Prosecution Insights
Last updated: August 17, 2026
Application No. 18/353,037

SPECIFYING LOUDNESS IN AN IMMERSIVE AUDIO PACKAGE

Non-Final OA §103
Filed
Jul 14, 2023
Priority
Mar 29, 2023 — provisional 63/492,927
Examiner
LIEBGOTT, TYLER MICHAEL
Art Unit
2694
Tech Center
2600 — Communications
Assignee
Google LLC
OA Round
3 (Non-Final)
73%
Grant Probability
Favorable
3-4
OA Rounds
0m
Est. Remaining
80%
With Interview

Examiner Intelligence

Grants 73% — above average
73%
Career Allowance Rate
19 granted / 26 resolved
+11.1% vs TC avg
Moderate +7% lift
Without
With
+7.2%
Interview Lift
resolved cases with interview
Typical timeline
2y 9m
Avg Prosecution
18 currently pending
Career history
55
Total Applications
across all art units

Statute-Specific Performance

§101
0.5%
-39.5% vs TC avg
§103
47.6%
+7.6% vs TC avg
§102
28.4%
-11.6% vs TC avg
§112
20.6%
-19.4% vs TC avg
Black line = Tech Center average estimate • Based on career data from 26 resolved cases

Office Action

§103
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 . Continued Examination Under 37 CFR 1.114 A request for continued examination under 37 CFR 1.114, including the fee set forth in 37 CFR 1.17(e), was filed in this application after final rejection. Since this application is eligible for continued examination under 37 CFR 1.114, and the fee set forth in 37 CFR 1.17(e) has been timely paid, the finality of the previous Office action has been withdrawn pursuant to 37 CFR 1.114. Applicant's submission filed on 04/16/2026 has been entered. Response to Amendment In response to the final office action dated 01/27/2026, applicant has submitted a request for continued examination. Claims 1, 3, 9, 12, 16-18, and 20 are amended. Claims 21 and 22 have been added. Claims 1-22 are currently pending in the application. Claim Rejections - 35 USC § 103 The text of those sections of Title 35, U.S. Code not included in this action can be found in a prior Office action. Claim(s) 1-3, 8-11, 16-19, and 21 is/are rejected under 35 U.S.C. 103 as being unpatentable over Riedmiller et al (U.S. Pub No. 20160196830, hereinafter Riedmiller) in view of Ahmed et al (US Pub No. 2022/0239947, hereinafter Ahmed). Regarding claim 1, Riedmiller teaches a method (See Riedmiller Abstract, methods for generating an encoded audio bitstream) comprising: generating an audio stream including a first substream as first audio data (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, first dependent substream) and a second substream as second audio data (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, second dependent substream); generating a first loudness parameter associated with playback of the first substream (See Riedmiller ¶ [0180], Loudness processing value of corresponding audio data); generating a second loudness parameter associated with playback of the second substream (See Riedmiller ¶ [0180], Loudness processing value of corresponding audio data); and generating an audio package (See Riedmiller Fig 4, encoded audio bitstream frame) including an identification corresponding to the first audio data (See Riedmiller ¶ [0055], each payload includes a header with a specific payload identifier), an identification corresponding to the second audio data (See Riedmiller ¶ [0055], each payload includes a header with a specific payload identifier), and a codec agnostic container including the first loudness parameter and the second loudness parameter (See Riedmiller Fig 8, metadata segment of encoded bitstream “container”). Riedmiller does not explicitly teach audio data (substreams) being asynchronous or having different frame sizes. Ahmed teaches audio data (substreams) being asynchronous (See Ahmed ¶ [0002], transmitting point cloud data, point cloud data uses asynchronous transmission allowing for independent processing of each point) or having different frame sizes (See Ahmed ¶ [0074], “An International Organization for Standardization (ISO) Base Media File Format (ISOBMFF) may define a structural, media-independent file format. An ISOBMFF (e.g., an ISOBMFF container file) may include structural and/or media data information, for example, for timed presentations of media contents such as audio, video, etc… ISOBMFF may be based on the concept of box-structured files. A box-structured file may include a series of boxes (e.g., atoms), which may have respective sizes and/or types (e.g., each box may be associated with a size and a type”). It would have been prima facie obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have incorporated the asynchronous audio data and box (frame) sizes as taught by Ahmed with the method taught by Riedmiller. Doing so provides three-dimensional point cloud for a high-level representation of immersion media and providing effective representation, compression and delivery techniques which are desired for storing and/or transmitting point cloud data (Ahmed ¶ [0002]). Regarding claim 2, Riedmiller in view of Ahmed teaches the method of claim 1, wherein the audio package is a first audio package (See Riedmiller Fig 4, encoded audio bitstream frame), the method further comprising: merging the first audio package with a second audio package to generate a third audio package (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, independent substream including first and