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 .
DETAILED ACTION
This action is in response to application filed 09/17/2024.
Claims 1-20 are pending in this application.
Information Disclosure Statement
The information disclosure statement (IDS) submitted on 09/17/2024, 11/20/2025 has been placed in record and considered by the examiner.
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 of this title, 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 set forth in Graham v. John Deere Co., 383 U.S. 1, 148 USPQ 459 (1966), that are applied 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 1-3, 9-13, 19-20 are rejected under 35 U.S.C. 103 as being unpatentable over Krishnaswamy (US 2007/0076693 A1) in view of Demircin et al. (US 2006/0095944 A1)
Regarding claim 1, Krishnaswamy discloses a method performed by a processing device, comprising:
acquiring a to-be-transmitted data frame, the to-be-transmitted data frame having a corresponding maximum transmission delay ([0028]: ensures that performance and delay constraints related to the delivery of the multimedia stream can be satisfied);
determining, based on a data volume of the to-be-transmitted data frame and an initial data transmission parameter, a first estimated transmission duration corresponding to the to-be-transmitted data frame ([0035]: the multimedia data to be transmitted is a real-time video stream, the transmission rate estimator may determine an initial set of transmitting bit rates for each transmitting node by using the information such as the size of packets/frames and the upper time limit within which each packet/frame has to be transmitted to the next node in order for the destination node to receive the video stream with a reasonable quality), the initial data transmission parameter being a data transmission parameter for transmitting the to-be-transmitted data frame ([0035]-[0036]: Each initial set of transmitting bit rates includes at least two rates: one represents a high bit rate for those large packets/frames and one represents a low bit rate for other packets/frames. Each initial set of transmitting bit rates may include many different rates);
adjusting, in response to the first estimated transmission duration exceeding a maximum transmission delay, the initial data transmission parameter by using reducing estimated transmission duration as an adjustment goal to obtain an actual data transmission parameter ([0028]: Equation (6) is a more general constraint for wireless transmissions in the network. Equation (3) ensures that performance and delay constraints related to the delivery of the multimedia stream can be satisfied. Equation (6) ensures that all links are able to meet their bandwidth requirements in every fractional period of time ΔTj. In the event that Equation (3) or Equation (6) is not satisfied, the values of the bandwidths in the set {B1, B2, . . . Bk−1} may be scaled down to lower values such that the equations is satisfied .[0036]: The transmission rate estimator may further adjust values of the high bit rate and the low bit rate for each transmitting node according to the available bandwidth sustainable by each link in the transmission path (e.g., based on Equations (3) and (6) above), and the estimated quality of the multimedia data received by the destination node, provided by the quality estimator); and
transmitting, based on the actual data transmission parameter, the to-be-transmitted data frame to a receive side of the to-be-transmitted data frame ([0041]: the data stream may need to be adjusted to reflect the change of the available bandwidth of the path. The produced data stream may then be transmitted directly from the source node to the destination node).
However, Krishnaswamy does not disclose an estimated transmission duration for the to-be-transmitted data frame determined based on the actual data transmission parameter being a second estimated transmission duration, and the second estimated transmission duration not exceeding the maximum transmission delay.
In an analogous art, Demircin discloses an estimated transmission duration for the to-be-transmitted data frame determined based on the actual data transmission parameter being a second estimated transmission duration, and the second estimated transmission duration not exceeding the maximum transmission delay ([0295]: to adapt the audio/video stream in real time, based on the varying network conditions and audio/video content characteristics. [0296]: Audio/video data unit has an associated delivery deadline, substantially resulting in a limit or constraint on the delay incurred by this data unit between the time of encoding (or transcoding) and the time of decoding. In delay-constrained audio/video transmission, the audio/video stream is adapted in such a manner that substantially all the data units arrive at the receiver before their respective delivery deadlines).
Therefore, it would have been obvious before the effective filed date of the claimed invention to a person having ordinary skill in the art to modify Krishnaswamy to comprise “an estimated transmission duration for the to-be-transmitted data frame determined based on the actual data transmission parameter being a second estimated transmission duration, and the second estimated transmission duration not exceeding the maximum transmission delay” taught by Demircin.
