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 .
Status of the Claims
The office action is in response to the claim amendments and remarks filed on December 08, 2025 for the application filed February 10, 2023. Claims 14 and 26 have been amended. New claims 27-30 have been added. Claims 14-30 are currently pending.
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.
The factual inquiries for establishing a background for determining obviousness under 35 U.S.C. 103 are summarized as follows:
1. Determining the scope and contents of the prior art.
2. Ascertaining the differences between the prior art and the claims at issue.
3. Resolving the level of ordinary skill in the pertinent art.
4. Considering objective evidence present in the application indicating obviousness or nonobviousness.
Claims 14, 21, 26-30 are rejected under 35 U.S.C. 103 as being unpatentable over Solum et al. (US2015/0201289A1) in view of Kuwelkar et al. (US2016/0080459A1), Haartsen et al. (US2009/0298420A1) and Eidson et al. (US5566180A).
Regarding claim 14, Solum teaches a wireless conference system adapted to enable a plurality of users to participate to a conference in a conference room, said wireless conference system comprising an access point and a plurality of conference units (Paragraph [0017]: FIG. 1 is a block diagram illustrating an embodiment of an audio system 100 including an audio source device 102, audio sink devices 104, and a wireless communication network 106. In the illustrated embodiment, audio sink devices 104 includes audio sink device 104A coupled to audio source device 102 via wireless communication link 106A, audio sink device 104B coupled to audio source device 102 via wireless communication link 106B, . . . , audio sink device 104N coupled to audio source device 102 via wireless communication link 106N. In various embodiments, N is an integer equal to or greater than 1.);
wherein said access point and one or more of said conference units comprise a transceiver configured for bi-directional, time division multiple access based or TDMA based wireless communication of latency sensitive audio data packets between said one or more conference units and said access point, said transceiver comprising a transmitter and receiver (Paragraph [0018]: Modern hearing instruments employ digital processing techniques to enhance the intelligibility of incoming acoustic information for the hearing instrument wearer. In addition, modern hearing instruments have wireless connectivity to audio sources such as cellphones, computers including desktop, laptop, and tablet computers, and other devices that can send and receive digital audio information. Other devices such as wireless headphones process digital audio information in a similar manner. Paragraph [0020]: In various embodiments, audio source device 100 and audio sink devices 104 each include a sample clock for processing audio signal that is digitized and transmitted via wireless communication network 106. The sample clock of each of audio sink devices 104A-N is synchronized to the sample clock of the audio source device 102 to prevent audio sample over-runs or under-runs causing undesirable audio artifacts. Paragraph [0029]: In various embodiments, synchronization may have to survive events such as lost packets while maintaining a close relationship between sample clocks 224 and 230. To do this, processing circuit 216 can be aided from the radio of wireless communication circuit 218 by being updated as to the status of a packet. This may include information from the radio as to whether a packet was empty, missed, or received with errors. Missed packets are much more easily identified by the radio which has employed time division multiple access techniques. These techniques have inherently good timing mechanisms with which to schedule packet arrivals. Should a scheduled receiving event take place without a received packet, the radio will immediately recognize this event and be able to inform the DSP of the missing packet.)
wherein said access point and said one or more conference units comprise respective clocks that are actively synchronized, a clock of said respective clocks being configured to generate a local audio clock signal used locally for processing said audio data packets and a local synchronization clock signal used for said TDMA based wireless communication (Paragraph [0022]: The processing circuit 212 is configured to process an acoustic signal for transmission to audio sink device 204 via wireless communication link 206 and includes an audio reference oscillator 222 and a sample clock 224. Sample clock 224 times the input sampling of the acoustic signal based on the frequency generated by audio reference oscillator 222. Paragraph [0023]: The processing circuit 216 is configured to process the acoustic signal transmitted from audio source device 202 via wireless communication link 206 and includes an audio reference oscillator 228 and a sample clock 230. Sample clock 230 times the output sampling of the acoustic signal based on the frequency generated by audio reference oscillator 228. Paragraph [0026]: Therefore it is necessary for the audio sink device to lock its audio sample rate to audio sample rate of the audio source device to avoid dropping or adding samples periodically which would cause audio artifacts. Paragraph [0027]: These adjustments eventually lead to a lock condition where sample clock 230 of audio sink device 204 has reached a tolerable error from audio source device 202. Paragraph [0027]: Once audio source device 202 starts sending data, it will be a short time to achieve the lock since audio sink device 204 has already “trained” its audio reference oscillator 228 to within an acceptable error using communication reference oscillator 232 as a reference.)
wherein said receiver comprises a packet loss detection unit configured to detect loss of an audio data packet transmitted from a conference unit to said access point or vice-versa, said packet loss detection unit comprising: and a packet loss detection unit configured to verify if an audio data packet is received by said expected arrival time and to detect that said audio data packet is lost if it has not arrived by said expected arrival time; and wherein said receiver comprises a packet loss concealment unit configured to generate a replacement packet for said audio data packet that is detected to be lost by said packet loss detection unit (Paragraph [0013]: In various embodiments, the sample clock of an audio sink device is synchronized to the sample clock of an audio source device using a radio employing a packet based method of transmission based on a receive time slot. Paragraph [0015]: In various embodiments, the link layer of a radio is used to determine when a packet is missed, sent with errors or is empty for the purpose of employing a packet loss concealment strategy for the missing, empty, or packet received in error. Paragraph [0029]: In various embodiments, synchronization may have to survive events such as lost packets while maintaining a close relationship between sample clocks 224 and 230. To do this, processing circuit 216 can be aided from the radio of wireless communication circuit 218 by being updated as to the status of a packet. This may include information from the radio as to whether a packet was empty, missed, or received with errors. Such link layer information can be sent through an interface between the radio of wireless communication circuit 218 and the DSP of processing circuit 216. The DSP can use this information to determine whether a packet loss concealment strategy should be employed to substitute the missing audio with silence, a previously received packet, or an interpolated packet based on linear prediction. Missed packets are much more easily identified by the radio which has employed time division multiple access techniques. These techniques have inherently good timing mechanisms with which to schedule packet arrivals. Should a scheduled receiving event take place without a received packet, the radio will immediately recognize this event and be able to inform the DSP of the missing packet. Paragraph [0031]: In various embodiments, the radio of wireless communication circuit 218 sends information to the DSP of processing circuit 216 that a packet was missing or was sent as an empty packet to allow the DSP to insert a packet loss concealment (PLC) frame of information. This may be silent packet, replay or Linear predicted frame based on previous frames.)
Solum does not explicitly teach an estimated time of arrival unit configured to determine an expected arrival time for said audio data packet from said local synchronization clock signal by increasing the time said audio data packet is ready for transmission at said conference unit with a predetermined expected transmission delay, the predetermined expected transmission delay being an acceptable time required for interrupt handling at the transmitter and receiver side, propagation through the air, and/or possible jitter.