second dependent substream), the second audio package including a third substream as third audio data (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, third dependent substream) and a third loudness parameter associated with playback of the third substream (See Riedmiller ¶ [0180], Loudness processing value of corresponding audio data); generating a normalization parameter associated with playback of the first substream, the second substream, and the third substream; and adding the normalization parameter to the codec agnostic container (See Riedmiller ¶ [0118-0119], loudness processing state metadata included in frame payload). Regarding claim 3, Riedmiller in view of Ahmed teaches the method of claim 2, wherein the merging of the first audio package with the second audio package includes mixing the first substream, the second substream, and the third substream as a fourth substream (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, independent substream including first, second, and third dependent substream), the normalization parameter is a target loudness associated with playback of the fourth substream (See Riedmiller ¶ [0118-0119], loudness processing state metadata included in frame payload); and the fourth substream replaces the first substream, the second substream, and the third substream in an audio presentation (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, independent substream including first, second, and third dependent substream). Regarding claim 8, Riedmiller in view of Ahmed teaches the method of claim 3, wherein if the first substream is associated with a surround speaker (See Riedmiller ¶ [0050], dependent substream indicative of standard format speaker channels including surround) and the second substream is associated with a top speaker (See Riedmiller ¶ [0050], dependent substream indicative of standard format speaker channels including center. Top speaker has no standard definition), the method further comprises: separately mixing the first substream and the second substream (See Riedmiller ¶ [0051], independent substream may be associated with up to eight dependent substreams). Regarding claim 9, Riedmiller teaches a non-transitory computer-readable storage medium (See Riedmiller ¶ [0233], storage media) comprising instructions stored thereon (See Riedmiller ¶ [0232], software instructions) that, when executed by at least one processor (See Riedmiller Fig 2, processor 103), are configured to cause a computing system to: generate an audio stream including a first substream as first audio data (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, first dependent substream) and a second substream as second audio data (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, second dependent substream); generate a first loudness parameter associated with playback of the first substream (See Riedmiller ¶ [0180], Loudness processing value of corresponding audio data); generate a second loudness parameter associated with playback of the second substream (See Riedmiller ¶ [0180], Loudness processing value of corresponding audio data); and generate an audio package (See Riedmiller Fig 4, encoded audio bitstream frame) including an identification corresponding to the first audio data, an identification corresponding to the second audio data (See Riedmiller ¶ [0055], each payload includes a header with a specific payload identifier), and a codec agnostic container including the first loudness parameter, and the second loudness parameter (See Riedmiller Fig 8, metadata segment of encoded bitstream “container”). Riedmiller does not explicitly teach audio data (substreams) being asynchronous or having different frame sizes. Ahmed teaches audio data (substreams) being asynchronous (See Ahmed ¶ [0002], transmitting point cloud data, point cloud data uses asynchronous transmission allowing for independent processing of each point) or having different frame sizes (See Ahmed ¶ [0074], “An International Organization for Standardization (ISO) Base Media File Format (ISOBMFF) may define a structural, media-independent file format. An ISOBMFF (e.g., an ISOBMFF container file) may include structural and/or media data information, for example, for timed presentations of media contents such as audio, video, etc… ISOBMFF may be based on the concept of box-structured files. A box-structured file may include a series of boxes (e.g., atoms), which may have respective sizes and/or types (e.g., each box may be associated with a size and a type”). It would have been prima facie obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have incorporated the asynchronous audio data and box (frame) sizes as taught by Ahmed with the non-transitory computer-readable storage medium taught by Riedmiller. Doing so provides three-dimensional point cloud for a high-level representation of immersion media and providing effective representation, compression and delivery techniques which are desired for storing and/or transmitting point cloud data (Ahmed ¶ [0002]). Regarding claim 10, Riedmiller in view of Ahmed teaches the non-transitory computer-readable storage medium of claim 9, wherein the audio package is a first audio package (See Riedmiller Fig 4, encoded audio bitstream frame), and the instructions are further configured to cause the