415 results about "Bit rate" patented technology

The invention discloses an extremely low bit rate image compression coding and decoding method of a stream matching diffusion model. The coding method comprises the following steps: acquiring an original image, and coding the original image into a compression latent variable and a hierarchical feature; based on the compression latent variable, generating an anchor point set and a mask parameter in a continuous domain; generating a visible index, a stream matching scheduling parameter and a confidence graph according to the compression latent variable, the hierarchical feature and the target bit rate; and combining the visible index, the anchor point set, the mask parameter, the stream matching scheduling parameter and the confidence map to form a code stream. According to the invention, through continuous domain mask and confidence gating reasoning of code rate perception, the method is suitable for image compression with an extremely low bit rate.

In some embodiments, a method determines whether to use first logic or second logic to select a profile from a list of profiles to download a segment of content. When using the second logic: selecting historical information from one or more previous playback sessions; determining a bandwidth using the historical information; and selecting the profile using a bitrate that is determined based on the bandwidth. When using the first logic, selecting the profile without using the historical information.

Systems and methods are disclosed for generating an adaptive bitrate ladder for a content item based on the actual quality of the content item. A user uploaded content item is determined to have different quality scores in different time portions of the content item, where the quality score is based on an analysis of visual characteristics of frames of the content item. In response, an adaptive bitrate ladder (ABR) is generated for each time portion which contains a set of bitrates based on the quality score.

Embodiments of the present disclosure relate to report triggering. The terminal device determines that a report is to be triggered when uplink data arrives in an empty buffer in the terminal device for at least one logic channel. The terminal device triggers the report after receiving an uplink grant from a network device. In this way. an efficient mechanism to trigger a report for every data burst may be defined. Such report triggering mechanism facilitates the network device to take the right scheduling decisions even for services with high variability of the bit rate.

The present disclosure relates to a 5G or 6G communication system for supporting higher data transmission rates. In addition, the present disclosure relates to a method performed by a terminal in a wireless communication system and a device for performing same, the method comprising the steps of: transmitting, to a base station, terminal performance information including at least one of first information indicating the performance for a query of a recommended bit rate (RBR) or second information indicating performance for application of the recommended bit rate; receiving, from the base station, a radio resource control (RRC) message including configuration information related to the recommended bit rate; and receiving, from the base station, a message including information about the recommended bit rate for a quality of service (QoS) flow on the basis of the configuration information.

A point cloud encoding and decoding method and apparatus, and pertains to the field of point cloud processing technologies. The point cloud encoding method in embodiments of this application includes: An encoder determines, based on attribute information of target point cloud, a target attribute information distribution characteristic value corresponding to a target bit rate point, where the target bit rate point is a bit rate point corresponding to an attribute quantization step of the attribute information; the encoder determines a target run-length encoding manner based on the target attribute information distribution characteristic value; and the encoder performs encoding processing on the attribute information based on the target run-length encoding manner.

An aspect of the disclosure related to methods and systems configured to detect that an item of streaming primary video content associated with a channel of scheduled streaming content is designated to be streamed to a device. Ancillary content is identified. Alternative versions of respective sections of the ancillary content are identified. Metadata corresponding to the alternative versions of ancillary content, metadata corresponding to the item of primary video content, and / or metadata corresponding to the channel are used in selecting a first section from among the alternative versions of a given section. The ancillary content, including the selected first section from among the alternative versions is streamed to the device immediately before, during, or immediately after the item of streaming video content. The alternative versions may be transcoded to generate multiple encoded iterations of respective alternative versions and the encoded versions are stored in a tree data structure.

Systems and methods are disclosed for generating an adaptive bitrate ladder for a content item based on the actual quality of the content item. A broadcaster determines a quality score of a content item and multiplexes a message indicating the quality score with data packets of the content item to a headend. The headend uses the quality score to determine an ABR ladder with a set of bitrates for the content item. The headend then transcodes the content item into the set of bitrates. The headend then transmits a plurality of network addresses, each corresponding to a stream at a particular bitrate, to a user gateway to enable the user gateway to make the streams available.

Systems and methods are described for transcoding at least a portion of a live media asset ingested from a media content source. The systems and methods may be configured to, in real time, after ingesting the at least a portion of the live media asset, determine parameters of the at least a portion of the live media asset. The systems and methods may be further configured to, in real time, after ingesting the at least a portion of the live media asset, determine, based on the parameters, a plurality of optimal bitrate-resolution pairs for the at least a portion of the live media asset. The systems and methods may be further configured to, in real time, after ingesting the at least a portion of the live media asset, cause the at least a portion of the live media asset to be transcoded based on the plurality of optimal bitrate-resolution pairs.