One of ordinary skilled in the art would have been motivated because it would have enabled to dynamically adapt the bit rate of a bit stream to the conditions of the channel (Demircin, [0189]).
Regarding claim 2, Krishnaswamy-Demircin discloses the method according to claim 1, wherein: the data transmission parameter comprises a packet sequence length and a packet sequence transmission interval, the packet sequence length identifying a quantity of to-be-transmitted data packets comprised in each packet sequence when a to-be-transmitted data packet is transmitted in a packet sequence (Demircin, [0205]: sending packets in such bursts is that the overall throughput of the system is approximately equal to the target bit rate of the streaming video. The effective throughput, E, can be modified by controlling the following three parameters: (1) The packet size (e.g., in the number of bytes), or the size of the data payload of each packet;(2) The number of packets in each burst of packets;(3) The time interval between subsequent burst of packets. [0212]: The packet payload may also include a burst sequence number. Such sequence numbers may be used by the receiver to detect packet losses) and the to-be-transmitted data packet is obtained by splitting the to-be-transmitted data frame based on a unit data volume of a data packet (Demircin, [0156] : me. A normal MPEG-2 GOP (Group-of-Pictures) of a single stream contains a number of I, P and B-type frames. A super GOP is formed over multiple MPEG-2 streams and consists of NGOP super frames, where a super frame is a set of frames containing one frame from each stream and all frames in a super frame coincide in time); determining the first estimated transmission duration corresponding to the to-be-transmitted data frame comprises: determining, based on the data volume of the to-be-transmitted data frame, the unit data volume of a data packet, an initial packet sequence length, and an initial packet sequence transmission interval, the first estimated transmission duration corresponding to the to-be-transmitted data frame (Demircin, [0205]: (1) The packet size (e.g., in the number of bytes), or the size of the data payload of each packet; (2) The number of packets in each burst of packets; (3) The time interval between subsequent burst of packets. [0237]: The input video 300 is provided to a MPEG transcoder 310 which adapts the video bit rate to the desired data sending rate for one or more video streams); and transmitting the to-be-transmitted data frame to the receive side comprises: splitting the to-be-transmitted data frame based on the unit data volume of a data packet to obtain a plurality of to-be-transmitted data packets corresponding to the to-be-transmitted data frame (Demircin, [0370]: A plurality of data packets may be grouped into a “burst”. The system generally refers to “data frames” as packets that are exchanged between the source and destination); generating, based on an actual packet sequence length in the actual data transmission parameter, a plurality of to-be-transmitted packet sequences corresponding to the plurality of to-be-transmitted data packets; and transmitting the plurality of to-be-transmitted packet sequences to the receive side based on an actual packet sequence transmission interval in the actual data transmission parameter (Demircin, [0237]: the input video 300 is provided to a MPEG transcoder 310 which adapts the video bit rate to the desired data sending rate for one or more video streams. The rate-reduced (or increased) MPEG video 320 is provided to a packetizer and packet scheduler 330 which schedules packets for transmission over the channel, adapting the data sending rate so that is preferably remains generally below the available bandwidth). The same rationale applies as in claim 1.