However, Kuwelkar teaches an estimated time of arrival unit configured to determine an expected arrival time for said audio data packet from said local synchronization clock signal by increasing the time said audio data packet is ready for transmission at said conference unit with a predetermined expected transmission delay (Paragraph [0028]: The presentation time may be calculated based on a clock signal from transmission media clock 209. Paragraph [0030]: In some examples, the presentation time may be utilized to synchronize the receive media clock 221 with the transmission media clock 209. For example, if the network delay (e.g., the max transit time) is known by both the talker and the listener, the listener may compare a receive time with an expected receive time (e.g., based on a known transmission delay and the presentation time) and adjust the receive media clock based on a calculated error (e.g., the difference between the measured receive time and the expected receive time).
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide an estimated time of arrival unit configured to determine an expected arrival time for said audio data packet from said local synchronization clock signal by increasing the time said audio data packet is ready for transmission at said conference unit with a predetermined expected transmission delay, as taught by Kuwelkar in the system of Solum, so that the expected time of arrival of the data packets can be used to ensure quality audio output and synchronization among listeners (Kuwelkar: Paragraphs [0003], [0026], [0028], [0030]).
The combination of Solum and Kuwelkar does not explicitly teach a local synchronization clock signal used for said TDMA based wireless communication; to determine an expected arrival time for said audio data packet from said local synchronization clock signal by increasing the time said audio data packet is ready for transmission at said conference unit with a predetermined expected transmission delay, the predetermined expected transmission delay being an acceptable time required for interrupt handling at the transmitter and receiver side, propagation through the air, and/or possible jitter.
However, Haartsen teaches a local synchronization clock signal used for said TDMA based wireless communication (Paragraph [0008]: the packets of audio data are transmitted from the audio source device to different ones of the speaker devices through corresponding different sequential communication frames of a time divisional multiple access (TDMA) network. Cycles of the common network clock are generated within the speaker devices relative to timing of receipt of defined signaling for the frames of the TDMA network. Each of the speaker devices initiates the sound generation in response to occurrence of a timing event defined by the command relative to cycles of the common network clock. Paragraph [0060]: The common network clock is thereby used to compensate for drift that occurs over time between the local clock device 400 (i.e., the crystal Xtal and the synthesizer) and local clock circuits of other associated speaker devices. The time offset 406 can be defined to align the adjusted clock signal 404 with signaling of the Bluetooth piconet. Such time alignment can be used to synchronize the timing of the audio output from a plurality of the slave speaker devices.)
to determine an expected arrival time for said audio data packet from said local synchronization clock signal by increasing the time said audio data packet is ready for transmission at said conference unit (Paragraph [0057]: FIG. 4 is a block diagram of receiver circuitry of a Bluetooth speaker device in accordance with some embodiments of the present invention. Referring to FIG. 4, the receiver circuit includes a local clock device 400 that generates a local clock signal 402 (e.g., 4 MHz). A common network clock is established that is timed relative to defined repetitively occurring signals of the Bluetooth piconet. More particularly, an adjusted clock signal 404 is generated that is time offset 406 a controllable amount relative to the local clock signal 402. The time offset 406 is controlled in response to phase differences between the local clock signal 402 and timing of defined repetitively occurring signals of the Bluetooth piconet (i.e., the common network clock). Paragraph [0060]: The common network clock is thereby used to compensate for drift that occurs over time between the local clock device 400 (i.e., the crystal Xtal and the synthesizer) and local clock circuits of other associated speaker devices. The time offset 406 can be defined to align the adjusted clock signal 404 with signaling of the Bluetooth piconet. Such time alignment can be used to synchronize the timing of the audio output from a plurality of the slave speaker devices, and may be used to assist with maintaining frequency hop synchronicity between the audio source device 110 and a plurality of speaker devices (e.g., operating as master and slave devices on the Bluetooth piconet) and allowing the speaker devices to more accurately predict the arrival of audio data packets from the audio source device 110. Paragraph [0074]: Using such signaling from the Bluetooth piconet, the time of arrival of an audio data packet may be determined with an accuracy of 0.25 μs or better. As described above, the arrival of an audio data packet may be accurately determined using a correlation unit (e.g., within the transceiver 502) that is matched to the bit sequence of the access code of each frame of the Bluetooth piconet.)
with the predetermined expected transmission delay being an acceptable time required for propagation through the air, and/or possible jitter (Paragraph [0042]: For a stereo headset, the Applicants hereof have determined that the distance between a person and the spaced apart speakers can affect the amount of time delay that can exist between arrival of corresponding sounds from the speakers before the person perceives an undesirable level of time delay (e.g., noise) therefrom. Paragraph [0043]: A person's hearing may be even more sensitive to rapid variation over time in the timing offsets due to, for example, jitter in the relative delay between the corresponding sounds from the speakers. Thus, for example, there may be an even greater need to avoid undesirable jitter (e.g., rapid change over time) in the relative timing between the corresponding sounds. Paragraph [0044]: Therefore, it can be important to control when each of the spaced apart speaker devices begins producing sound that is generated from decoding audio data that is received through a streaming wireless channel, and it can be further important to control drifting between local clocks that are used within each of the speaker devices to control audio processing and/or other sound generation operations and which can undesirably accumulate relative time skew over time.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide a local synchronization clock signal used for said TDMA based wireless communication; to determine an expected arrival time for said audio data packet from said local synchronization clock signal by increasing the time said audio data packet is ready for transmission at said conference unit; with the predetermined expected transmission delay being an acceptable time required for, propagation through the air, and/or possible jitter, as taught by Haartsen in the combined system of Solum and Kuwelkar, so that the receiver devices can synchronize the start of decoding their received data and compensate for timing offsets and skew over time of internal clocks in response to occurrence of timing events that are defined relative to signaling from the wireless communication network, configured as a time division multiple access (TDMA) network (Haartsen: Paragraphs [0005], [0008], [0042], [0057], [0060]).
The combination of Solum, Kuwelkar, and Haartsen does not explicitly teach delay being an acceptable time required for interrupt handling at the transmitter and receiver side.
However, Eidson teaches delay being an acceptable time required for interrupt handling at the transmitter and receiver side (Col 2, lines 11-19: However, as shown in FIG. 1, this technique may remove the effects of the operating system but does not remove the jitter and latency of the protocol stack of the communication system. Implementing the synchronization unit in a microprocessor may introduce jitter of its own due to operating system or interrupt behavior of the microprocessor. This system also introduces an unknown latency within the synchronization unit itself.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide delay being an acceptable time required for interrupt handling at the transmitter and receiver side, as taught by Eidson in the combined system of Solum, Kuwelkar, and Haartsen, so that the delay due to the interrupt behavior of the microprocessor can be factored into determining the correction for clock synchronization (Eidson: Col 2, lines 1-19).