The invention discloses a low-delay adaptive code rate control system for a meteorological monitoring video stream, and relates to the technical field of adaptive code rate control in video stream media transmission, and the system comprises a content importance analysis module which is used for carrying out the real-time analysis of a video block to be transmitted, generating a content importance score representing the meteorological information value; and the decision arbitration module is connected with the content importance analysis module and is used for receiving the content importance score, the network state information and the buffer area state information. According to the low-delay adaptive code rate control system for the meteorological monitoring video stream, bandwidth resources can be allocated according to the difference of meteorological information values of video contents, the quality of key monitoring pictures is guaranteed preferentially when network conditions fluctuate, unnecessary code rate switching is inhibited actively through internal look-ahead simulation and risk assessment, and the quality of the meteorological monitoring video stream is improved. Therefore, balance between continuous and clear transmission of high-value information and overall watching fluency is realized in a complex network environment.

A communication apparatus includes a reception unit configured to receive uplink scheduling information notifying an uplink scheduling method from a base station, a setting unit configured to set a bit rate of a plurality of video frames using information that is different depending on the uplink scheduling method, an encoding unit configured to encode the plurality of video frames at the set bit rate, and a transmission unit configured to transmit at least one video frame in the plurality of encoded video frames to the base station.

A user device may be configured to make a better choice between switching to a different content bit rate or switching to a different content server when responding to interruptions during content output, based on an awareness of a cause of the interruptions. Based on conditions associated with content delivery or information concerning playback quality at one or more user devices, an alternative bit rate ladder for encoding content may be determined. Such dynamic generation of an alternative bit rate ladder may enable fine tuning of load reductions in the content delivery network and optimized delivery of the encoded content to the user devices.

The present disclosure provides techniques for improving signaling efficiency in the context of rate control governing a desired ratio of bitrate load and quality in image compression. It is provided a method for encoding an image comprising obtaining the image, encoding the image into a bitstream based on a first coding parameter, wherein the first coding parameter is indicative of a first displacement between a first target rate control parameter and a first reference rate control parameter, and encoding the first coding parameter into the bitstream. Further, it is provided a method for decoding an image, comprising receiving a bitstream comprising coded data of the image, parsing the bitstream to obtain a first coding parameter, wherein the first coding parameter is indicative of a first displacement between a first target rate control parameter and a first reference rate control parameter, and reconstructing the image based on the first coding parameter.

The invention discloses a dynamic self-adaptive edge-cloud cooperative computing segmentation point and compression strategy adjustment method. The core of the method is to deploy a dynamic decision engine on edge equipment. The engine senses a network state (such as bandwidth and delay) and a task priority specified by an application in real time, and according to a preset decision logic (such as a look-up table or a lightweight prediction model), a deep learning model which supports multiple segmentation points and is equipped with a variable bit rate hierarchical quantization feature compression model for each segmentation point is used for performing a multi-segmentation-point hierarchical quantization feature compression model on the basis of the multi-segmentation-point hierarchical quantization feature compression model, so that the multi-segmentation-point hierarchical quantization feature compression model is obtained. And dynamically selecting an optimal model segmentation point and feature compression level combination. The combination is transmitted to the cloud, and the cloud loads the corresponding back-end model to complete calculation. Through the dynamic adjustment mechanism, the real-time optimal balance among the reasoning delay, the task precision and the network overhead is realized, and the performance stability, the resource utilization rate and the adaptability of the system in a variable environment are remarkably improved.

The invention discloses an intelligent super-division coding system for a low-bit-rate video, which comprises multiple stages of assembly lines and a merging module, each stage of the multiple stages of assembly lines is responsible for different GPUs / threads, and the multiple stages of assembly lines are integrated into a pipeline parallel task. The multi-stage assembly line comprises a video input and analysis module, a task segmentation and dynamic scheduling module, an AI super-division module, an HDR / SDR self-adaptive conversion module, a video coding module and a quality control and closed-loop optimization module which are sequentially processed. And time consistency and efficient utilization are ensured through dynamic scheduling, overlapping slicing and inter-frame constraint on multiple GPU hosts, so that the system can still improve subjective and objective image quality on the premise of remarkably reducing the code rate.

