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 .
Specification
The abstract of the disclosure is objected to because it contains 173 words and should not exceed 150 words in length. A corrected abstract of the disclosure is required and must be presented on a separate sheet, apart from any other text. See MPEP § 608.01(b).
Claim Rejections - 35 USC § 102
In the event the determination of the status of the application as subject to AIA 35 U.S.C. 102 and 103 (or as subject to pre-AIA 35 U.S.C. 102 and 103) is incorrect, any correction of the statutory basis (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.
(a)(1) the claimed invention was patented, described in a printed publication, or in public use, on sale, or otherwise available to the public before the effective filing date of the claimed invention.
Claims 15, 17, 20, 22-27, 29, 32 and 34 are rejected under 35 U.S.C. 102(a)(1) as being anticipated over Fisher et al. (US 20220303502 A1).
Regarding claim 15, Fisher teaches:
An apparatus comprising (the devices collectively implementing the multi-mic system are an apparatus, , see Element 600 in Fig. 6A, 6B, and comprise a conference bridge/bridge server/hosting platform, see Element 610 in Fig. 6A, also see [0022]: “A hosting platform (e.g., a conference bridge or bridge server)…”):
detect, during transmission of first and second data streams to a destination terminal,
wherein the first data stream carries audio of a first sound source captured by a first capture device, and the second data stream carries audio of a second sound source captured by a second capture device, that the first sound source is in proximity of the second capture device such that the second capture device is capable of capturing the audio of the first sound source (device 120a is the first capture device capturing the first data stream from the first sound source, i.e. person A, at least device 120b can be part of the second capture device capturing the second data stream from the second sound source, i.e. person B, both streams are transmitted to the device used by person Z, which can be part of a destination terminal, see [0011]: “Person A and person B use their own conferencing devices 120a and 120b, respectively, to have a conference call with person Z”, also see Fig. 2; Person A, i.e. the first sound source is also picked up by the second conferencing device, i.e. the second capture device, see [0011]: “When person A speaks (“1, 2, 3, 4, . . . ”; 201), her voice reaches the microphones of conferencing devices 120a and 120b at different times (203), e.g., because she is closer to one microphone than to the other microphone”; Detection the first sound source is closer to the second capture device is achieved by at least latency estimation methods, and/or conference bridge determining one audio stream, e.g. the first data stream, contains no significant signal, see [0112]: “The “Local Active Speaker” (the person speaking in a room at a particular moment) can be used to identify which client's microphone should be used at that particular moment in the conference. This helps ensure the shortest possible distance between the microphone to the active speaker…through the dynamic echo cancelation and latency estimation methods described above… If the conference bridge determines that a particular audio stream contains no significant signal“);
suspend, at a particular time following said detection, transmission of the audio of the first sound source in the first data stream when the audio of the first sound source is captured by the second capture device and transmitted in the second data stream, the particular time being determined based on one or more characteristics of the audio captured by at least one of the first capture device or the second capture device (One audio data is captured by both capture devices and the audio data can be the first sound source; Then at least the conference bridge determining that a particular audio stream, e.g. the first data stream, contains no significant signal, e.g., from the volume metadata, can be part of the detection; After the detection, the packets from the audio stream, i.e. the first data stream, is discarded, i.e. suspended, at the time when the audio stream is determined that it contains no significant signal; at least the volume metadata from the audio stream is one of the characteristics of the audio captured by the first capture device, see [0112]: “…audio and video data are captured from every participant's device and sent to the conference bridge…If the conference bridge determines that a particular audio stream contains no significant signal, e.g., from the volume metadata sent with every RTP packet, then the conference bridge may discard packets in that audio stream to conserve processing overhead”).
It is inherent that the host platform 610 in Fig. 6A which is a bridge server must have a processor and a memory to store software for computer system operation, otherwise it cannot function as a computer server to carry out all of those functions described in the reference of Fisher.
Regarding claim 17, Fisher teaches all the claim elements previously stated in claim 15’s 102
rejection.