Regarding claim 21, the combination of Solum, Kuwelkar, Haartsen, and Eidson teaches the wireless conference system according to claim 14 (see rejection for claim 14);
Solum does not explicitly teach wherein said wireless communication uses Wi-Fi.
However, Kuwelkar teaches wherein said wireless communication uses Wi-Fi (Paragraph [0017]: FIG. 1 shows an example partial view of one type of environment for a communication system: an interior of a cabin 100 of a vehicle 102, in which a driver and/or one or more passengers may be seated. Paragraph [0021]: The mobile device 128 may be connected to the in-vehicle computing system via communication link 130. The communication link 130 may be wired (e.g., via Universal Serial Bus [USB], Mobile High-Definition Link [MHL], High-Definition Multimedia Interface [HDMI], etc.) or wireless (e.g., via BLUETOOTH, WI-FI, Near-Field Communication [NFC], cellular connectivity, etc.) and configured to provide two-way communication between the mobile device and the in-vehicle computing system. For example, the communication link 130 may provide sensor and/or control signals from various vehicle systems (such as vehicle audio system). Paragraph [0027]: It is to be understood that FIG. 1 depicts one example environment, however the communication systems and methods described herein may be utilized in any suitable environment. As another example, speakers in a professional audio environment (e.g., an arena, stadium, concert hall, amphitheater, recording studio, etc.) may be utilized as listeners that receive audio data from a talker device.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide wherein said wireless communication uses Wi-Fi, as taught by Kuwelkar in the system of Solum, so that communication across speakers in a professional audio environment can be conveniently achieved (Kuwelkar: Paragraphs [0017], [0021], [0027]).
Regarding claim 26, Solum teaches a method for transfer of latency sensitive audio data packets between one or more conference units and an access point in a wireless conference system adapted to enable a plurality of users to participate to a conference in a conference room, said method for transfer comprising bi-directional, time division multiple access based or TDMA based wireless communication of said audio data packets, said method further comprising: actively synchronizing respective clocks in said one or more conference units and said access point, a clock of said respective clocks being configured to generate a local audio clock signal used locally for processing said audio data packets and a local synchronization clock signal used for said TDMA based wireless communication; detecting loss of an audio data packet transmitted from a conference unit to said access point or vice-versa; and verify if an audio data packet is received by said expected arrival time and detecting that said audio data packet is lost if it has not arrived by said expected arrival time; and generating a replacement packet for said audio data packet that is detected to be lost through packet loss concealment (see rejection for claim 14);
Solum does not explicitly teach comprising: determining an expected arrival time for said audio data packet from said local synchronization clock signal by increasing the time said audio data packet is ready for transmission at said conference unit with a predetermined expected transmission delay, the predetermined expected transmission delay being an acceptable time required for interrupt handling at the transmitter and receiver side, propagation through the air, and/or possible jitter.
However, Kuwelkar teaches comprising: determining an expected arrival time for said audio data packet from said local synchronization clock signal by increasing the time said audio data packet is ready for transmission at said conference unit with a predetermined expected transmission delay (see rejection for claim 14);
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide comprising: determining an expected arrival time for said audio data packet from said local synchronization clock signal by increasing the time said audio data packet is ready for transmission at said conference unit with a predetermined expected transmission delay, as taught by Kuwelkar in the system of Solum, so that the expected time of arrival of the data packets can be used to ensure quality audio output and synchronization among listeners (Kuwelkar: Paragraphs [0003], [0026], [0028], [0030]).
The combination of Solum and Kuwelkar does not explicitly teach a local synchronization clock signal used for said TDMA based wireless communication; determining an expected arrival time for said audio data packet from said local synchronization clock signal by increasing the time said audio data packet is ready for transmission at said conference unit with a predetermined expected transmission delay, the predetermined expected transmission delay being an acceptable time required for interrupt handling at the transmitter and receiver side, propagation through the air, and/or possible jitter.
However, Haartsen teaches a local synchronization clock signal used for said TDMA based wireless communication; determining an expected arrival time for said audio data packet from said local synchronization clock signal by increasing the time said audio data packet is ready for transmission at said conference unit; with the predetermined expected transmission delay being an acceptable time required for propagation through the air, and/or possible jitter (see rejection for claim 14);
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide a local synchronization clock signal used for said TDMA based wireless communication; determining an expected arrival time for said audio data packet from said local synchronization clock signal by increasing the time said audio data packet is ready for transmission at said conference unit; with the predetermined expected transmission delay being an acceptable time required for, propagation through the air, and/or possible jitter, as taught by Haartsen in the combined system of Solum and Kuwelkar, so that the receiver devices can synchronize the start of decoding their received data and compensate for timing offsets and skew over time of internal clocks in response to occurrence of timing events that are defined relative to signaling from the wireless communication network, configured as a time division multiple access (TDMA) network (Haartsen: Paragraphs [0005], [0008], [0042], [0057], [0060]).
The combination of Solum, Kuwelkar, and Haartsen does not explicitly teach delay being an acceptable time required for interrupt handling at the transmitter and receiver side.
However, Eidson teaches delay being an acceptable time required for interrupt handling at the transmitter and receiver side (see rejection for claim 14);
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide delay being an acceptable time required for interrupt handling at the transmitter and receiver side, as taught by Eidson in the combined system of Solum, Kuwelkar, and Haartsen, so that the delay due to the interrupt behavior of the microprocessor can be factored into determining the correction for clock synchronization (Eidson: Col 2, lines 1-19).
Regarding claim 27, the combination of Solum, Kuwelkar, Haartsen, and Eidson teaches the wireless conference system according to claim 14 (see rejection for claim 14);
The combination of Solum, Kuwelkar, and Haartsen does not explicitly teach wherein the predetermined expected transmission delay is an acceptable time required for interrupt handling at the transmitter and receiver side.
However, Eidson teaches wherein the predetermined expected transmission delay is an acceptable time required for interrupt handling at the transmitter and receiver side (Col 2, lines 11-19: However, as shown in FIG. 1, this technique may remove the effects of the operating system but does not remove the jitter and latency of the protocol stack of the communication system. Implementing the synchronization unit in a microprocessor may introduce jitter of its own due to operating system or interrupt behavior of the microprocessor. This system also introduces an unknown latency within the synchronization unit itself.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide wherein the predetermined expected transmission delay is an acceptable time required for interrupt handling at the transmitter and receiver side, as taught by Eidson in the combined system of Solum, Kuwelkar, and Haartsen, so that the delay due to the interrupt behavior of the microprocessor can be factored into determining the correction for clock synchronization (Eidson: Col 2, lines 1-19).
Regarding claim 28, the combination of Solum, Kuwelkar, Haartsen, and Eidson teaches the wireless conference system according to claim 14 (see rejection for claim 14);
The combination of Solum, Kuwelkar, and Eidson does not explicitly teach wherein the predetermined expected transmission delay is an acceptable time required for propagation through the air.