A commander substation for a serial bus system and a method for communication in a serial bus system are provided. The commander substation has a frame validity check block for checking whether a frame corresponding to a received signal is equal to a predetermined frame and thus error-free and therefore valid. The frame validity check block is configured to perform a first check to determine whether the frame corresponding to the received signal is valid if the bit time of the first communication phase of the predetermined frame has a predetermined value corresponding to a bit rate that is equal to or less than a predetermined bit rate. The frame validity check block is configured to perform a second check to determine whether the frame corresponding to the received signal is valid.if the bit time of the first communication phase of the predetermined frame has a predetermined value corresponding to a bit rate that is greater than the predetermined bit rate, and where the first check and the second check are at least partially different.

Various aspects of the present disclosure generally relate to wireless communication. In some aspects, a user equipment (UE) may receive a configuration message conveying recommended bit rate (RBR) information, wherein the configuration message includes a type indicator that indicates whether the RBR information is to be interpreted as being an enhanced type of RBR information or a non-enhanced type of RBR information. The UE may communicate extended reality (XR) traffic in accordance with a data stream configuration of the RBR information and based on the type indicator indicating the RBR information is to be interpreted as the enhanced type of RBR information. Numerous other aspects are described.

The invention relates to a method for processing a received signal derived from two continuous phase-modulated signals, comprising the following steps: frequency shifting (102) of the squared received signal so as to produce: a first and a second shifted signal having a spectrum shifted by D / 2 and -D / 2 respectively with respect to a spectrum of the squared signal, where D is a bit rate of each of the two continuous phase-modulated signals; determination (108): of a first and a second frequency at which a power spectral density of the first and second shifted signals takes a first and second maximum value respectively, of a third frequency at which a power spectral density of the squared signal takes a third maximum value; central tendency measurement of the frequencies to estimate (110) a frequency difference between the received signal and each of the two continuous phase-modulated signals. Figure for the abstract: Fig. 3

Systems and methods are disclosed for generating an adaptive bitrate ladder for a content item based on the actual quality of the content item. A broadcaster determines a quality score of a content item and multiplexes a message indicating the quality score with data packets of the content item to an OTT provider. The OTT provider uses the quality score to determine an ABR ladder with a set of bitrates for the content item. The OTT provider then encodes a plurality of portions of the content item for each bitrate. The OTT provider then generates a manifest indicating a portion of a URL corresponding to each portion and transmits the manifest to a client device.

An operating method of an electronic apparatus includes receiving multi-streams that include a first to N-th streams in a descending order of priority, obtaining multi-content corresponding to the multi-streams, and outputting the multi-content in a multi-view mode so that the multi-content corresponding to the multi-streams is respectively output to a plurality of partial screens. The receiving of the multi-streams includes determining a bit rate of each of the multi-streams according to priorities among the multi-streams and buffering each stream at the determined bit rate, and the multi-streams include a main stream having a higher priority and one or more sub-streams having a lower priority than the main stream.

A method to estimate an inter-channel time difference (ITD) in an encoder using a discontinuous transmission (DTX) is disclosed. The method includes receiving a time domain audio input including audio input signals and processing the audio input signals in frames to produce a mono mixdown signal and one or more stereo parameters. The method further includes encoding the mono mixdown signal on a frame-by-frame basis by: encoding of active content of the mono mixdown signal at a first bit rate until a pause period is detected; estimating ITD parameters during the encoding of active content; switching the encoding from the active encoding content to inactive encoding to encode background noise at a second bit rate during the pause period; and estimating ITD parameters during the pause period. The method further includes encoding the ITD estimated parameters and other stereo parameters periodically during the pause period.

The invention discloses a token stream guide variable code rate image compression method oriented to unification of perception and understanding. The method comprises the following steps of: 1, acquiring a two-dimensional image, and processing the two-dimensional image to acquire a one-dimensional token sequence; 2, processing the one-dimensional token sequence through a variable token mask to generate a binary compressed bit stream; 3, decoding the compressed bit stream to obtain an unmasked token sequence, and carrying out dynamic prediction on the unmasked token sequence to recover a complete token sequence; and 4, realizing human perception through the receiving end I after the complete token sequence is recovered, realizing machine perception through the receiving end II after the complete token sequence is recovered, and finally realizing image compression. The method has the capability that a single model supports the continuous variable bit rate, supports a large language model to directly perform semantic understanding based on the compressed code stream, and has the characteristics of low calculation complexity and low delay.