Fisher also teaches the particular time is a time at which the first capture device captures no audio from the first sound source (a particular audio stream is the first data stream from the first capture device for the first sound source; no significant signal based on the volume metadata can be volume is zero, i.e. no audio, the time when no audio is in the first sound source is the particular time when the conference bridge can discard, i.e. suspend the first data stream, see [0112]: “…audio and video data are captured from every participant's device and sent to the conference bridge…If the conference bridge determines that a particular audio stream contains no significant signal, e.g., from the volume metadata sent with every RTP packet, then the conference bridge may discard packets in that audio stream to conserve processing overhead”).
Regarding claim 20, Fisher teaches all the claim elements previously stated in claim 15’s 102 rejection.
Fisher also teaches suspending transmission of the audio of the first sound source in the first data stream comprises suspending transmission of the first data stream (the conference bridge suspends the transmission of the first source in the first data stream by discarding the packets in the audio stream, i.e. data stream. Discarding the packets interrupts and suspends the data stream transmission, see [0112]: “…If the conference bridge determines that a particular audio stream contains no significant signal, e.g., from the volume metadata sent with every RTP packet, then the conference bridge may discard packets in that audio stream to conserve processing overhead”).
Regarding claim 22, Fisher teaches all the claim elements previously stated in claim 15’s 102 rejection.
Fisher also teaches an acoustic beamforming module for generating spatial audio data which comprises the captured audio of the first sound source, the captured audio of the second sound source and respective directional components for each said sound source (the acoustic beamforming module mixes signals captured by the capture devices into one audio data; the captured signals are sound sources from clients in one speaker group, which can be from the first sound source and the second sound source; the generated audio data from the beamforming module contains delay information and gain information from selected directions, therefore, it is a spatial audio data and the beamforming module is a spatial audio device; at least the gain information for audio signals from selected directions are directional components for each sound source; the acoustic beamforming module and the upstream device, e.g. the hosting platform can be a separate device in the second capture device and does not bring any side effects, see [0069]: “The media processor 630, also called a media processing layer…creating synthesized (mixed) audio and video streams…speaker attribution…The media processor 630 includes an acoustic beamforming module 632 that provides delays and mixes the audio signals from the clients 620 in each room to provide preferential gain for audio signals…from selected directions”, also see Fig. 6A),
Regarding claim 23, Fisher teaches all the claim elements previously stated in claim 22’s 102 rejection.
Fisher also teaches microphone can be directional (each client can adopt a directional microphone; two clients for different sound sources can adopt the directional microphones with each pointing to different respective directions; these two clients with the acoustic beamforming module and the hosting platform can be a separate device as set forth in claim 22’s rejection, which does not bring any side effects, see [0020]: “These microphones also tend to be directional…”; because the microphones are directional, and the directional components e.g. gain information, is direction based, therefore, the directional components can be determined based on which of the microphone the captured audio of each sound source is captured by, see[0069}: “an acoustic beamforming module 632 that provides delays and mixes the audio signals from the clients 620 in each room to provide preferential gain for audio signals…from selected directions”, also see Fig. 6A)
Regarding claim 24, Fisher teaches all the claim elements previously stated in claim 22’s 102 rejection.
Fisher also teaches separate the captured audio of the first sound source and the second sound source into respective first and second sound objects; and transmit the first and second sound objects and their respective directional components to the destination terminal (diarization is to separate the captured sound based on the speaker identity, i.e. the sound source; segments are respective first and second sound objects according to the speaker identify, see [0018]: “Some ASR systems may also include or use a diarization model to partition an input audio stream into segments according to the speaker identity”, also see Fig. 6A; The first and second sound objects are then transmitted to the media database which can be part of the destination terminal, see [0069]: “…the speaker assignment module 636 uses the beamforming module's output for diarization”, also see Fig. 6A; applying directional-based gain for audio signals results in respective volumes, which can be part of the directional components, this volume can be transmitted as metadata in network packets, see [0069]: “The media processor 630, also called a media processing layer…creating synthesized (mixed) audio and video streams…speaker attribution…The media processor 630 includes an acoustic beamforming module 632 that provides delays and mixes the audio signals from the clients 620 in each room to provide preferential gain for audio signals…from selected directions”, also see [0112]: “…the volume metadata sent with every RTP packet…”).
