2192 results about "Code rate" patented technology

The present disclosure provides a variable-bit-rate image compression method and system, an apparatus, a terminal, and a storage medium. The variable-bit-rate image compression method includes: obtaining an initial feature map from a to-be-encoded image; quantizing the initial feature map by a dead-zone quantizer; performing entropy encoding on the quantized feature map and hyper-prior information to obtain a compressed bit-stream; performing entropy decoding on the compressed bit-stream, and recovering quantized hyper-prior information and the quantized feature map; performing inverse quantization on the quantized feature map to obtain a reconstructed feature map; obtaining a reconstructed image from the reconstructed feature map; and adjusting quantization and inverse quantization parameters according to a target bit-rate or target distortion. The present disclosure provides a precise bit-rate control solution, makes the bit-rate of the compressed bit-stream better adapt to the dynamic change of a network bandwidth, and has an extremely high actual application value.

The invention provides a monitoring video transmission method and system based on wireless communication, and relates to the technical field of monitoring video transmission, and the method comprises the steps: obtaining and preprocessing network state data, the network state data comprising bandwidth, time delay and packet loss rate; establishing a network state prediction model based on the LSTM neural network, and predicting a future network state according to the network state prediction model; based on the future network state, a multi-objective optimization algorithm is adopted to perform transmission path selection; and for the selected transmission path, video coding parameters are adjusted, the coding parameters comprise a code rate, a resolution ratio and a frame rate, and a self-adaptive transmission protocol is adopted to dynamically adjust the transmission rate according to a real-time network state. According to the method, different paths can be dynamically selected, network congestion or degradation is avoided, and the video transmission quality and rate are adaptively and flexibly adjusted according to the current network state, so that stable transmission of the video quality under different network conditions is ensured.

The invention relates to the technical field of audio coding and decoding, and discloses an audio coding and decoding method and device based on a neural network, equipment and a storage medium. The method comprises the following steps of: firstly, gradually quantizing residual information of an audio signal by introducing an improved residual vector quantization method, and more finely retaining audio characteristics in a compression process; then, a loss function fusing reconstruction loss, perception loss, residual vector quantization loss and commitment loss is adopted, so that the reconstruction quality of the audio signal under different bit rates is remarkably improved, and efficient audio compression can be realized while high tone quality is ensured; and finally, an entropy coding module is introduced after quantization, so that the code rate of coding is further reduced, and the bandwidth requirement of audio real-time transmission is remarkably reduced.

An apparatus may include a transmitter and one or more processors. The one or more processors may identify, based on a first parity check matrix of a first quasi-cyclic-low-density parity-check (QC-LDPC) code according to a code rate of 3 / 4, a second parity check matrix corresponding to a first exponent matrix comprising 576 values for a second QC-LDPC code. The second QC-LDPC code may have a code block size that is twice a code block size of the first QC-LDPC code. The one or more processors may encode data using the second parity check matrix. The transmitter may be configured to transmit the encoded data.

The present disclosure is related to multi-access traffic management in multi-access computing environments. A reliability enhancement engine (REE) operated by a multi-access multipath traffic manager identifies traffic flows that have high reliability requirements (“reliability flows”) and performs admission control for the reliability flows. The REE determines a coding rate (e.g., a redundancy factor) and traffic distribution strategies for the identified reliability flows. The REE applies cross-access network coding (NC) on the reliability flows, and handles multi-access acknowledgements and traffic pacing, which may involve multiplexing traffic of the reliability flows with non-high-reliability traffic. Other embodiments may be described and / or claimed.

The invention belongs to the technical field of video communication processing, and particularly relates to an image quality improvement method for a video call of a fusion scheduling system, which comprises the following steps: performing quality evaluation on collected video frames, determining image quality information according to a quality evaluation value, a preset threshold value and statistical historical video frame quality, and sending the image quality information to a server; and then the code rate and the resolution used by the current video coding are obtained, and the video coding parameters are dynamically adjusted based on the image quality information, the code rate and the resolution, so that the self-adaptive video coding parameter adjustment based on the video content is realized, and the optimal video quality and transmission efficiency are achieved. According to the invention, the video data in the video call of the converged communication scheduling system is processed, and the problems that the image quality is reduced and the jamming is obvious are eliminated, so that high-definition and smooth video image quality is provided, the user experience is improved, and the use requirements are met.

