186 results about "Adaptive coding" patented technology

The present invention relates to transmission and reception of digital broadcast in a digital broadcast network supporting a configuration of multiple physical layer pipes (PLPs). In particular, signalling parameters relating to a complete PLP are transmitted within layer 1 signalling related to the PLP. The baseband frames mapped on the pipe are configured according to this layer 1 signalling in the same way at the transmitter as they are demapped on the receiver side. The baseband frames are transmitted and received without including these parameters, in particular, at least one of parameters indicating (i) an input stream format, (ii) a single or a multiple input stream, (iii) constant or adaptive coding and modulation, (iv) presence of input stream synchronization, (v) presence of null packet deletion, or (vi) input stream identifier.

The present invention provides a unique feedback system in a wireless communication system wherein the transmission channel is always overloaded with data, which is configured and modulated at a select order of modulation. Based on the channel conditions, only a portion of the data can be successfully received. As such, the receiver will attempt to receive the data using demodulation techniques corresponding to the different levels of modulation used to encode the data transmitted. The receiver will determine the data that is recovered, or the highest order of modulation at which the data is recovered, and provide feedback to the transmitter. The transmitter will then retransmit the data that was not properly received. By overloading the channel at all times, the maximum amount of data capable of being transmitted over the channel is optimized without requiring or heavily relying on the receiver feeding back channel condition information to the transmitter.

The invention discloses a frame field self adaptive coding method based on image piece cutting structure. It includes following steps: the image is cut into pieces; carries on frame coding and field coding to each piece; selects one with small cost as the coding mode; adds a image piece head signing the image piece for each image piece; adds a macro index for each image piece; adds a image head for each image, it forms the coding data of image together with the coding data of several image piece and macro index. The invention overcomes the deficiency that the flexibility of the whole image frame field coding is not enough.

Code compression techniques and decompression architectures for embedded systems are disclosed, providing good compression ratio while improving decompression time for VLIW instructions and reducing bus power consumption. The invention includes two fixed-to-variable (F2V) length code compression schemes based on a reduced arithmetic code compression algorithm combining arithmetic coding with probability models; a static probability model using static coding and semi-adaptive coding using a Markov model. Multi-bit decompression methods for the F2V techniques are presented, together with a parallel decompression scheme that tags and divides a compressed block into smaller sub-blocks. The Markov model provides better compression ratio, but the static model has a less complicated decompression unit design. The invention also includes two variable-to-fixed (V2F) length coding algorithms, one based on Tunstall coding and another on arithmetic coding. The V2F algorithms are also combined with a static model and a Markov model.

When transmitting data from a transmission unit to a reception unit over a channel in a wireless communication system, it has been observed that there are inherent losses in throughput, especially for a channel experiencing high channel quality variations over its frequency range. To improve the throughput over such a channel, it is proposed to: estimate the quality of each subchannel of a channel, classify the subchannels into a number of quality groups based on the estimated quality, and select a code rate per quality group. This selected code rate per quality group is then used when a data word to be transmitted is coded. Simulations have shown that by grouping the subchannels into a limited number of groups, for example one to four groups depending on the channel quality variation, a high throughput could be achieved with a minimum of signaling cost in the form of extra overhead.

Methods and devices for a media processing is provided. In one respect, the methods can provide initiating a bandwidth throttle or a frame rate throttle when resources of a network exceed resources of client device. The methods of the present disclosure may also provide techniques for handling lost packets during transmission using wavelet coefficients.