However, Haartsen teaches wherein the predetermined expected transmission delay is an acceptable time required for propagation through the air (Paragraph [0042]: For a stereo headset, the Applicants hereof have determined that the distance between a person and the spaced apart speakers can affect the amount of time delay that can exist between arrival of corresponding sounds from the speakers before the person perceives an undesirable level of time delay (e.g., noise) therefrom. Paragraph [0044]: Therefore, it can be important to control when each of the spaced apart speaker devices begins producing sound that is generated from decoding audio data that is received through a streaming wireless channel, and it can be further important to control drifting between local clocks that are used within each of the speaker devices to control audio processing and/or other sound generation operations and which can undesirably accumulate relative time skew over time.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide wherein the predetermined expected transmission delay is an acceptable time required for propagation through the air, as taught by Haartsen in the combined system of Solum, Kuwelkar, and Eidson, so that the receiver devices can synchronize the start of decoding their received data and compensate for timing offsets and skew over time of internal clocks in response to occurrence of timing events that are defined relative to signaling from the wireless communication network, configured as a time division multiple access (TDMA) network (Haartsen: Paragraphs [0005], [0008], [0042], [0057], [0060]).
Regarding claim 29, the combination of Solum, Kuwelkar, Haartsen, and Eidson teaches the wireless conference system according to claim 14 (see rejection for claim 14);
The combination of Solum, Kuwelkar, and Eidson does not explicitly teach wherein the
predetermined expected transmission delay is an acceptable time required for possible jitter.
However, Haartsen teaches wherein the predetermined expected transmission delay is an
acceptable time required for possible jitter (Paragraph [0043]: A person's hearing may be even more sensitive to rapid variation over time in the timing offsets due to, for example, jitter in the relative delay between the corresponding sounds from the speakers. Thus, for example, there may be an even greater need to avoid undesirable jitter (e.g., rapid change over time) in the relative timing between the corresponding sounds. Paragraph [0044]: Therefore, it can be important to control when each of the spaced apart speaker devices begins producing sound that is generated from decoding audio data that is received through a streaming wireless channel, and it can be further important to control drifting between local clocks that are used within each of the speaker devices to control audio processing and/or other sound generation operations and which can undesirably accumulate relative time skew over time.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective
filing date of the claimed invention to provide wherein the predetermined expected transmission delay is an acceptable time required for possible jitter, as taught by Haartsen in the combined system of Solum and Kuwelkar, so that the receiver devices can synchronize the start of decoding their received data and compensate for timing offsets and skew over time of internal clocks in response to occurrence of timing events that are defined relative to signaling from the wireless communication network, configured as a time division multiple access (TDMA) network (Haartsen: Paragraphs [0005], [0008], [0042], [0057], [0060]).
Regarding claim 30, the combination of Solum, Kuwelkar, Haartsen, and Eidson teaches the wireless conference system according to claim 14 (see rejection for claim 14);
The combination of Solum and Kuwelkar does not explicitly teach wherein the predetermined
expected transmission delay is an acceptable time required for interrupt handling at the transmitter and receiver side, propagation through the air, and possible jitter.
However, Haartsen teaches wherein the predetermined expected transmission delay is an
acceptable time required for propagation through the air, and possible jitter (Paragraph [0042]: For a stereo headset, the Applicants hereof have determined that the distance between a person and the spaced apart speakers can affect the amount of time delay that can exist between arrival of corresponding sounds from the speakers before the person perceives an undesirable level of time delay (e.g., noise) therefrom. Paragraph [0043]: A person's hearing may be even more sensitive to rapid variation over time in the timing offsets due to, for example, jitter in the relative delay between the corresponding sounds from the speakers. Thus, for example, there may be an even greater need to avoid undesirable jitter (e.g., rapid change over time) in the relative timing between the corresponding sounds. Paragraph [0044]: Therefore, it can be important to control when each of the spaced apart speaker devices begins producing sound that is generated from decoding audio data that is received through a streaming wireless channel, and it can be further important to control drifting between local clocks that are used within each of the speaker devices to control audio processing and/or other sound generation operations and which can undesirably accumulate relative time skew over time.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective
filing date of the claimed invention to provide wherein the predetermined expected transmission delay is an acceptable time required for propagation through the air, and possible jitter, as taught by Haartsen in the combined system of Solum and Kuwelkar, so that the receiver devices can synchronize the start of decoding their received data and compensate for timing offsets and skew over time of internal clocks in response to occurrence of timing events that are defined relative to signaling from the wireless communication network, configured as a time division multiple access (TDMA) network (Haartsen: Paragraphs [0005], [0008], [0042], [0057], [0060]).
The combination of Solum, Kuwelkar, and Haartsen does not explicitly teach wherein the
predetermined expected transmission delay is an acceptable time required for interrupt handling at the transmitter and receiver side.
However, Eidson teaches wherein the predetermined expected transmission delay is an
acceptable time required for interrupt handling at the transmitter and receiver side (Col 2, lines 11-19: However, as shown in FIG. 1, this technique may remove the effects of the operating system but does not remove the jitter and latency of the protocol stack of the communication system. Implementing the synchronization unit in a microprocessor may introduce jitter of its own due to operating system or interrupt behavior of the microprocessor. This system also introduces an unknown latency within the synchronization unit itself.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective
filing date of the claimed invention to provide wherein the predetermined expected transmission delay is an acceptable time required for interrupt handling at the transmitter and receiver side, as taught by Eidson in the combined system of Solum, Kuwelkar, and Haartsen, so that the delay due to the interrupt behavior of the microprocessor can be factored into determining the correction for clock synchronization (Eidson: Col 2, lines 1-19).
Claims 15, 16 are rejected under 35 U.S.C. 103 as being unpatentable over Solum et al. (US2015/0201289A1) in view of Kuwelkar et al. (US2016/0080459A1), Haartsen et al. (US2009/0298420A1), Eidson et al. (US5566180A), and further in view of Gilson et al. (US2020/0243103A1).
Regarding claim 15, the combination of Solum, Kuwelkar, Haartsen, and Eidson teaches the wireless conference system according to claim 14 (see rejection for claim 14);
The combination of Solum, Kuwelkar, Haartsen, and Eidson does not explicitly teach wherein said access point and said one or more conference units are configured to not acknowledge receipt of audio data packets.