The invention discloses a compression transmission method and device for an electric power safety supervision video and electronic equipment. The method comprises the steps that an electric power safety monitoring video needing to be transmitted is acquired, the electric power safety monitoring video is used for representing the electric power safety monitoring video, the electric power safety monitoring video at least comprises an image frame set and an audio, and the image frame set at least comprises a current image frame; identifying a dynamic area and a static area in the current image frame; code rate distribution is carried out on the current image frame based on the dynamic area and the static area, a code rate distribution result is obtained, and the code rate distributed to the dynamic area in the code rate distribution result is larger than the code rate distributed to the static area; compressing the image frame set based on the code rate distribution result to obtain a compressed image frame set; and constructing a compressed safety supervision video based on the compressed image frame set and the audio, and transmitting the compressed safety supervision video. According to the invention, the technical problem that the compression transmission efficiency of the electric power safety supervision video is low in the prior art is solved.

An apparatus may include a transmitter and one or more processors. The one or more processors may identify, based on a first parity check matrix of a first quasi-cyclic-low-density parity-check (QC-LDPC) code according to a code rate of 5 / 6, a second parity check matrix corresponding to a first exponent matrix comprising 384 values for a second QC-LDPC code. The second QC-LDPC code may have a code block size that is twice a code block size of the first QC-LDPC code. The one or more processors may encode data using the second parity check matrix. The transmitter may be configured to transmit the encoded data.

The present technology relates to a transmission device, a transmission method, a reception device, and a reception method capable of ensuring good communication quality in data transmission using an LDPC code. The LDPC coding is performed on the basis of the parity check matrix of the LDPC code with the code length N of 1224 or 1152 bits and the coding rate r of 144 / 1224 or 144 / 1152. The parity check matrix includes an A matrix of M1 rows and K columns represented by a predetermined value M1 and an information length K=N×r of the LDPC code, a B matrix of a staircase structure of M1 rows and M1 columns, a Z matrix which is a zero matrix of M1 rows and N−K−M1 columns, a C matrix of N−K−M1 rows and K+M1 columns, and a D matrix which is an identity matrix of N−K−M1 rows and N−K−M1 columns. The A matrix and the C matrix are represented by a parity check matrix initial value table, and the parity check matrix initial value table is a table representing the positions of the elements of 1 of the A matrix and the C matrix for every 36 columns, and is a predetermined table. The present technology can be applied to, for example, data transmission using an LDPC code.

The invention discloses a video compression intelligent preprocessing method, system and device based on frequency domain perception optimization and a medium. The method comprises the following steps: constructing a preprocessing model; preprocessing the input coding frame and the reference frame; enabling the preprocessed coding frame and reference frame to pass through a virtual encoder to obtain a frequency domain coefficient of a residual frame, and calculating a code rate of the coding frame; enabling the frequency domain coefficient of the residual frame to pass through a virtual decoder to obtain a decoded frame; training a preprocessing model; testing the effect; the system, the equipment and the medium are used for implementing the method. According to the method, a frequency domain sensing module is introduced into a virtual encoder in a training stage, modeling is carried out on a frequency domain information context, and statistical redundancy between transformation units is eliminated; an optical flow collaboration module is introduced into a virtual encoder, local distortion of a prediction frame is repaired in a feature domain by using high-level semantic information, and the difference between training and actual deployment is effectively reduced; through the optimization, the preprocessing model can better learn the rate distortion characteristic of the standard encoder, and the quality of the preprocessed video is remarkably improved.

The invention discloses an extreme compressed video coding method based on an intelligent reference frame, comprising the following steps: S1, acquiring an original video stream, dividing the original video stream into frame groups according to a time sequence, each group comprising a current frame and a candidate reference frame; s2, carrying out region positioning on the candidate reference frame, extracting motion, edge and background regions, and generating a mapping graph; s3, calculating a quality score based on the mapping graph according to a motion vector, gradient change and a region overlapping rate; s4, selecting three frames with highest scores to form a reference set, and establishing an index structure; s5, predicting the current frame by using the reference set to generate a predicted frame and a residual error; s6, multi-path coding cost is calculated, and a path with the minimum cost is selected; and S7, entropy coding is carried out on the reference index, the motion vector, the residual error and the control parameter, a code stream is output, and calling information is recorded. According to the method, the compression ratio and prediction precision of video coding are improved, the image quality is maintained while the code rate is reduced, and the method is suitable for efficient transmission and storage of high-resolution videos.

An apparatus may include a transmitter and one or more processors. The one or more processors may identify, based on a first parity check matrix of a first quasi-cyclic-low-density parity-check (QC-LDPC) code according to a code rate of 2 / 3, a second parity check matrix corresponding to a first exponent matrix comprising 768 values for a second QC-LDPC code. The second QC-LDPC code may have a code block size that is twice a code block size of the first QC-LDPC code. The one or more processors may encode data using the second parity check matrix. The transmitter may be configured to transmit the encoded data.