Regarding claim 25, Fisher teaches all the claim elements previously stated in claim 20’s 102 rejection. The system as taught by Fisher can be the apparatus, which includes the second capture device (at least the device 120b can be part of the second capture device, see Fig. 2).
Regarding claim 26, Fisher teaches all the claim elements previously stated in claim 20’s 102 rejection.
Fisher also teaches the internet or packet switched network connecting to and communicating with client including the second capture device ( The internet or packet switched network is a network entity, see [0003]: “a conferencing device or client, to connect to a conference server or host platform 110, via the internet or another suitable packet-switched network”).
Regarding claim 27, since the claimed method comprises the same operations conducted by the apparatus in claim 15, claim 27 is rejected as being anticipated over Fisher for the reasons mentioned in claim 15’s 102 rejection.
Regarding claim 29, since the claimed method comprises the same operations conducted by the apparatus in claim 17, claim 29 is rejected as being anticipated over Fisher for the reasons mentioned in claim 17’s 102 rejection.
Regarding claim 32, since the claimed method comprises the same operations conducted by the apparatus in claim 20, claim 32 is rejected as being anticipated over Fisher for the reasons mentioned in claim 20’s 102 rejection.
Regarding claim 34, since the claimed CRM comprises the same operations conducted by the apparatus in claim 15, claim 34 is rejected as being anticipated over Fisher for the reasons mentioned in claim 15’s 102 rejection.
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 16 and 28 are rejected under 35 U.S.C. 103 as being obvious over Fisher et al. (US 20220303502 A1) in view of Serbajlo et al. (US 20210058517 A1).
Regarding claim 16, Fisher teaches all the claim elements previously stated in claim 15’s 102 rejection
Fisher does not specifically teach the particular time is a time at which the second capture device captures audio from the second sound source.
Serbajlo teaches a suitable microphone can be enabled based on proximity values (a suitable microphone can be the second capture device, the time when it is enabled is the particular time when the second capture device captures audio from the second sound source, see: “A server can compare proximity values and enable a suitable microphone, while disabling the remaining microphones”, Abstract).
At the time of the invention was effectively filed, it would have been obvious to one of ordinary skill in the art to have set the particular time to be when a capture device is enabled to capture audio from the sound source as taught by Serbajlo in the apparatus as taught by Fisher. It would have yielded predictable results and resulted in an improved device. One of ordinary skill in the art would have been motivated to do so for “automatic enabling and disabling of microphones” (Serbajlo: [0039]).
Regarding claim 28, since the claimed method comprises the same operations conducted by the apparatus in claim 16, claim 28 is rejected as being obvious over Fisher for the reasons mentioned in claim 16’s 103 rejection.
Claims 18-19, and 30-31 are rejected under 35 U.S.C. 103 as being obvious over Fisher et al. (US 20220303502 A1) in view of Lovich (US 20200169822 A1).
Regarding claim 18, Fisher teaches all the claim elements previously stated in claim 15’s 102 rejection.
Fisher does not teach disabling capture of the audio of the first sound source by the first capture device at the particular time.
Lovich teaches disabling capture of the audio of the first sound source by the first capture device (The first device microphone can be the first capture device; When it is switched to inactive mode, capturing the audio of the first sound source in the first capture device is disabled, see [0004- 0005]: “the first device including a speaker, a microphone configured to operate in an active mode wherein it captures audio input for transmission to a computing device, and in an inactive mode in which it does not capture audio input… switching the first device microphone to the inactive mode” ).
At the time of the invention was effectively filed, it would have been obvious to one of ordinary
skill in the art to have disabled capture of the first sound source by the first capture device at a particular time as taught by Lovich in the apparatus as taught by Fisher. It would have yielded predictable results and resulted in an improved device. One of ordinary skill in the art would have been motivated to do so for “providing an improved user experience” by “automatic” and “seamless” switching (Lovich: [0020]).
Regarding claim 19, Fisher teaches all the claim elements previously stated in claim 18’s 103 rejection.
