiner can normally be reached M-F 8:00 AM - 5:00 PM.
Examiner interviews are available via telephone, in-person, and video conferencing using a USPTO supplied web-based collaboration tool. To schedule an interview, applicant is encouraged to use the USPTO Automated Interview Request (AIR) at http://www.uspto.gov/interviewpractice.
If attempts to reach the examiner by telephone are unsuccessful, the examiner’s supervisor, Daniel Washburn can be reached at (571)272-5551. The fax phone number for the organization where this application or proceeding is assigned is 571-273-8300.
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/JAMES BOGGS/Examiner, Art Unit 2657
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