A method and apparatus for adaptive inter prediction mode coding are disclosed. In the current HEVC, a fixed set of variable length codes is used for the underlying video data, which may not optimally match the statistics of underlying video data. Consequently, the compression efficiency associated with the fixed set of variable length codes will be compromised. Accordingly, an adaptive coding scheme for inter prediction modes is disclosed. The variable length codes used for each inter prediction mode in each coding unit depth is adaptively determined by its respective statistics. The statistics can be measured as the frequency of occurrence of each mode. In one embodiment according to the present invention, counters are used to collect the statistics. According to one embodiment of the present invention, the statistics of inter prediction modes are collected from the previous slice and the set of variable length codes is determined for the subsequent slice (immediately following the previous slice) accordingly. According to another embodiment of the present invention, the statistics of inter prediction modes are updated for each coding unit and the variable length code for each mode is adjusted according to the statistics change during the coding process. According to another embodiment of the present invention, the variable length code for each mode is reset in the beginning of each slice. The reset code word table is either a predefined code word table for whole sequence or a code word table determined by the previous slice.

In a communication where a retransmission is attained based on a Hybrid-ARQ (Automatic repeat ReQuest) system, retransmission having no gain is reduced, and improvement of the transmission efficiency is intended. A mode determining unit 111 of a base station carries out calculation to obtain a received signal quality difference information indicative of a difference between a received signal quality at the current stage and that of a past stage at a terminal, and supplies to a control unit 112. The control unit 112 controls a power setting unit 113 based on the received signal quality difference information supplied from the mode determining unit 111, and thus a transmission power of a signal transmitted from an adaptive coding and modulation unit 13 can be controlled. The present invention can be applied to a base station of a portable telephone communication network, for example.

A method and system of lossless compression of integer data using a novel backward-adaptive technique. The adaptive Run-Length and Golomb / Rice (RLGR) encoder and decoder (codec) and method switches between a Golomb / Rice (G / R) encoder mode only and using the G / R encoder combined with a Run-Length encoder. The backward-adaptive technique includes novel adaptation rules that adjust the encoder parameters after each encoded symbol. An encoder mode parameter and a G / R parameter are adapted. The encoding mode parameter controls whether the adaptive RLGR encoder and method uses Run-Length encoding and, if so, it is used. The G / R parameter is used in both modes to encode every input value (in the G / R only mode) or to encode the number or value after an incomplete run of zeros (in the RLGR mode). The adaptive RLGR codec and method also includes a decoder that can be precisely implemented based on the inverse of the encoder rules.

A process is described to build physical layer frames with a modcode adapted to the signal quality of a destination terminal. Data packets assigned to the same modcode may be sent in the same frame, although packets associated with higher modcodes may be used to complete a frame before switching to the applicable higher modcode for construction of subsequent frames. After an interval, the order of progression is restarted with an out of order packet above a threshold age. Flow control filtering mechanisms and a variable reliability margin may be used to adapt dynamically to the current data traffic conditions.

The embodiment of the invention discloses a video processing method and device. One specific embodiment of the method comprises steps: a to-be-processed video is acquired, and the video is divided toat least one video clip according to a scene; as for the video clip in the at least one video clip, the features of the video clip are extracted; the features and a target video quality are inputted to a pre-trained coding parameter prediction model to obtain the coding parameter of the video clip, wherein the coding parameter prediction model is used for characterizing the corresponding relationship between the features of the video clip and the coding parameter under the target video quality; and according to the coding parameter corresponding to the video clip in the at least one video clip, the video is subjected to coding processing, and a video after coding is generated. The implementation mode realizes adaptive coding based on the scene content, the compression rate can be improved,and the video quality is enhanced.

A video quality adaptive coding artifact reduction system has a video quality analyzer, an artifact reducer, and a filter strength controller. The video quality analyzer employs input video quality analysis to control artifact reduction. The video quality analyzer accesses the video quality of the decoded video sequence to estimate the input video quality. The filter strength controller globally controls the filter strength of the artifact reducer based on the video quality estimate by the video quality analyzer. For low quality input video, the filter strength controller increases the artifact reduction filter strength to more efficiently reduce the artifact. For high quality input video, the filter strength controller decreases the artifact reduction filter strength to avoid blurring image detail.

