Abstract: This disclosure describes one or more implementations of a video inference system that utilizes machine-learning models to efficiently and flexibly process digital videos utilizing various improved video inference architectures. For example, the video inference system provides a framework for improving digital video processing by increasing the efficiency of both central processing units (CPUs) and graphics processing units (GPUs). In one example, the video inference system utilizes a first video inference architecture to reduce the number of computing resources needed to inference digital videos by analyzing multiple digital videos utilizing sets of CPU/GPU containers along with parallel pipeline processing. In a further example, the video inference system utilizes a second video inference architecture that facilitates multiple CPUs to preprocess multiple digital videos in parallel as well as a GPU to continuously, sequentially, and efficiently inference each of the digital videos.

Abstract: A video stream scheduling unit may schedule resource allocations for each video frame of a plurality of video frames of video streams from a plurality of video sources based on a compressed frame type of the video frame, determine that a total frequency bandwidth of scheduled resource allocations for the frames concurrently due for transmission is greater than or equal to a threshold bandwidth, and receive the video streams from the plurality of video sources based on the scheduled resource allocations. The scheduling unit may delay or cancel a video frame with low priority or may instruct a video source to increase the compression rate of the video stream.

Abstract: “Binge watching” of multiple episodes of a program is improved by the player device automatically skipping repeated portions of the program. Opening and closing credit scenes, for example, can be automatically skipped to thereby allow the viewer to progress through the entire season of programming at an even faster rate than was previously thought possible. Programming to be skipped may be identified by detecting audio or other digital fingerprints in the content itself, for example. Content to be skipped may be identified to the playback device according to presentation time stamp (PTS) or other time markers.

Abstract: A vehicle control system comprises one or more mobile devices that perform wireless communication, and a vehicle control device that sets a maximum value of processing delay times obtained from the one or more mobile devices to a reference time, calculates a first time point at which an image is to be output, a second time point at which sound is to be output, a third time point at which sound is to be transmitted to the one or more mobile devices based on the reference time, and performs control to output the image at the first time point, output the sound at the second time point, and transmit the sound at the third time point.

Abstract: Provided are a wireless network access method and apparatus. The method includes: in a wireless network supporting a plurality of channel access approaches, using an access point to maintain a status of a traffic-stream-based or station-based traffic connection with a station; and using the access point to negotiate with the station or transmit control information to the station and to change a channel access approach used by the traffic-stream-based or station-based traffic connection.

Abstract: A jitter determination method for determining jitter associated with a packet-based data stream is described. The jitter determination method includes, for example, receiving at least two data packets associated with the packet-based data stream; selecting one data packet of the at least two data packets to be a reference data packet; determining initial jitter data based on the at least two data packets, wherein the initial jitter data comprises information on actual arrival times of the at least two data packets, respectively; determining expected arrival times for the at least two data packets, respectively; determining a time bias associated with the packet-based data stream based on at least the determined expected arrival time and based on at least the actual arrival time of the data packet not selected to be the reference data packet; and determining corrected jitter data based on the initial jitter data and based on the determined time bias.

Abstract: Embodiments herein relate to replacing a legacy Pick environment with a modern microservice architecture. A legacy database and a modern database may be operated in parallel for data validation. During the validation process, the legacy database may be used as the master copy of the data. After verifying that the modern database satisfies the data needs of a system, the system can switch to using the new modern database as the master copy.

Abstract: According to at least one embodiment, a method for determining a level of synchronicity between subtitle text in audiovisual content and speech that would be heard during display of the audiovisual content includes: accessing a first binary sequence, each bit of the first binary sequence indicating whether the speech is provided at a respective sampling time of a plurality of sampling times; and accessing a second binary sequence, each bit of the second binary sequence indicating whether the subtitle text is provided at a respective sampling time of a plurality of sampling times. The method further includes comparing the first binary sequence and the second binary sequence to determine the level of synchronicity between the subtitle text and the speech.

Abstract: A system includes a first module with a first clock; a second module with a second clock; and an Ethernet network interconnecting the first module and the second module by N Ethernet paths, N?2; wherein the first module is configured to provide timestamps encapsulated in replicated Ethernet packets to the second module over each of the N Ethernet paths for redundancy. The first module can be configured to obtain timestamps from a first clock with each timestamp having a sequence identifier, replicate each timestamp and its sequence identifier, and encapsulate each replicated timestamp and its sequence identifier in an Ethernet packet and transmit each Ethernet packet over one of the N Ethernet paths. The second module can be configured to receive Ethernet packets over the N Ethernet paths; and utilize a first Ethernet packet with a given sequence identifier for synchronization of the second clock with the first clock.

Abstract: Methods, systems, and media for modifying the presentation of video content on a user device based on a consumption mode of the user device are provided. In some embodiments, a computer-implemented method comprises: receiving, from a user device, a request to present a video content item, wherein the video content item includes an audio portion; identifying a segment of the audio portion as including a particular type of audio content (e.g., silent content, music content, speech content, theatrical content, non-musical content, etc.); determining a consumption mode associated with the user device; generating a modified video content item based on the consumption mode associated with the user device by removing a portion of the video content item that corresponds to the segment of the audio portion identified as including the particular type of audio content; and causing the modified video content item to be presented on the user device.

Abstract: According to the present embodiment, there is provided a content delivery apparatus applied to a content delivery system which delivers content, which is sent from a transmission apparatus, to a reception apparatus by a multicast method via an IP network. The content delivery apparatus includes an analyzer configured to detect a start point or an end point of the content by analyzing control information of the content, and a manager configured to control a reception start or a reception end of the content, based on the start point or the end point.