computing system to: merge the first audio package with a second audio package to generate a third audio package (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, independent substream including first and second dependent substream), the second audio package including a third substream as third audio data (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, third dependent substream) and a third loudness parameter associated with playback of the third substream (See Riedmiller ¶ [0180], Loudness processing value of corresponding audio data); generate a normalization parameter associated with playback of the first substream, the second substream, and the third substream; and add the normalization parameter to the codec agnostic container (See Riedmiller ¶ [0118-0119], loudness processing state metadata included in frame payload). Regarding claim 11, Riedmiller in view of Ahmed teaches the non-transitory computer-readable storage medium of claim 10, wherein, the merging of the first audio package with the second audio package includes mixing the first substream, the second substream, and the third substream as a fourth substream (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, independent substream including first, second, and third dependent substream), the normalization parameter is a target loudness associated with playback of the fourth substream (See Riedmiller ¶ [0118-0119], loudness processing state metadata included in frame payload); and the fourth substream replaces the first substream, the second substream, and the third substream in an audio presentation (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, independent substream including first, second, and third dependent substream). Regarding claim 16, Riedmiller in view of Ahmed teaches the non-transitory computer-readable storage medium of claim 11, wherein if the first substream is associated with a surround speaker (See Riedmiller ¶ [0050], dependent substream indicative of standard format speaker channels including surround) and the second substream is associated with a top speaker (See Riedmiller ¶ [0050], dependent substream indicative of standard format speaker channels including center. Top speaker has no standard definition), wherein the instructions are further configured to cause the computing system to: separately mixing the first substream and the second substream (See Riedmiller ¶ [0051], independent substream may be associated with up to eight dependent substreams). Regarding claim 17, Riedmiller teaches an apparatus (See Riedmiller Abstract, apparatus for generating an encoded audio bitstream) comprising at least one processor (See Riedmiller Fig 2, processor 103) and at least one memory (See Riedmiller ¶ [0233], storage media) including computer program code (See Riedmiller ¶ [0232], software instructions), the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus to: generate an audio stream including a first substream as first audio data (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, first dependent substream) and a second substream as second audio data (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, second dependent substream); generate a first loudness parameter associated with playback of the first substream (See Riedmiller ¶ [0180], Loudness processing value of corresponding audio data); generate a second loudness parameter associated with playback of the second substream (See Riedmiller ¶ [0180], Loudness processing value of corresponding audio data); and generate an audio package (See Riedmiller Fig 4, encoded audio bitstream frame) including an identification corresponding to the first audio data (See Riedmiller ¶ [0055], each payload includes a header with a specific payload identifier), an identification corresponding to the second audio data (See Riedmiller ¶ [0055], each payload includes a header with a specific payload identifier), and a codec agnostic container including the first loudness parameter, and the second loudness parameter (See Riedmiller Fig 8, metadata segment of encoded bitstream “container”). Riedmiller does not explicitly teach audio data (substreams) being asynchronous or having different frame sizes. Ahmed teaches audio data (substreams) being asynchronous (See Ahmed ¶ [0002], transmitting point cloud data, point cloud data uses asynchronous transmission allowing for independent processing of each point) or having different frame sizes (See Ahmed ¶ [0074], “An International Organization for Standardization (ISO) Base Media File Format (ISOBMFF) may define a structural, media-independent file format. An ISOBMFF (e.g., an ISOBMFF container file) may include structural and/or media data information, for example, for timed presentations of media contents such as audio, video, etc… ISOBMFF may be based on the concept of box-structured files. A box-structured file may include a series of boxes (e.g., atoms), which may have respective sizes and/or types (e.g., each box may be associated with a size and a type”). It would have been prima facie obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have incorporated the asynchronous audio data and box (frame) sizes as taught by Ahmed with the apparatus taught by Riedmiller. Doing so provides three-dimensional point cloud for a high-level representation