Fisher also teaches “enable capture of the audio of the first sound source by the second capture device at the particular time” (Person A, i.e. the first sound source is also picked up already by the second conferencing device, i.e. the second capture device; therefore capturing the audio of the first sound source by the second capture device is enabled at the particular time, see [0011]: “When person A speaks (“1, 2, 3, 4, . . . ”; 201), her voice reaches the microphones of conferencing devices 120a and 120b at different times (203), e.g., because she is closer to one microphone than to the other microphone”;)
Regarding claim 30, since the claimed method comprises the same operations conducted by the apparatus in claim 18, claim 30 is rejected as being obvious over Fisher for the reasons mentioned in claim 18’s 103 rejection.
Regarding claim 31, since the claimed method comprises the same operations conducted by the apparatus in claim 19, claim 31 is rejected as being obvious over Fisher for the reasons mentioned in claim 19’s 103 rejection.
Claims 21, and 33 are rejected under 35 U.S.C. 103 as being obvious over Fisher et al. (US 20220303502 A1) in view of Albadawi et al. (US 20180232563 A1).
Regarding claim 21, Fisher teaches all the claim elements previously stated in claim 20’s 103 rejection.
Fisher also teaches the underlined limitation “further detect, after suspending transmission of the
first data stream, that the first sound source is no longer in proximity of the second capture device “ ( Person A, i.e. the first sound source, can also be closer to the first capture device, i.e. no longer close to the second capture device, see [0011]: “When person A speaks (“1, 2, 3, 4, . . . ”; 201), her voice reaches the microphones of conferencing devices 120a and 120b at different times (203), e.g., because she is closer to one microphone than to the other microphone”; at least latency estimation methods can be used to detect when the sound source is no longer close to the second capture device, see [0112]: “The “Local Active Speaker” (the person speaking in a room at a particular moment) can be used to identify which client's microphone should be used at that particular moment in the conference. This helps ensure the shortest possible distance between the microphone to the active speaker…through the dynamic echo cancelation and latency estimation methods described above… If the conference bridge determines that a particular audio stream contains no significant signal“);
Fisher does not specifically mention further detection after suspending transmission of the first data stream, then resume transmission of the first data stream.
Albadawi teaches detecting which capture device is closer to the sound source as well as resumption data stream transmission after suspending ( the vicinity of sound source to the microphone, i.e. capture device, is used to determine which capture device to be selected, e.g. when the first sound source is closer to the first capture device and farther away from the second capture device, this process is also detection, see [0272]: “…the aggregator 182 may utilize these metrics to select one of the audio data streams and its corresponding microphone(s) for use in interacting with the user. In some examples, the microphone(s) that is determined to be closest to the user may be selected…”; the aggregator can be part of the apparatus, the first sound capture device can be one of “the other two devices”, transmitting of audio data streams, which can include the first data stream, can be resumed after being suspended, see [0286]: “the aggregator also may cause the other two devices to refrain from sending audio data streams that are associated with the same speaker ID (i.e., person) that is associated with the analyzed data stream…the other two devices may resume sending audio data streams when the same person speaks the keyword or keyword phrase”).
At the time of the invention was effectively filed, it would have been obvious to one of ordinary skill in the art to have integrated the resumption data stream transmission after it has been suspended scheme as taught by Albadawi when the first sound source is detected to be away from the second device as taught by Fisher. It would have yielded predictable results and resulted in an improved device. One of ordinary skill in the art would have been motivated to do so to “identify and select the desired device/data stream” (Albadawi: [0278]).
Regarding claim 33, since the claimed method comprises the same operations conducted by the apparatus in claim 21, claim 33 is rejected as being obvious over Fisher for the reasons mentioned in claim 21’s 103 rejection.
Conclusion
The prior arts made of record and not relied upon are considered pertinent to applicant's disclosure:
Fernandez et al. (US 20240406622 A1) teaches selecting audio signals based on certain indicators (Abstract)
Kupryjanow et al. (US 20240357285 A1) teaches a host-based system that has microphone auto-muting features based on a signal-based score (Abstract)
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/SHIN LEE/Examiner, Art Unit 2695
/VIVIAN C CHIN/Supervisory Patent Examiner, Art Unit 2695
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