However, Gilson teaches wherein said access point and said one or more conference units are configured to not acknowledge receipt of audio data packets (Paragraph [0027]: A user device such as a remote control for a cable set-top box may have access to a transmit input (microphone) signal directly from a microphone array when it is located in the user device. The receive output (the signal travelling toward the speaker) may travel from the set-top box to a television. The television may perform the digital to analog conversion and may feed the converted signal to speakers or other audio output devices associated with the television. A copy of the receive audio stream may be sent from the set-top box to the user device via a WiFi (or other) connection. This enables the first condition above to be satisfied. However, WiFi networks can exhibit considerable packet jitter, making it difficult to resolve the relative timing between the receive and transmit signals. Paragraph [0028]: Methods, devices, and systems are described herein that resolve the relative timing between the receive and transmit signals. In some implementations, an Internet Protocol (IP) stream of audio packets may be synchronized with audio received at a microphone. Paragraph [0030]: In the systems and methods described herein, two connections may be maintained between the computing device and the device to provide the audio timing reference:….. a Radio Frequency for Consumer Electronics (RF4CE) connection, which is faster and may act as a reference constant. Paragraph [0031]: The RF4CE connection may comprise parameter modifications. Parameter modifications including but not limited to the following may reduce latency and increase the predictability of the control/timing stream over RF4CE: Paragraph [0035]: Each frame in the RF4CE connection may disable retransmissions and may not be acknowledged.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide wherein said access point and said one or more conference units are configured to not acknowledge receipt of audio data packets, as taught by Gilson in the combined system of Solum, Kuwelkar, Haartsen, and Eidson, so that latency can be reduced by avoiding time taken to acknowledge receipt of audio data packets, and a more deterministic transmission timing can be achieved (Gilson: Paragraphs [0029], [0031], [0035]).
Regarding claim 16, the combination of Solum, Kuwelkar, Haartsen, and Eidson teaches the wireless conference system according to claim 14 (see rejection for claim 14);
The combination of Solum, Kuwelkar, Haartsen, and Eidson does not explicitly teach wherein said access point and said one or more conference units are configured to not retransmit a lost audio data packet.
However, Gilson teaches wherein said access point and said one or more conference units are configured to not retransmit a lost audio data packet (Paragraph [0028]: Methods, devices, and systems are described herein that resolve the relative timing between the receive and transmit signals. In some implementations, an Internet Protocol (IP) stream of audio packets may be synchronized with audio received at a microphone. Paragraph [0030]: In the systems and methods described herein, two connections may be maintained between the computing device and the device to provide the audio timing reference:….. a Radio Frequency for Consumer Electronics (RF4CE) connection, which is faster and may act as a reference constant. Paragraph [0031]: The RF4CE connection may comprise parameter modifications. Parameter modifications including but not limited to the following may reduce latency and increase the predictability of the control/timing stream over RF4CE: Paragraph [0035]: Each frame in the RF4CE connection may disable retransmissions and may not be acknowledged.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide wherein said access point and said one or more conference units are configured to not retransmit a lost audio data packet, as taught by Gilson in the combined system of Solum, Kuwelkar, Haartsen, and Eidson, so that latency can be reduced by avoiding time taken to retransmit audio data packets, and a more deterministic transmission timing can be achieved (Gilson: Paragraphs [0029], [0031], [0035]).
Claim 17 is rejected under 35 U.S.C. 103 as being unpatentable over Solum et al. (US2015/0201289A1) in view of Kuwelkar et al. (US2016/0080459A1), Haartsen et al. (US2009/0298420A1), Eidson et al. (US5566180A), and further in view of Lee et al. (US2013/0163428A1).
Regarding claim 17, the combination of Solum, Kuwelkar, Haartsen, and Eidson teaches the wireless conference system according to claim 14 (see rejection for claim 14);
The combination of Solum, Kuwelkar, Haartsen, and Eidson does not explicitly teach wherein said latency sensitive audio data packets have a round trip time latency limit of 25 milliseconds for wireless transfer from a conference unit to said access point, and wireless transfer from said access point to said conference unit.
However, Lee teaches wherein said latency sensitive audio data packets have a round trip time latency limit of 25 milliseconds for wireless transfer from a conference unit to said access point, and wireless transfer from said access point to said conference unit (Paragraph [0061]: In the first set of experiments the effort was made to measure the performance characteristics of the LTE network. Table 3 summarizes the measured RTT and packet loss rate at sending rate ranging from 1.6 Mbps to 80 Mbps. Paragraph [0062]: There are two observations. First, while the RTT is very short at low data rates (e.g., 25 ms at 1.6 Mbps), it begins to increase significantly even at medium data rates (e.g., 171 ms at 36 Mbps). Paragraph [0063]: Second, the packet loss rate is not insignificant even at low data rates (e.g., 3.7% at 16 Mbps) and increases rapidly with higher data rates. This suggests that the performance of TCP could be severely degraded by the frequent packet losses.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide wherein said latency sensitive audio data packets have a round trip time latency limit of 25 milliseconds for wireless transfer from a conference unit to said access point, and wireless transfer from said access point to said conference unit, as taught by Lee in the combined system of Solum, Kuwelkar, Haartsen, and Eidson, so that packet loss rate can be lower for shorter RTT and lower data rates (Lee: Paragraphs [0061] - [0063]).
Claim 18 is rejected under 35 U.S.C. 103 as being unpatentable over Solum et al. (US2015/0201289A1) in view of Kuwelkar et al. (US2016/0080459A1), Haartsen et al. (US2009/0298420A1), Eidson et al. (US5566180A), and further in view of Xiong et al. (US2018/0295555A1).
Regarding claim 18, the combination of Solum, Kuwelkar, Haartsen, and Eidson teaches the wireless conference system according to claim 14 (see rejection for claim 14);
The combination of Solum, Kuwelkar, Haartsen, and Eidson does not explicitly teach wherein said latency sensitive audio data packets have a round trip time latency limit of 15 milliseconds for wireless transfer from a conference unit to said access point, and wireless transfer from said access point to said conference unit.
However, Xiong teaches wherein said latency sensitive audio data packets have a round trip time latency limit of 15 milliseconds for wireless transfer from a conference unit to said access point, and wireless transfer from said access point to said conference unit (Paragraph [0011]: In a double-SEQ mechanism, any lost data packet may be detected within a round trip time (RTT), and a delay is shorter compared with timeout retransmission used in the TCP. Paragraph [0073]: The RRT is Round Trip Time in full name, and refers to a time during which a data packet is transmitted once between the sending node and the receiving node in a round-trip manner. In addition, the threshold herein may be set by a user, and the threshold is set to 15 ms in this embodiment of the present application.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide wherein said latency sensitive audio data packets have a round trip time latency limit of 15 milliseconds for wireless transfer from a conference unit to said access point, and wireless transfer from said access point to said conference unit, as taught by Xiong in the combined system of Solum, Kuwelkar, Haartsen, and Eidson, so that a lost data packet may be detected within a round trip time (Xiong: Paragraphs [0011], [0073]).
Claim 19 is rejected under 35 U.S.C. 103 as being unpatentable over Solum et al. (US2015/0201289A1) in view of Kuwelkar et al. (US2016/0080459A1), Haartsen et al. (US2009/0298420A1), Eidson et al. (US5566180A), and further in view of Kondylis et al. (US6721290B1).