of immersion media and providing effective representation, compression and delivery techniques which are desired for storing and/or transmitting point cloud data (Ahmed ¶ [0002]). Regarding claim 18, Riedmiller in view of Ahmed teaches the apparatus of claim 17, wherein the audio package is a first audio package (See Riedmiller Fig 4, encoded audio bitstream frame), wherein the computer program code is further configured to cause the apparatus to: merging the first audio package with a second audio package to generate a third audio package (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, independent substream including first and second dependent substream), the second audio package including a third substream as third audio data (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, third dependent substream) and a third loudness parameter associated with playback of the third substream (See Riedmiller ¶ [0180], Loudness processing value of corresponding audio data); generating a normalization parameter associated with playback of the first substream, the second substream, and the third substream; and adding the normalization parameter to the codec agnostic container (See Riedmiller ¶ [0118-0119], loudness processing state metadata included in frame payload). Regarding claim 19, Riedmiller in view of Ahmed teaches the apparatus of claim 18, wherein, the merging of the first audio package with the second audio package includes mixing the first substream, the second substream, and the third substream as a fourth substream (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, independent substream including first, second, and third dependent substream), the normalization parameter is a target loudness associated with playback of the fourth substream (See Riedmiller ¶ [0118-0119], loudness processing state metadata included in frame payload); and the fourth substream replaces the first substream, the second substream, and the third substream in an audio presentation (See Riedmiller ¶ [0051], each independent substream can consist of up to eight dependent substreams, independent substream including first, second, and third dependent substream). Regarding claim 21, Riedmiller in view of Ahmed teaches the method of claim 1, wherein the codec agnostic container further includes automation data represented in blocks with time intervals decoupled from audio frame intervals of the first substream and the second substream (See Riedmiller ¶ [0072], adaptive and automated loudness processing state metadata included in bitstream). Claim(s) 4-7, 12-15, and 20 is/are rejected under 35 U.S.C. 103 as being unpatentable over Riedmiller et al (U.S. Pub No. 20160196830, hereinafter Riedmiller) in view of Ahmed et al (US Pub No. 2022/0239947, hereinafter Ahmed) as applied to claims above, and further in view of Seo (U.S. Pub No. 20170147282, hereinafter Seo). Regarding claim 4, Riedmiller in view of Ahmed teaches the method of claim 3, wherein the mixing of the first substream, the second substream, and the third substream as the fourth substream includes: determining a target sampling rate associated with playback of the first substream, the second substream, and the third substream (See Riedmiller ¶ [0037-0038], bitstream sampling rate). Riedmiller in view of Ahmed does not explicitly teach determining whether the substream sampling rates differ from the target sampling rate and if so, re-sampling the substream sampling rates to match the target sampling rate. Seo teaches determining whether the substream sampling rates differ from the target sampling rate (See Seo ¶ [0050], determines if sampling rate of audio data matches a sampling rate supported by the computing system) and if so, re-sampling the substream sampling rates to match the target sampling rate (See Seo ¶ [0050], sampling rate correction by up-sampling which includes re-sampling at an increased sampling rate). It would have been prima facie obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have included the sampling rate correction taught by Seo with the audio bitstream encoding method taught by Riedmiller in view of Ahmed. Sampling rate correction is well known in the art and plays a crucial role in audio processing to ensure accurate representation of signals. Regarding claim 5, Riedmiller in view of Ahmed and Seo teaches the method of claim 4. Riedmiller in view of Ahmed does not explicitly teach target sampling rate correction using up-sampling and down-sampling. Seo teaches sampling rate correction using up-sampling (See Seo ¶ [0050], sampling rate correction by up-sampling) and down-sampling (See Seo ¶ [0050], sampling rate correction by down-sampling). It would have been prima facie obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have included the sampling rate correction taught by Seo with the audio bitstream encoding method taught by Riedmiller in view of Ahmed and Seo. Both up-sampling and down-sampling are well known in the art and provide various benefits. Up-sampling increases the sampling rate allowing for better accuracy and reduced aliasing while down-sampling decreases the sampling rate which can reduce file size and processing time. Regarding claim 6, Riedmiller in view of Ahmed teaches the method of claim 2, further comprising: determining a target sampling