Abstract: An interrupt signal is provided to a guest operating system. A bus connected module is operationally connected with a plurality of processors via a bus attachment device. The bus attachment device receives an interrupt signal from the bus connected module with an interrupt target ID identifying one of the processors assigned for use by the guest operating system as a target processor for handling the interrupt signal. The bus attachment device checks whether the target processor is running using a running indicator provided by an interrupt table entry stored in a memory operationally connected with the bus attachment device. If the target processor is running, the bus attachment device forwards the interrupt signal to the target processor for handling. A translation of the interrupt target ID to a logical processor ID of the target processor is used to address the target processor directly.

Abstract: Disclosed herein are new techniques carried out by a computing system for determining delays of various components of an audio system to allow for accurate correction of these delays, which may improve the audio quality of live performances for listeners who hear audio reproduced by loudspeakers at live performance venues. In one implementation the computing system, which may comprise a transmitter device and one or more receiver devices, may be configured to perform functions, including receiving a first audio signal, receiving, via an audio input interface of the receiver, a second audio signal, and determining, based on the first audio signal and the second audio signal, an audio delay that is associated with the second audio signal. The computing system may be configured to perform further functions, including based on a determined cross-correlation between a downsampled audio signal and a filtered second audio signal, determining the audio signal delay.

Abstract: A method for detecting an available bandwidth based on a burstiness of a key frame of a video stream, includes: establishing a video connection between a sender and a receiver, and sending the video stream by the sender to the receiver; recording a sending timestamp of each packet at the sender; recording a receiving timestamp of each packet received by the receiver; calculating a difference between the receiving timestamp and the sending timestamp of each packet, and obtaining the packet with the smallest difference and the packet with the greatest difference by traversing the packets in the key frame; and estimating a currently available bandwidth according to the packet with the smallest difference and the packet with the greatest difference.

Abstract: Example systems, apparatus, and methods receive audio information including a plurality of frames from a source device, wherein each frame of the plurality of frames includes one or more audio samples and a time stamp indicating when to play the one or more audio samples of the respective frame. In an example, the time stamp is updated for each of the plurality of frames using a time differential value determined between clock information received from the source device and clock information associated with the device. The updated time stamp is stored for each of the plurality of frames, and the audio information is output based on the plurality of frames and associated updated time stamps. A number of samples per frame to be output is adjusted based on a comparison between the updated time stamp for the frame and a predicted time value for play back of the frame.

Abstract: “Binge watching” of multiple episodes of a program is improved by the player device automatically skipping repeated portions of the program. Opening and closing credit scenes, for example, can be automatically skipped to thereby allow the viewer to progress through the entire season of programming at an even faster rate than was previously thought possible. Programming to be skipped may be identified by detecting audio or other digital fingerprints in the content itself, for example. Content to be skipped may be identified to the playback device according to presentation time stamp (PTS) or other time markers.

Abstract: Exemplary methods include a processor executing instructions stored in a memory for generating an electronic count-in, binding it to a first performance to generate a master clock and transmitting a first musician's first performance and first timing information to a network caching, storage, timing and mixing module. The first musician's first performance may be recorded locally at full resolution and transmitted to a full resolution media server and the first timing information may be transmitted to the master clock. The first musician's first performance is transmitted to a sound device of a second musician and the second musician creates a second performance, transmits it and second timing information to a network caching, storage, timing and mixing module. The first and second performances are mixed along with the first and the second timing information to generate a first mixed audio, which can be transmitted to a sound device of a third musician.

Abstract: Example systems, apparatus, and methods receive audio information including a plurality of frames from a source device, wherein each frame of the plurality of frames includes one or more audio samples and a time stamp indicating when to play the one or more audio samples of the respective frame. In an example, the time stamp is updated for each of the plurality of frames using a time differential value determined between clock information received from the source device and clock information associated with the device. The updated time stamp is stored for each of the plurality of frames, and the audio information is output based on the plurality of frames and associated updated time stamps. A number of samples per frame to be output is adjusted based on a comparison between the updated time stamp for the frame and a predicted time value for play back of the frame.

Abstract: A method, includes: storing at least one set of data in a memory space, wherein the at least one set of data stored has a memory footprint in the memory space; and coupling, to the at least one set of data, a respective counter indicative of the at least one set of data, wherein the respective counter is embedded in the at least one set of data without increasing the memory footprint in the memory space.

Abstract: A method includes: obtaining a video data file and an audio data file of streaming media content to be played from a server, when the streaming media content is played using DASH; obtaining a video bitrate from the video data file, and obtaining an audio bitrate from the audio data file; and determining a capacity of a video buffer area and a capacity of an audio buffer area according to the video bitrate and the audio bitrate.

Abstract: The present invention provides an apparatus, system and method for reproducing data in groups. According to the apparatus, system and method for reproducing data in groups, some piece of data of sound source data that is already being reproduced through a specific terminal or external speaker is input through a microphone and compared with the entire data of the sound source data, thus to identify a current reproduction point, and synchronize the reproduction point with sound source data being played externally, thereby reproducing the sound source data at the same time.

Abstract: An IP network includes a central entity and at least one customer premises equipment (CPE) device. The central entity generates a program clock reference (PCR) clock and provides audio-visual packets to a CPE based on the PCR clock. The CPE sets a first clock based on the PCR clock for decoding operations. The CPE sets a second clock that is independent from the first clock for audio and video output operations. For example, the CPE can process the audio-visual packets using the second clock.

Abstract: A method of encoding a local time base embedded in the compressed data is disclosed. The local time base is encoded in two parts. The first part has a modulo time base that indicates the specific interval in the reference time base and the second part has a time base increment relative to the reference time. Two forms of time base increment is used to allow for the possibility of different encoding order and displaying order. A mechanism for the synchronization of multiple compressed streams with local time base is also described. A time base offset mechanism is also used to allow finer granularity synchronization of the multiple compressed streams.