Regarding claim 19, the combination of Solum, Kuwelkar, Haartsen, and Eidson teaches the wireless conference system according to claim 14 (see rejection for claim 14);
The combination of Solum, Kuwelkar, Haartsen, and Eidson does not explicitly teach wherein said TDMA based wireless communication uses TDMA frames of 5 milliseconds.
However, Kondylis teaches wherein said TDMA based wireless communication uses TDMA frames of 5 milliseconds (Col 5, lines 7-13: A focus of the present invention is the facilitation of multicast streaming of real-time CBR data in a wireless ad-hoc networks. The CBR traffic class cannot tolerate delay jitter. However, it may tolerate a small amount of packet losses. In order to ensure the provisioning of the desired quality of service (QoS) by controlling delay jitter, bandwidth is reserved on the multicast structure. Col 16, lines 31-32: A TDMA frame of 5 ms was assumed.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide wherein said TDMA based wireless communication uses TDMA frames of 5 milliseconds, as taught by Kondylis in the combined system of Solum, Kuwelkar, Haartsen, and Eidson, so that bandwidth can be reserved while ensuring the quality of service (Kondylis: Col 5, lines 7-13).
Claim 20 is rejected under 35 U.S.C. 103 as being unpatentable over Solum et al. (US2015/0201289A1) in view of Kuwelkar et al. (US2016/0080459A1), Haartsen et al. (US2009/0298420A1), Eidson et al. (US5566180A), and further in view of Cahill et al. (US2012/0034937A1).
Regarding claim 20, the combination of Solum, Kuwelkar, Haartsen, and Eidson teaches the wireless conference system according to claim 14 (see rejection for claim 14);
The combination of Solum, Kuwelkar, Haartsen, and Eidson does not explicitly teach wherein said transmitter is configured to listen for interfering traffic within an assigned timeslot within a TDMA frame before transmitting an audio data packet therein.
However, Cahill teaches wherein said transmitter is configured to listen for interfering traffic within an assigned timeslot within a TDMA frame before transmitting an audio data packet therein (Paragraph [0002]: For example, a DECT system monitors channels using a least-interfered-channel/listen-before-talk algorithm to select a channel and timeslot to use. Paragraph [0033]: In part one of the maintenance scanning process, the device periodically checks for RSSI indication in each timeslot in each channel available to it defined by the native protocol. This testing allows the device to maintain a list of channel/timeslot sets which other devices in proximity using the same protocol are known to be not using. Paragraph [0038]: This allows rapid link-establishment between a headset mobile communication device and a base unit when the headset enters range, reduces the current drain necessary to periodically check for re-entering range when a headset is out of range, and minimizes audio degradation to the device caused by interference from the overlaid 802.11 system. Paragraph [0065]: A constant scan is performed for interference levels for each time slot and for each carrier. In the DECT band, the process of scanning carriers for interference levels is set forth in the DECT regulatory protocol. Referring again to FIG. 4, in this TDMA system with 10 ms transmit framing, within the transmit frame there are twenty four time slots, with twelve for transmit and twelve for receive. For any individual time slot pair interference levels in both timeslots of a transmit/receive pair are scanned.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide wherein said transmitter is configured to listen for interfering traffic within an assigned timeslot within a TDMA frame before transmitting an audio data packet therein, as taught by Cahill in the combined system of Solum, Kuwelkar, Haartsen, and Eidson, in order to minimize audio degradation caused by interfering users (Cahill: Paragraphs [0033], [0038], [0065]).
Claim 22 is rejected under 35 U.S.C. 103 as being unpatentable over Solum et al. (US2015/0201289A1) in view of Kuwelkar et al. (US2016/0080459A1), Haartsen et al. (US2009/0298420A1), Eidson et al. (US5566180A), and further in view of Aggarwal et al. (US2017/0142535A1).
Regarding claim 22, the combination of Solum, Kuwelkar, Haartsen, and Eidson teaches the wireless conference system according to claim 14 (see rejection for claim 14);
The combination of Solum, Kuwelkar, Haartsen, and Eidson does not explicitly teach wherein said one or more conference units comprise clock synchronization units, configured to actively synchronize their respective clocks with a clock in said access point based on a timestamp inserted in beacon messages regularly broadcasted by said access point.
However, Aggarwal teaches wherein said one or more conference units comprise clock synchronization units, configured to actively synchronize their respective clocks with a clock in said access point based on a timestamp inserted in beacon messages regularly broadcasted by said access point (Paragraph [0116]: The time and/or rate at which audio packets are transmitted by the source device is based on a system clock signal (e.g., a transmitter clock signal). Each of the sink devices also utilize a system clock signal (e.g., a receiver clock signal). The time and/or rate at which audio packets are played back by the sink device(s) is based on the receiver clock signal. Ideally the transmitter clock signal and the receiver clock signal should be synchronized. Paragraph [0118]: As shown in FIG. 8, source device 802 includes synchronization logic 806, and each of sink devices 804A-804E comprise synchronization logic 808. In a typical wireless network, such as a Wi-Fi network, source device 802 and each of sink devices 804A-804E are tuned to a known carrier frequency for source device 802 to reliably transmit wireless data to sink devices 804A-804E. One way that sink devices 804A-804E tune their oscillators to the same frequency as source device 802 is by source device 802 sending a beacon signal at a fixed, pre-determined time interval (e.g., every 100 ms). The beacon signal may include a timestamp indicative of the time at which the beacon signal was transmitted by source device 802. Paragraph [0119]: In accordance with an embodiment, the beacon signal may be used to determine a relationship between the transmitter clock signal of source device 102 and the receiver clock signal for each of sink devices 804A-804E and adjust the playback of audio packets at sink devices 804A-804E accordingly. For example, source device 802 may determine a first timing parameter that is indicative of a relationship of the transmitter clock signal of source device 802 and the timestamp to be included the beacon. Source device 802 includes the first timing parameter in the beacon signal and transmits the beacon signal (that also includes the timestamp) to sink devices 804A-804E. Synchronization logic 808 of each of sink devices 804A-804E receives the beacon signal and determines the timestamp and first timing parameter included therein.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide wherein said one or more conference units comprise clock synchronization units, configured to actively synchronize their respective clocks with a clock in said access point based on a timestamp inserted in beacon messages regularly broadcasted by said access point, as taught by Aggarwal in the combined system of Solum, Kuwelkar, Haartsen, and Eidson, so that the transmitter and receiver clocks of the source device and sink devices can be synchronized and if necessary, tuned appropriately (Aggarwal: Paragraphs [0116], [0118], [0119]).
Claim 23 is rejected under 35 U.S.C. 103 as being unpatentable over Solum et al. (US2015/0201289A1) in view of Kuwelkar et al. (US2016/0080459A1), Haartsen et al. (US2009/0298420A1), Eidson et al. (US5566180A), and further in view of Williams et al. (US2006/0280182A1).