rate associated with playback of the first substream, the second substream, and the third substream (See Riedmiller ¶ [0037-0038], bitstream sampling rate). Riedmiller in view of Ahmed does not explicitly teach determining whether the substream sampling rates differ from the target sampling rate and if so, re-sampling the substream sampling rates to match the target sampling rate. Seo teaches determining whether the substream sampling rates differ from the target sampling rate (See Seo ¶ [0050], determines if sampling rate of audio data matches a sampling rate supported by the computing system) and if so, re-sampling the substream sampling rates to match the target sampling rate (See Seo ¶ [0050], sampling rate correction by up-sampling which includes re-sampling at an increased sampling rate). It would have been prima facie obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have included the sampling rate correction taught by Seo with the audio bitstream encoding method taught by Riedmiller in view of Ahmed. Sampling rate correction is well known in the art and plays a crucial role in audio processing to ensure accurate representation of signals. Regarding claim 7, Riedmiller in view of Ahmed and Seo teaches the method of claim 6. Riedmiller in view of Ahmed does not explicitly teach target sampling rate correction using up-sampling and down-sampling. Seo teaches sampling rate correction using up-sampling (See Seo ¶ [0050], sampling rate correction by up-sampling) and down-sampling (See Seo ¶ [0050], sampling rate correction by down-sampling). It would have been prima facie obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have included the sampling rate correction taught by Seo with the audio bitstream encoding method taught by Riedmiller in view of Ahmed and Seo. Both up-sampling and down-sampling are well known in the art and provide various benefits. Up-sampling increases the sampling rate allowing for better accuracy and reduced aliasing while down-sampling decreases the sampling rate which can reduce file size and processing time. Regarding claim 12, Riedmiller in view of Ahmed teaches the non-transitory computer-readable storage medium of claim 11, wherein the mixing of the first substream, the second substream, and the third substream as a fourth substream includes: determining a target sampling rate associated with playback of the first substream, the second substream, and the third substream (See Riedmiller ¶ [0037-0038], bitstream sampling rate). Riedmiller in view of Ahmed does not explicitly teach determining whether the substream sampling rates differ from the target sampling rate and if so, re-sampling the substream sampling rates to match the target sampling rate. Seo teaches determining whether the substream sampling rates differ from the target sampling rate (See Seo ¶ [0050], determines if sampling rate of audio data matches a sampling rate supported by the computing system) and if so, re-sampling the substream sampling rates to match the target sampling rate (See Seo ¶ [0050], sampling rate correction by up-sampling which includes re-sampling at an increased sampling rate). It would have been prima facie obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have included the sampling rate correction taught by Seo with the audio bitstream encoding method taught by Riedmiller in view of Ahmed. Sampling rate correction is well known in the art and plays a crucial role in audio processing to ensure accurate representation of signals. Regarding claim 13, Riedmiller in view of Ahmed and Seo teaches the non-transitory computer-readable storage medium of claim 12. Riedmiller in view of Ahmed does not explicitly teach target sampling rate correction using up-sampling and down-sampling. Seo teaches sampling rate correction using up-sampling (See Seo ¶ [0050], sampling rate correction by up-sampling) and down-sampling (See Seo ¶ [0050], sampling rate correction by down-sampling). It would have been prima facie obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have included the sampling rate correction taught by Seo with the audio bitstream encoding method taught by Riedmiller in view of Ahmed and Seo. Both up-sampling and down-sampling are well known in the art and provide various benefits. Up-sampling increases the sampling rate allowing for better accuracy and reduced aliasing while down-sampling decreases the sampling rate which can reduce file size and processing time. Regarding claim 14, Riedmiller in view of Ahmed teaches the non-transitory computer-readable storage medium of claim 10, wherein the instructions are further configured to cause the computing system to: determine a target sampling rate associated with playback of the first substream, the second substream, and the third substream (See Riedmiller ¶ [0037-0038], bitstream sampling rate). Riedmiller in view of Ahmed does not explicitly teach determining whether the substream sampling rates differ from the target sampling rate and if so, re-sampling the substream sampling rates to match the target sampling rate. Seo teaches determining whether the substream sampling rates differ from the target sampling rate (See Seo ¶ [0050], determines if sampling rate of audio data matches a sampling rate supported by the computing system) and if so, re-sampling the