Regarding claim 3, Krishnaswamy-Demircin discloses the method according to claim 2, wherein the to-be-transmitted data frame is generated by an encoder, and the method further comprises: acquiring packet sequence reception durations corresponding to a plurality of received packet sequences within a measurement period, the plurality of received packet sequences being a packet sequences received by the receive side (Krishnaswamy, [0042]: the size of frames/packets in the multimedia data and the transmission time period of a frame/packet which is permissible to ensure an acceptable quality of the multimedia data at the destination node may be used to determine an initial set of bit rates for each frame/packet in the multimedia data for each transmitting node); determining, based on the packet sequence reception durations and packet sequence lengths corresponding to the plurality of received packet sequences, data packet reception speeds corresponding to the plurality of received packet sequences, each of the data packet reception speeds corresponding to a respective received packet sequence in the plurality of received packet sequences (Krishnaswamy, [0028]: Equation (6) is a more general constraint for wireless transmissions in the network. Equation (3) ensures that performance and delay constraints related to the delivery of the multimedia stream can be satisfied. [0042]: For example, if a link cannot sustain the high bit rate, the high bit rate for that link may be adjusted to the bandwidth sustainable by the link and the time period of using the high bit rate may be adjusted according to Equations (3) and (6)); determining a target bit rate based on the data packet reception speeds; and adjusting a bit rate corresponding to the encoder to the target bit rate, so that the encoder generates the to-be-transmitted data frame based on the target bit rate (Krishnaswamy, [0037]: Quality estimator 540 estimates the quality of the multimedia data, when received by the destination node, according to the transmission bit rates used by each transmitting node. The estimated quality may be used by the transmission rate estimator to determine the initial set of transmitting bit rate and to adjust the values of the transmitting bit rates for each transmitting node. The estimated quality may also be used by data stream producer & regulator to produce/regulate the data stream to be transmitted across the selected transmission path).
Regarding claim 9, Krishnaswamy-Demircin discloses the method according to claim 3, wherein the target bit rate does not exceed a data transmission speed corresponding to a data transmission link, and the data transmission link is for transmitting the to-be-transmitted packet sequence to the receive side (Krishnaswamy, [0023]: For example, in a time interval ΔT, it is possible that a large fraction of the time may be used by the node that needs to transfer packets at a high data rate, which may involve several packet transfers; other nodes may transfer their data at a low data rate during such a period of time. Bandwidth values shown in FIGS. 4A-4C only indicate the average bandwidth requirement for each of the nodes during a time interval. In the above discussions, it is assumed that Bmax is smaller than the minimum of the maximum bandwidth sustainable on each of the links based on link conditions). The same rationale applies as in claim 3.
Regarding claim 10, Krishnaswamy-Demircin discloses the method according to claim 3, wherein the actual packet sequence length is between a lower limit threshold of the packet sequence length and an upper limit threshold of the packet sequence length, the upper limit threshold of the packet sequence length is determined based on: a data transmission speed of a data transmission link for transmitting the to-be-transmitted packet sequence, and a minimum transmission interval threshold, the packet sequence transmission interval is between the minimum transmission interval threshold and a maximum transmission interval threshold, and the maximum transmission interval threshold is a maximum possible transmission interval that still ensures that a quantity of to-be-transmitted packet sequences receivable by the receive side within the measurement period is greater than a preset quantity (Demircin, [0205]: if a payload size of 1472 bytes, and the number of packets in the burst is 10, and the time interval between bursts is 40 milliseconds, the effective throughput is: 10 (packets per burst)×1472 (bytes per packet)×8 (bits per byte)/0.040 (seconds per burst)=2,944,000 bits per second, or approximately 2.9 Mbps. Therefore, an audiovisual stream with a bit rate of 2.9 Mbps can be streamed at that rate using that wireless connection. It may be observed, that the packets are the actual video signal and not merely additional test traffic imposed on the network. In addition, the system may sequentially transmit the packet bursts in a manner such that the average data rate matches (within 10%) the video bit rate). The same rationale applies as in claim 3.
Regarding claims 11 and 19; the claims are interpreted and rejected for the same reason as set forth in claim 1.
Regarding claims 12 and 20; the claims are interpreted and rejected for the same reason as set forth in claim 2.
Regarding claim 13; the claim is interpreted and rejected for the same reason as set forth in claim 3.
Claims 4-6, 8, 14-16, 18 are rejected under 35 U.S.C. 103 as being unpatentable over Krishnaswamy in view of Demircin, as applies to claim 3, in view of Niina et al. (US 2020/0280766 A1).
Regarding claim 4, Krishnaswamy-Demircin discloses the method according to claim 3.