The present disclosure relates to a 5G or 6G communication system for supporting a data transmission rate higher than that of a 4G communication system such as LTE. The present disclosure relates to a method and a device therefor, the method comprising: determining a target bitrate and a neural network model on the basis of an E2E latency requirement, network state information, and computing state information; dividing the E2E latency requirement into a first latency requirement and a second latency requirement on the basis of the target bitrate, the neural network model, a first latency distribution, and a second latency distribution; and transmitting the first latency requirement and encoding frame information to a base station, wherein a terminal is allocated, from the base station, a resource block determined on the basis of the first latency requirement, the encoding frame information, and the network state information, and a GPU is allocated to the terminal on the basis of the second latency requirement and the computing state information.

The invention discloses a 360-degree video edge caching joint bit rate allocation method based on double time scales, and belongs to the technical field of 360-degree video streaming and edge caching. The method comprises the following steps that: in a long-time scale period, an edge server collects and analyzes historical request data of a user, calculates a comprehensive value priority for candidate video slices according to long, medium and short-term popularity of the slices, and further generates an optimal caching strategy which takes effect in a next period. In a short time scale period, according to the caching strategy, the edge server processes the real-time requests of the user and distributes bit rates to all concurrent requests fairly and efficiently. Through a self-adaptive optimization mechanism combining long-term planning and real-time response, the high cache hit rate of high-value slices can be ensured in different storage spaces, the data request quantity to the cloud is reduced, the pressure of a return link is relieved, and the comprehensive experience quality of a user is remarkably improved.

This invention discloses an adaptive frame abstraction interface management method and system for multiple encoding formats, relating to the field of data processing technology. The method includes: constructing a format configuration state diagram and a page transition time sequence; identifying the page transition time interval for corresponding format jumps; extracting structural difference parameters of the preceding and following formats and making them dimensionless to obtain fragmentation expectation parameters; making the actual time dimensionless and adding it to the expectation parameters to obtain reassembly resistance parameters; calculating the pre-allocated memory capacity based on the resistance parameters and the real-time bit rate; and performing physical isolation and decoding context preloading. This invention solves the problem of latency accumulation and screen stuttering caused by insufficient underlying memory reassembly and state machine switching during dynamic switching of multiple encoded video streams, achieving smooth transitions and low-latency decoding of video streams.

A method includes determining, by a user equipment (UE) operating at a first bitrate, that a congestion condition exists. The UE transmits a first message to a network apparatus, the first message including an indication that a bit rate is to be lowered, receives, from the network apparatus, a second message indicating a second bitrate, and adjusts the first bitrate to the second bitrate for operation.

A method for reducing artifacts in streaming video implemented by a computing device involves determining whether a video quality assessment score for a current segment of a channel of a video stream with a first resolution and a first bit rate meets a quality threshold, replacing the current segment with a replacement segment with a lower resolution than the first resolution, in response to the current segment failing to meet the quality threshold, and publishing the replacement segment at the first bit rate as part of the video stream.

Disclosed herein are system, apparatus, article of manufacture, method and / or computer program product embodiments, and / or combinations and sub-combinations thereof, for evaluating bitrate selection models. An example embodiment operates by aggregating data associated with a data streaming session comprising a plurality of segments. The embodiment then determines, for each segment of the plurality of segments, segment data comprising streaming parameters by analyzing the aggregated data. The embodiment then generates a simulation input based on the segment data for each segment. The embodiment then performs a first simulation, based at least on the segment data for each segment and a first bitrate selection model, to simulate streaming each segment. The embodiment then generates a first simulation result for each segment. The embodiment then aggregates each first simulation result to provide a first aggregate simulation result of the data streaming session.

Methods, systems, and devices for wireless communications are described. A wireless communications system may support immersive applications such as extended reality, mixed reality, interactive mapping, integration of artificial intelligence, among other services which may utilize high quality and low latency video streams. To support improved integration between network components providing applications, and to increase quality of experience (QoE) for users, the wireless communications system may implement radio access network (RAN)-assisted QoE-aware source bitrate selection. In some aspects, an application server may provide a RAN node with bitrate information for one or more users, including at least a first bitrate that maintains a threshold QoE for a threshold duration. The RAN node may determine, based on various factors such as network load, load balancing, obtainable QoE across users, among other factors, a selected bitrate for the one or more users and may provide the selected bitrate to the application server.