substream sampling rates to match the target sampling rate (See Seo ¶ [0050], sampling rate correction by up-sampling which includes re-sampling at an increased sampling rate). It would have been prima facie obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have included the sampling rate correction taught by Seo with the audio bitstream encoding method taught by Riedmiller in view of Ahmed. Sampling rate correction is well known in the art and plays a crucial role in audio processing to ensure accurate representation of signals. Regarding claim 15, Riedmiller in view of Ahmed and Seo teaches the non-transitory computer-readable storage medium of claim 14. Riedmiller in view of Ahmed does not explicitly teach target sampling rate correction using up-sampling and down-sampling. Seo teaches sampling rate correction using up-sampling (See Seo ¶ [0050], sampling rate correction by up-sampling) and down-sampling (See Seo ¶ [0050], sampling rate correction by down-sampling). It would have been prima facie obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have included the sampling rate correction taught by Seo with the audio bitstream encoding method taught by Riedmiller in view of Ahmed. Both up-sampling and down-sampling are well known in the art and provide various benefits. Up-sampling increases the sampling rate allowing for better accuracy and reduced aliasing while down-sampling decreases the sampling rate which can reduce file size and processing time. Regarding claim 20, Riedmiller in view of Ahmed teaches the apparatus of claim 18, wherein the computer program code is further configured to cause the apparatus to: determining a target sampling rate associated with playback of the first substream, the second substream, and the third substream (See Riedmiller ¶ [0037-0038], bitstream sampling rate). Riedmiller in view of Ahmed does not explicitly teach determining whether the substream sampling rates differ from the target sampling rate and if so, re-sampling the substream sampling rates to match the target sampling rate. Seo teaches determining whether the substream sampling rates differ from the target sampling rate (See Seo ¶ [0050], determines if sampling rate of audio data matches a sampling rate supported by the computing system) and if so, re-sampling the substream sampling rates to match the target sampling rate (See Seo ¶ [0050], sampling rate correction by up-sampling which includes re-sampling at an increased sampling rate). It would have been prima facie obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to have included the sampling rate correction taught by Seo with the audio bitstream encoding method taught by Riedmiller in view of Ahmed. Sampling rate correction is well known in the art and plays a crucial role in audio processing to ensure accurate representation of signals. Allowable Subject Matter Claim 22 objected to as being dependent upon a rejected base claim, but would be allowable if rewritten in independent form including all of the limitations of the base claim and any intervening claims. Conclusion The prior art made of record and not relied upon is considered pertinent to applicant's disclosure. Atti et al (US Pub No. 2019/0013028) teaches multi-stream audio coding with parallel encoding and varying frame sizes. Any inquiry concerning this communication or earlier communications from the examiner should be directed to TYLER LIEBGOTT whose telephone number is (703)756-1818. The examiner can normally be reached Mon-Fri 10-6:30 EST. 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, Fan Tsang can be reached at (571)272-7547. The fax phone number for the organization where this application or proceeding is assigned is 571-273-8300. Information regarding the status of published or unpublished applications may be obtained from Patent Center. Unpublished application information in Patent Center is available to registered users. To file and manage patent submissions in Patent Center, visit: https://patentcenter.uspto.gov. Visit https://www.uspto.gov/patents/apply/patent-center for more information about Patent Center and https://www.uspto.gov/patents/docx for information about filing in DOCX format. For additional questions, contact the Electronic Business Center (EBC) at 866-217-9197 (toll-free). If you would like assistance from a USPTO Customer Service Representative, call 800-786-9199 (IN USA OR CANADA) or 571-272-1000. /T.M.L./Examiner, Art Unit 2694 /FAN S TSANG/Supervisory Patent Examiner, Art Unit 2694
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Prosecution Timeline

Show 1 earlier event
May 28, 2025
Non-Final Rejection mailed — §103
Aug 15, 2025
Applicant Interview (Telephonic)
Aug 15, 2025
Examiner Interview Summary
Aug 26, 2025
Response Filed
Jan 27, 2026
Final Rejection mailed — §103
Apr 16, 2026
Request for Continued Examination
Apr 19, 2026
Response after Non-Final Action
Jun 16, 2026
Non-Final Rejection mailed — §103 (current)

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Prosecution Projections

3-4
Expected OA Rounds
73%
Grant Probability
80%
With Interview (+7.2%)
2y 9m (~0m remaining)
Median Time to Grant
High
PTA Risk
Based on 26 resolved cases by this examiner. Grant probability derived from career allowance rate.

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