However, Krishnaswamy-Demircin does not disclose wherein determining the target bit rate based on the data packet reception speeds comprises: generating a reception speed distribution parameter based on the data packet reception speeds, the reception speed distribution parameter identifying a quantity of received packet sequences distributed in each of a plurality of reception speed ranges having a same range length; and determining the target bit rate based on a mode range corresponding to the reception speed distribution parameter, the mode range being a reception speed range among a plurality of reception speed ranges, in which a quantity of received packet sequences distributed is greater than quantities of received packet sequences distributed in adjacent reception speed ranges at both sides of the reception speed range.
In an analogous art, Niina discloses wherein determining the target bit rate based on the data packet reception speeds comprises: generating a reception speed distribution parameter based on the data packet reception speeds, the reception speed distribution parameter identifying a quantity of received packet sequences distributed in each of a plurality of reception speed ranges having a same range length ([0047]: the criterion determining unit 22 generates a combination of a communication state representing a high communication speed and a reception form representing a bit rate of intermediate image quality, a combination of a communication state representing an intermediate communication speed and a reception form representing a bit rate of low image quality, and a combination of a communication state representing a low communication speed and a reception form representing a bit rate of low image quality and a large buffer size as criteria); and determining the target bit rate based on a mode range corresponding to the reception speed distribution parameter, the mode range being a reception speed range among a plurality of reception speed ranges, in which a quantity of received packet sequences distributed is greater than quantities of received packet sequences distributed in adjacent reception speed ranges at both sides of the reception speed range ([0028]: For example, the criterion determining unit 22 generates a combination of a communication state representing a high communication speed (equal to or higher than 2 Mbps) and a reception form representing a bit rate (700 Kbps) of intermediate image quality, a combination of a communication state representing an intermediate communication speed (equal to or higher than 1 Mbps and lower than 2 Mbps) and a reception form representing a bit rate (350 Kbps) of low image quality, and a combination of a communication state representing a low communication speed (lower than 1 Mbps) and a reception form representing a bit rate (350 Kbps) of low image quality and a large buffer size as criteria).
Therefore, it would have been obvious before the effective filed date of the claimed invention to a person having ordinary skill in the art to modify Krishnaswamy-Demircin to comprise “wherein determining the target bit rate based on the data packet reception speeds comprises: generating a reception speed distribution parameter based on the data packet reception speeds, the reception speed distribution parameter identifying a quantity of received packet sequences distributed in each of a plurality of reception speed ranges having a same range length; and determining the target bit rate based on a mode range corresponding to the reception speed distribution parameter, the mode range being a reception speed range among a plurality of reception speed ranges, in which a quantity of received packet sequences distributed is greater than quantities of received packet sequences distributed in adjacent reception speed ranges at both sides of the reception speed range” taught by Niina.
One of ordinary skilled in the art would have been motivated because it would have enabled to receive content by the receiver in a reception form corresponding to the communication state such as communication speed (Niina, [0027]).
Regarding claim 5, Krishnaswamy-Demircin-Niina discloses the method according to claim 4, wherein the reception speed distribution parameter corresponds to only a single mode range, and wherein determining the target bit rate based on the mode range corresponding to the reception speed distribution parameter comprises: determining, in response to not acquiring a congestion signal within the measurement period, an average value of the data packet reception speeds corresponding to the plurality of received packet sequences distributed in the mode range as the target bit rate, the congestion signal identifying that an abnormality occurs in a data transmission speed corresponding to a data transmission link for transmitting the to-be-transmitted packet sequences to the receive side (Niina, [0056]: the reception control unit 212 can perform increasing of the buffer size, changing of the bit rate, and the like such that video content can be reproduced continuously before and after a change in the communication environment. In addition, in a case in which communication with a network is predicted to be disconnected over a long time, the reception control unit 212 may store content of a low bit rate. In addition, in a case in which reproduction is predicted to stop, the reception control unit 212 may control display of the display such that a guide “The communication environment is bad. Please wait for a moment” is displayed); or determining, in response to acquiring the congestion signal within the measurement period, a difference between the average value and a standard deviation of the data packet reception speeds corresponding to the plurality of received packet sequences within the measurement period as the target bit rate.
The same rationale applies as in claim 4.