Regarding claim 23, the combination of Solum, Kuwelkar, Haartsen, and Eidson teaches the wireless conference system according to claim 14 (see rejection for claim 14);
The combination of Kuwelkar, Solum, and Haartsen does not explicitly teach wherein said predetermined expected transmission delay is determined as a sum of a propagation delay, jitter, an interrupt handling delay, processing delay and clock synchronization inaccuracy.
However, Williams teaches wherein said predetermined expected transmission delay is determined as a sum of a propagation delay, jitter, processing delay and clock synchronization inaccuracy (Paragraph [0009]: The concept of a network clock has been used to address timing problems in data networks. A network clock signal is typically generated at a specific point in the network and this becomes the system time signal received by devices on the network. The system time signal is then used as a time reference for every device that receives the system time signal. Because of the topology of the network, devices at different locations on the network will receive the clock signal with a phase offset from the network clock, depending on the propagation delay from the clock to the device. A further consequence is that the different remote devices will have received clock signals that have phase offsets with respect to each other, as well as with respect to the network clock. Paragraph [0023]: The network time protocol may use a bidirectional exchange of messages to enable the calculation of the transmission delay between master and local slave clocks which can be used to calculate a more accurate estimate of the local clock offset. This enables the local clocks to compensate for variable network delay and achieve tighter synchronization regardless of their location in the network topology. Paragraph [0027]: The playout time for media signals must take account of network transmission delays, network time protocol synchronization errors, media clock synthesis errors, sender timer jitter and network jitter. These are all factors which may delay or produce the appearance of the delay in the receiving of media packets. The playout time of received media signals must be delayed enough to allow for late arrival of media packets due to any of these causes since if a playout time is selected that is too early, any delayed media packets containing media signals that must be played out in synchronization with the received media signals will not be available for playout when required. Paragraph [0048]: he device 111 shown in FIG. 2 is able to perform the functions of both network devices 108 and 110. Further, this network device 111 can be used for processing media signals in a digital form. In this case media packets are received from one or more senders at the network port 144 and processed within the network device 111 at processor 146. The timestamps of the received media packets are used to align the digital media signals of the packets in time, if necessary.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide wherein said predetermined expected transmission delay is determined as a sum of a propagation delay, jitter, processing delay and clock synchronization inaccuracy, as taught by Williams in the combined system of Solum, Kuwelkar, and Haartsen so that the transmission delay can be determined to estimate the playout time (Williams: Paragraph [0027]);
Williams does not explicitly teach an interrupt handling delay.
However, Eidson teaches an interrupt handling delay (Col 2, lines 11-19: However, as shown in FIG. 1, this technique may remove the effects of the operating system but does not remove the jitter and latency of the protocol stack of the communication system. Implementing the synchronization unit in a microprocessor may introduce jitter of its own due to operating system or interrupt behavior of the microprocessor. This system also introduces an unknown latency within the synchronization unit itself.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide an interrupt handling delay, as taught by Eidson in the combined system of Solum, Kuwelkar, and Williams, so that the delay due to the interrupt behavior of the microprocessor can be factored into determining the correction for clock synchronization (Eidson: Col 2, lines 1-19).
Claim 24 is rejected under 35 U.S.C. 103 as being unpatentable over Solum et al. (US2015/0201289A1) in view of Kuwelkar et al. (US2016/0080459A1), Haartsen et al. (US2009/0298420A1), Eidson et al. (US5566180A), Williams et al. (US2006/0280182A1), and further in view of Lopez et al. (US2023/0199842A1).
Regarding claim 24, the combination of Solum, Kuwelkar, Haartsen, Eidson, and Williams teaches the wireless conference system according to claim 23 (see rejection for claim 23);
The combination of Solum, Kuwelkar, Haartsen, Eidson, and Williams does not explicitly teach wherein said jitter delay comprises a listen-before-talk jitter contribution.
However, Lopez teaches wherein said jitter delay comprises a listen-before-talk jitter contribution (Paragraph [0065]: Also generally, references to a listen-before-talk (LBT) procedure are meant to include any procedure where a transmitter is required to perform measurements to determine that the channel is available (e.g., idle) before starting to transmit. Examples include carrier sense multiple access with collision avoidance (CSMA/CA). Paragraph [0068]: FIG. 1 illustrates an example method 100 according to some embodiments. The method is for a transmitter configured to transmit a physical layer packet in accordance with a listen-before-talk procedure. Paragraph [0196]: Ideally, a sensing STA would receive an LTF at each specified time (e.g., with a fixed periodicity). This is, however, typically difficult to achieve since channel access in WLAN is based on an LBT procedure (CSMA/CA) with random backoff. For this reason, there will typically be some jitter in the transmission times of LTFs; in relation to the specified times.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide wherein said jitter delay comprises a listen-before-talk jitter contribution, as taught by Lopez in the combined system of Solum, Kuwelkar, Haartsen, Eidson, and Williams, so that the jitter delay due to listen-before-talk procedure can be taken into account for determining transmission delay (Lopez: Paragraphs [0004], [0065], [0068, [0196]).
Claim 25 is rejected under 35 U.S.C. 103 as being unpatentable over Solum et al. (US2015/0201289A1) in view of Kuwelkar et al. (US2016/0080459A1), Haartsen et al. (US2009/0298420A1), Eidson et al. (US5566180A), and further in view of Shi et al. (US2023/0069653A1).
Regarding claim 25, the combination of Solum, Kuwelkar, Haartsen, and Eidson teaches the wireless conference system according to claim 14 (see rejection for claim 14);
The combination of Solum, Kuwelkar, Haartsen, and Eidson does not explicitly teach wherein said predetermined expected transmission delay is set at a value between 1.5 milliseconds and 2 milliseconds.
However, Shi teaches wherein said predetermined expected transmission delay is set at a value between 1.5 milliseconds and 2 milliseconds (Paragraph [0037]: As shown in FIG. 2 , this application relates to a first device and a second device. The first device and the second device are connected to each other through a wireless channel. The first device may be configured to perform encoding to obtain audio data, and then sends the audio data to the second device through the wireless channel. The second device may be configured to decode the audio data, and then plays decoded audio. Paragraph [0040]: A connection manner between the first device and the second device in this application may include a wireless fidelity (wireless fidelity, Wi-Fi) connection, a Bluetooth connection, and the like. Paragraph [0119]: Step S23: The wireless headset determines a transmission delay of the first audio data. Paragraph [0120]: The transmission delay is, for example, 1.5 ms. Paragraph [0127]: In addition, that the transmission delay is less than 2 ms indicates that Bluetooth signal quality is good.)