A selection unit selects ‘0’ for the first through the (N−1)th frames, and selects a predicted image for the subsequent frames. Thus, an intra-frame coding process is performed on the first through the (N−1)th frames, and an adaptive coding process is performed on the subsequent frames. An information amount reduction unit reduces the amount of information about data in the first through the (N−1)th frames. An information amount adjustment unit stepwise increases the amount of information reduced by the information amount reduction unit.

A system for, and method of, entropy coding. In one embodiment, the system includes: (1) a memory configured to contained initialized accumulated statistics coding variable values and (2) a processor configured to employ the initialized accumulated statistics coding variable values to context-adaptive encode multiple symbols until an accumulated statistics update condition occurs and updating and store the accumulated statistics coding variable values in the memory only upon occurrence of the accumulated statistics update condition.

A Second Generation Digital Video Broadcasting via Satellite (DVB-S2) system is provided. More particularly, a method and apparatus for maintaining synchronization of a signal by changing an Adaptive Coding and Modulation (ACM) method that is used for a conventional DVB-S2 system are provided. In the apparatus and method, an FEC frame of a variable length is formed by turbo encoding rather than Bose-Chaudhuri-Hocquenghem (BCH) and Low Density Parity Check (LDPC) encoding, and a Physical Layer (PL) frame of a specific length is formed regardless of a modulation method, so that a satellite terminal receives a signal transmitted at a specific length regardless of a modulation method or a coding rate to easily maintain synchronization without interruption and efficiently transmit the signal.

The invention discloses an adaptive coding method based on VR terminal feedback and belongs to the technical field of multimedia transmission. The core concept of the method comprises the steps that a transmission mechanism is improved; a panorama video is segmented into a plurality of visual angle videos; each piece of visual angle video information is independently coded and transmitted; the visual angle video information demanded by a user is transmitted in real time according to a terminal visual angle tracking technology; other visual angle information is transmitted in a low bit rate; various visual angle videos are spliced into the panorama video at the terminal; the user is enabled to obtain suitable visual angle information through adoption of the terminal scoring feedback mechanism; a scoring instruction of the user is transmitted to a server; and the server maps different bit rates to the terminal according to the user scores. The invention has the advantages that under the existing transmission channel capacity, the panorama video sill can be transmitted, so a VR technology can be realized on various platforms, the method is applicable to wide groups, a channel itself does not need to be excessively improved, and the user is enabled to select the suitable subjective watching experience through adoption of the terminal scoring feedback mechanism.

A method and system of lossless adaptive Golomb / Rice (G / R) encoding of integer data using a novel backward-adaptive technique having novel adaptation rules. The adaptive G / R encoder and decoder (codec) and method uses adaptation rules that adjust the G / R parameter after each codeword is generated. These adaptation rules include defining an adaptation value and adjusting the G / R parameter based on the adaptation value. If the adaptation value equals zero, then the G / R parameter is decreased by an integer constant. If the adaptation value equals one, then the G / R parameter is left unchanged. If the adaptation value is greater than one, then the G / R parameter is increased by the adaptation value. In addition, the adaptive G / R encoder and method include fractional adaptation, which defines a scaled G / R parameter in terms of the G / R parameter and updates and adapts the scaled G / R parameter to slow down the rate of adaptation.

A method for determining coding unit (CU) partitioning of a largest coding unit (LCU) of a picture is provided that includes computing a first statistical measure and a second statistical measure for the LCU, selecting the LCU as the CU partitioning when the first statistical measure does not exceed a first threshold and the second statistical measure does not exceed a second threshold, and selecting CUs in one or more lower layers of a CU hierarchy of the LCU to form the CU partitioning when the first statistical measure exceeds the first threshold and / or the second statistical measure exceeds the second threshold.

Bits are stored by attempting to set parameter value(s) of (a) cell(s) to represent some of the bits. In accordance with the attempt, an adaptive mapping of the bits to value ranges is provided and the value(s) is / are adjusted accordingly as needed. Or, to store (a) bit(s) in (a) cell(s), a default mapping of the bit(s) to a predetermined set of value ranges is provided and an attempt is made to set the cell value(s) accordingly. If, for one of the cells, the attempt sets the value such that the desired range is inaccessible, an adaptive mapping is provided such that the desired range is accessible. Or, to store (a) bit(s) in (a) cell(s), several mappings of the bit(s) to a predetermined set of ranges is provided. Responsive to an attempt to set the cell value(s) according to one of the mappings, the mapping to actually use is selected.