Regarding claim 6, Krishnaswamy-Demircin-Niina discloses the method according to claim 4, wherein the reception speed distribution parameter corresponds a plurality of mode ranges, a quantity of the plurality of mode ranges is less than a preset quantity of ranges, and the determining the target bit rate based on a mode range corresponding to the reception speed distribution parameter comprises: determining, in response to not receiving a congestion signal within the measurement period, an average value of data packet reception speeds corresponding to a plurality of received packet sequences distributed in a first target range as the target bit rate, the first target range being a mode range corresponding to a lowest reception speed of the plurality of mode ranges, and the congestion signal being configured for identifying that an abnormality occurs in a data transmission speed corresponding to a data transmission link configured for transmitting the to-be-transmitted packet sequences to the receive side (Niina, [0028]: in a case in which the movement speed of the receiver 10 is high, the criterion determining unit 22, for example, can generate a combination of a communication state representing a high communication speed and a reception form representing a low bit rate as a criterion. In addition, the criterion determining unit 22 may generate a combination of a communication state representing a high communication speed and a reception form representing a large buffer size as a criterion. For example, the criterion determining unit 22 generates a combination of a communication state representing a high communication speed (equal to or higher than 2 Mbps) and a reception form representing a bit rate (700 Kbps) of intermediate image quality, a combination of a communication state representing an intermediate communication speed (equal to or higher than 1 Mbps and lower than 2 Mbps) and a reception form representing a bit rate (350 Kbps) of low image quality, and a combination of a communication state representing a low communication speed (lower than 1 Mbps) and a reception form representing a bit rate (350 Kbps) of low image quality and a large buffer size as criteria); or
determining, in response to receiving the congestion signal within the measurement period, a difference between the average value and a standard deviation of data packet reception speeds corresponding to a plurality of received packet sequences distributed in a second target range as the target bit rate, the second target range being obtained by partitioning using a right endpoint of a first negative mode range and a left endpoint of a second negative mode range, the first negative mode range being a negative mode range located on a left side of the first target range and closest to the first target range, the second negative mode range being a negative mode range located on a right side of the first target range and closest to the first target range, and the negative mode range being a reception speed range of the plurality of reception speed ranges, in which a quantity of received packet sequences distributed is less than quantities of received packet sequences distributed in adjacent reception speed ranges at both ends of the reception speed range.
The same rationale applies as in claim 4.
Regarding claim 8, Krishnaswamy-Demircin-Niina discloses the method according to claim 4, wherein: a quantity of a plurality of mode ranges corresponding to the reception speed distribution parameter exceeds a preset range quantity; and determining the target bit rate based on the mode range corresponding to the reception speed distribution parameter comprises: determining, in response to a quantity of the plurality of received packet sequences within the measurement period being greater than a first preset quantity and not receiving a congestion signal within the measurement period, an average value of data packet reception speeds corresponding to a plurality of received packet sequences distributed in a first target range as the target bit rate, the first target range being a mode range corresponding to a lowest reception speed of the plurality of mode ranges, and the congestion signal identifying that an abnormality occurs in a data transmission speed corresponding to a data transmission link for transmitting the to-be-transmitted packet sequence to the receive side (Niina, [0028]: in a case in which the movement speed of the receiver 10 is high, the criterion determining unit 22, for example, can generate a combination of a communication state representing a high communication speed and a reception form representing a low bit rate as a criterion. In addition, the criterion determining unit 22 may generate a combination of a communication state representing a high communication speed and a reception form representing a large buffer size as a criterion. For example, the criterion determining unit 22 generates a combination of a communication state representing a high communication speed (equal to or higher than 2 Mbps) and a reception form representing a bit rate (700 Kbps) of intermediate image quality, a combination of a communication state representing an intermediate communication speed (equal to or higher than 1 Mbps and lower than 2 Mbps) and a reception form representing a bit rate (350 Kbps) of low image quality, and a combination of a communication state representing a low communication speed (lower than 1 Mbps) and a reception form representing a bit rate (350 Kbps) of low image quality and a large buffer size as criteria); and reducing, in response to not receiving the congestion signal within the measurement period, an initial bit rate corresponding to the encoder by multiplying a preset ratio to obtain target bit rate (Niina, [0028]: a combination of a communication state representing a high communication speed and a reception form representing a low bit rate as a criterion). The same rationale applies as in claim 4.