Therefore, it would have been obvious to one of ordinary skill in the art before the effective filing date of the claimed invention to provide wherein said predetermined expected transmission delay is set at a value between 1.5 milliseconds and 2 milliseconds, as taught by Shi in the combined system of Solum, Kuwelkar, Haartsen, and Eidson, so that the first device can determine the bit rate for transmission, based on the transmission efficiency indicated by a transmission delay between 1.5 ms and 2 ms, which indicates good signal quality (Shi: Paragraphs: [0007], [0119], [0120], [0127]).
Response to Arguments
Applicant's arguments filed December 08, 2025 with respect to Interpretation of claim features under 35 U.S.C. § 112(f) for claim 14 as invoking 35 U.S.C. 112(f) or pre-AIA 35 U.S.C. 112, sixth paragraph has been fully considered. Claim 14 does not invoke and has not been interpreted according to 35 U.S.C. 112(f).
Applicant's arguments filed December 08, 2025 with respect to claims 14, 21, and 26 being rejected under 35 U.S.C. 103 as being unpatentable over Solum (US2015/0201289) in view of Kuwelkar (US2016/0080459); claims 15 and 16 being rejected under 35 U.S.C. 103 as being unpatentable over Solum in view of Kuwelkar and further in view of Gilson (US2020/0243103); claim 17 being rejected under 35 U.S.C. 103 as being unpatentable over Solum in view of Kuwelkar and further in view of Lee (US2013/0163428); claim 18 being rejected under 35 U.S.C. 103 as being unpatentable over Solum in view of Kuwelkar and further in view of Xiong (US2018/0295555); claim 19 being rejected under 35 U.S.C. 103 as being unpatentable over Solum in view of Kuwelkar and further in view of Kondylis (US6,721,290); claim 20 being rejected under 35 U.S.C. 103 as being unpatentable over Solum in view of Kuwelkar and further in view of Cahill (US2012/0034937); claim 22 being rejected under 35 U.S.C. 103 as being unpatentable over Solum in view of Kuwelkar and further in view of Aggarwal (US2017/0142535); claim 23 being rejected under 35 U.S.C. 103 as being unpatentable over Solum in view of Kuwelkar and further in view of Williams (US2006/0280182) and Eidson (US 5,566,180); claim 24 being rejected under 35 U.S.C. 103 as being unpatentable over Solum in view of Kuwelkar and further in view of Williams, Eidson, and Lopez (US2023/0199842); claim 25 being rejected under 35 U.S.C. 103 as being unpatentable over Solum in view of Kuwelkar and further in view of Shi (US2023/0069653) have been fully considered.
Applicant submits that the combination of Solum and Kuwelkar fails to disclose or suggest the features of amended independent claim 14 which recites in part: "wherein said access point and said one or more conference units comprise respective clocks that are actively synchronized, a clock of said respective clocks being configured to generate a local audio clock signal used locally for processing said audio data packets and a local synchronization clock signal used for said TDMA based wireless communication; wherein said receiver comprises a packet loss detection unit configured to detect loss of an audio data packet transmitted from a conference unit to said access point or vice-versa, said packet loss detection unit comprising: an estimated time of arrival unit configured to determine an expected arrival time for said audio data packet from said local synchronization clock signal by increasing the time said audio data packet is ready for transmission at said conference unit with a predetermined expected transmission delay, the predetermined expected transmission delay being an acceptable time required for interrupt handling at the transmitter and receiver side, propagation through the air, and/or possible jitter, and a packet loss detection unit configured to verify if an audio data packet is received by said expected arrival time and to detect that said audio data packet is lost if it has not arrived by said expected arrival time."
Solum teaches that the processing circuit needs to be updated as to the status of a packet, such as whether a packet was empty, missed, or received with errors, and determine whether a packet loss concealment strategy should be employed to substitute the missing audio with silence, a previously received packet, or an interpolated packet based on linear prediction. Solum teaches that missed packets are much more easily identified by the radio which has employed time division multiple access techniques, which have inherently good timing mechanisms with which to schedule packet arrivals. When a scheduled packet is not received, the radio will immediately recognize this event and informs the DSP of the missing packet (Para [0029]).
Kuwelkar teaches that the presentation time may be utilized to synchronize the receive media clock with the transmission media clock. If the network delay is known by both the talker and the listener, the listener may compare a receive time with an expected receive time (e.g., based on a known transmission delay and the presentation time) and adjust the receive media clock based on a calculated error (e.g., the difference between the measured receive time and the expected receive time) (Parag [0030]).
Haartsen et al. (US2009/0298420A1) teaches a local synchronization clock signal used for said TDMA based wireless communication to determine an expected arrival time for said audio data packet from said local synchronization clock signal by increasing the time said audio data packet is ready for transmission at said conference unit with the predetermined expected transmission delay being an acceptable time required for propagation through the air, and/or possible jitter. Haartsen teaches that the packets of audio data are transmitted from the audio source device to different ones of the speaker devices through corresponding different sequential communication frames of a time divisional multiple access (TDMA) network (Para [0008]). Haartsen teaches that the common network clock is used to compensate for drift that occurs over time between the local clock device, and local clock circuits of other associated speaker devices. A time offset is used to align the adjusted clock signal with signaling of the Bluetooth piconet to synchronize the timing of the audio output from the slave speaker devices (Para [[0060]). Hartsen also teaches that the arrival of an audio data packet may be accurately determined using a correlation unit (Para [0074]). Haartsen further discloses that the distance between distance between the devices can affect the amount of time delay between arrival of corresponding audio packets. Further, timing offsets due to jitter in the delay between the audio from the speakers needs to be controlled in order to avoid accumulative skew over time (Para [0042], [0043]).
Eidson et al. (US5566180A), teaches that a delay being an acceptable time required for interrupt handling at the transmitter and receiver side can be introduced due to the interrupt behavior of the microprocessor. Which can introduce a latency within the synchronization unit (Col 2, lines 11-19).
Thus, the combination of Solum, Kuwelkar, Haartsen, and Eidson teaches amended independent claim 14, and also amended independent claim 26 which recites similar features.
Dependent claims 15-25, and 27-30 are also taught by a combination of the cited references.
Conclusion
Applicant's amendment necessitated the new ground(s) of rejection presented in this Office action. Accordingly, THIS ACTION IS MADE FINAL. See MPEP § 706.07(a). Applicant is reminded of the extension of time policy as set forth in 37 CFR 1.136(a).
A shortened statutory period for reply to this final action is set to expire THREE MONTHS from the mailing date of this action. In the event a first reply is filed within TWO MONTHS of the mailing date of this final action and the advisory action is not mailed until after the end of the THREE-MONTH shortened statutory period, then the shortened statutory period will expire on the date the advisory action is mailed, and any nonprovisional extension fee (37 CFR 1.17(a)) pursuant to 37 CFR 1.136(a) will be calculated from the mailing date of the advisory action. In no event, however, will the statutory period for reply expire later than SIX MONTHS from the mailing date of this final action.
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/L.C./Examiner, Art Unit 2461
/KIBROM T HAILU/Primary Examiner, Art Unit 2461