An adaptive coding and modulation (ACM) apparatus and method for a forward link in satellite communication is provided. The ACM apparatus may include a receiving unit to receive a signal-to-noise ratio (SNR) of a received signal, a determination unit to determine whether the SNR of the received signal is less than or equal to a threshold value, and a processing unit to execute a channel prediction algorithm when the SNR of the received signal is determined to be less than or equal to the threshold value.

Embodiments provide methods, systems, and apparatuses for adaptive coding, sampling, and modulating over a satellite communication channel. In some embodiments, a method of adaptive coding, sampling, and modulating over a satellite communication channel may include providing multiple data frames. At least one modcode partition may be provided for each data frame. The modcode partitions may represent modulation, coding, spreading, and / or frame size information for respective data frames. Each data frame and the respective modcode partition may be combined to form a modcode data unit. Multiple superframes may be formed. Each superframe may include a first known sequence and multiple subframes. Each subframe for a respective superframe may include a portion of a respective modcode data unit. Multiple superframes may be transmitted across a wireless channel.

The utility model discloses a regulation method used for modulating the codes and a adaptive coding modulator approach, which comprise the steps as follows: according to the handing capacity and the signal-to-noise ratio curve, the initial value of the encoding conditions can be ascertained; as per signal-to-noise ratio, the degree of encoding can be adjusted; the adjusted encoding degree can be used to adjust the initial value of the encoding conditions, so as to obtain the encoding status. Recording the channel enveloping variation and the variance ratio thereof as the signal-to-noise ratio leaps over the limits of the setting time; also, the channel enveloping variation and the variance ratio thereof can be handled via the fuzzy algorithm, so as to obtain the second trim. The signal-to-noise ratio-based trim can be used to adjust the encoding status to improve the control precision of the transmission mode. The fuzzy algorithm can be adopted to track the channel characteristics of the existing occasion quickly and accurately and forecast the channel variation of the next occasion, so as to further improve the control precision of the transmission mode. Thus, more handing capacity can be obtained to enhance the transmission rate of the system.

A method of encoding a digital video sequence is provided that includes computing a motion vector (MV) histogram and a motion threshold for a first frame of the digital video sequence, and using the MV histogram and the motion threshold for one selected from a group consisting of adaptive fcode determination and adaptive coding structure determination.

The method containing a pre-analysis phase with pel-recursive type motion estimation and an actual coding phase with block matching motion estimation for picture blocks constituted by one or more sub-blocks, wherein the coding phase also comprises a motion vector selection stage for the picture sub-blocks based on the motion vectors calculated by matching for the block containing the sub-block and based on the motion vectors calculated by the pel-recursive type estimation for the pixels or the pixel groups belonging to the sub-block, and a calculation stage of the best partition of a picture block into sub-blocks based on the correlation calculations taking into account the motion vectors selected from the sub-blocks constituting a block.

The invention discloses an uplink adaptive modulation and coding method and a base station implementing the method, wherein the method includes: the base station locally configures the corresponding relationship between uplink Signal to Interference-plus-Noise Ratio (SINR) and uplink Modulation and Coding Scheme (MCS), measures uplink SINR to obtain the uplink SINR measurement value, performs filtering and smoothing processing on the SINR measurement value and history SINR filtering values to obtain uplink SINR smoothing filtered value, and determines the MCS adopted by the uplink based on the uplink SINR smoothing filtered value and the corresponding relationship. Through the invention, the dependence on the corresponding relationship between the uplink SINR and the MCS is reduced, and the current channel conditions can be reflected more in real, responses can be made quickly and stability and reliability of the system can be effectively guaranteed.