Regarding claim 14; the claim is interpreted and rejected for the same reason as set forth in claim 4.
Regarding claim 15; the claim is interpreted and rejected for the same reason as set forth in claim 5.
Regarding claim 16; the claim is interpreted and rejected for the same reason as set forth in claim 6.
Regarding claim 18; the claim is interpreted and rejected for the same reason as set forth in claim 8.
Claims 7 and 17 are rejected under 35 U.S.C. 103 as being unpatentable over Krishnaswamy in view of Demircin in view of Niina, as applies to claim 4, in view of Dabbadi et al. (US 2026/0059130 A1 – Priority date 10/18/2022).
Regarding claim 7, Krishnaswamy-Demircin-Niina discloses the method according to claim 4.
However, Krishnaswamy-Demircin-Niina does not disclose wherein: the reception speed distribution parameter corresponds to a plurality of mode ranges, and a quantity of the plurality of mode ranges exceeds a preset range quantity, the method further comprises: performing noise smoothing processing on the reception speed distribution parameter, the noise smoothing processing being used for reducing a quantity of mode ranges in the reception speed distribution parameter; and determining the target bit rate based on the mode range corresponding to the reception speed distribution parameter comprises: determining the target bit rate based on mode ranges corresponding to a reception speed distribution parameter undergone the noise smoothing processing.
In an analogous art, Dabbadi discloses wherein: the reception speed distribution parameter corresponds to a plurality of mode ranges, and a quantity of the plurality of mode ranges exceeds a preset range quantity, the method further comprises: performing noise smoothing processing on the reception speed distribution parameter, the noise smoothing processing being used for reducing a quantity of mode ranges in the reception speed distribution parameter ([0036]: by filtering the noise out before encoding a video, generating grain parameters based on the noise that is filtered out, reconstructing the noise based on the grain parameters, and adding the reconstructed noise back into the decoded video, the encoded video can be compressed more heavily by the codec systems and techniques described herein, and still reconstruct film grain, than by codec systems that attempt to preserve film grain during encoding); and determining the target bit rate based on the mode range corresponding to the reception speed distribution parameter comprises: determining the target bit rate based on mode ranges corresponding to a reception speed distribution parameter undergone the noise smoothing processing ([0036]: This increased compression can correspond to reduced bit rate (in an illustrative example, by approximately 50%) over codec systems that attempt to preserve film grain during encoding).
Therefore, it would have been obvious before the effective filed date of the claimed invention to a person having ordinary skill in the art to modify Krishnaswamy-Demircin to comprise “wherein: the reception speed distribution parameter corresponds to a plurality of mode ranges, and a quantity of the plurality of mode ranges exceeds a preset range quantity, the method further comprises: performing noise smoothing processing on the reception speed distribution parameter, the noise smoothing processing being used for reducing a quantity of mode ranges in the reception speed distribution parameter; and determining the target bit rate based on the mode range corresponding to the reception speed distribution parameter comprises: determining the target bit rate based on mode ranges corresponding to a reception speed distribution parameter undergone the noise smoothing processing” taught by Dabbadi.
One of ordinary skilled in the art would have been motivated because it would have enabled to compress video data into a form that uses a lower bit rate, while avoiding or minimizing degradations to video quality (Dabbadi, [0003]).
Regarding claim 17; the claim is interpreted and rejected for the same reason as set forth in claim 7.
Additional References
The prior art made of record and not relied upon is considered pertinent to applicants disclosure.
Sudak et al., US 2020/0107069 A1: Systems and Methods for Reducing Latency of a Video Transmission System.
Lee et al., US 10,492,097 B2: Method and Apparatus for Controlling Congestion in Communication System.
Conclusion
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/J.C.T/Examiner, Art Unit 2446
/BRIAN J. GILLIS/Supervisory Patent Examiner, Art Unit 2446
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