The invention discloses a low signal-to-noise ratio direct spread signal detection method based on noise cancellation, and belongs to the field of communication spectrum sensing. The self-adaptive noise cancellation method based on the minimum mean square error is adopted, non-stationary noise interference can be tracked and eliminated in real time, filtering parameters are automatically optimized in an unknown channel environment, and a signal detection system can adapt to different background noise conditions. And stable signal detection is realized in a low signal-to-noise ratio environment by utilizing a cyclic spectrum analysis method and extracting the cyclic stability characteristic of the direct spread signal. The existence of the signal is judged through the characteristic spectrum peak on the non-zero cyclic frequency, and the carrier frequency and the pseudo code rate are further estimated, so that more accurate signal identification and parameter extraction are realized, the influence of noise uncertainty on the detection performance is avoided, and the detection robustness and reliability are improved. Welch smoothing processing and short-time Fourier transform are combined, the variance of spectrum estimation is reduced when the cyclic spectrum is calculated, and the detection robustness is improved.

The invention discloses an anti-turbulence free space optical communication method based on receiving and transmitting joint optimization, which balances the code length and the code rate through a Polar coding mode of an atmospheric turbulence channel, and ensures the coding and decoding overhead; meanwhile, a Transform model based on convolution enhancement is designed at a receiving end, learning of local features is enhanced through convolution, the relation between phase screen pixels is established, and a conjugate phase screen is given by the model for turbulence compensation. In order to solve the problem that in a free space optical transmission system, the atmospheric turbulence effect interferes with an OAM light beam, the influence of atmospheric turbulence interference on free space optical communication is relieved in a mode of joint optimization of the receiving end and the transmitting end. Error codes caused by atmospheric turbulence are corrected through Polar coding of the transmitting end, and the coupling efficiency of receiving light spots is improved after phase compensation of the receiving end, so that the mode purity obtained by corrected light spot images is improved, and the error rate of the system is reduced.

The invention discloses a 5.5 G network encryption communication method based on quantum key distribution, and relates to the technical field of quantum encryption communication, a channel characteristic data set, an equipment parameter data set and an environment parameter data set are collected in advance, a channel prediction model is constructed based on the channel characteristic data set and the environment parameter data set, and the channel prediction model is constructed based on the equipment parameter data set and the environment parameter data set. And constructing an equipment optimization model based on the equipment parameter data set and the prediction result of the channel prediction model, constructing an error rate optimization model for outputting a key management strategy based on the prediction result of the channel prediction model and the optimization result of the equipment optimization model, and sending the optimized key management strategy. According to the method, a key management strategy is established, quantum key distribution is executed according to the key management strategy, changes of channel parameters and equipment parameters are monitored in real time through a 5.5 G network, quantum signal generation and transmission strategies are adjusted by using a genetic algorithm based on the changes of the channel parameters and the equipment parameters monitored in real time, and the robustness and flexibility of 5.5 G network communication are enhanced.

The invention belongs to the field of image / video compression, and discloses an intelligent image compression encoder based on a conditional reversible neural network, which comprises an image enhancement module, a reversible multi-frequency fusion network RMFFN and an entropy model, and is characterized in that the image enhancement module comprises a multi-expansion channel refiner module MDCR and an expansion residual attention module RDAM; the image enhancement module optimizes input features and constructs a conditional reversible neural network by using a flow model, namely, multi-stage nonlinear mapping is realized by stacking four reversible multi-frequency fusion networks; the super-prior codec of the entropy model extracts hidden variable features by stacking multi-scale residual attention blocks, and introduces a discrete Gaussian mixture likelihood model in an entropy coding stage. According to the method, high-fidelity reconstruction is still kept at a low bit rate, and the technical problems of bit rate-distortion balance, high-frequency detail retention and calculation efficiency of a traditional method are solved.

An apparatus may include a transmitter and one or more processors. The one or more processors may identify, based on a first parity check matrix of a first quasi-cyclic-low-density parity-check (QC-LDPC) code according to a code rate of 1 / 2, a second parity check matrix corresponding to a first exponent matrix comprising 1152 values for a second QC-LDPC code. The second QC-LDPC code may have a code block size that is twice a code block size of the first QC-LDPC code. The one or more processors may encode data using the second parity check matrix. The transmitter may be configured to transmit the encoded data.

A communication method and device are applied to the technical field of communication, can be applied to a wireless local area network system of a next generation Wi-Fi protocol of IEEE 802.11 ax, such as 802.11 be, Wi-Fi 7 or EHT, and 802.11 series protocols such as 802.11 bn, Wi-Fi 8, UHR, Wi-Fi AI, millimeter wave (MMW) and the like, and can also be applied to a WPAN system supporting UWB, a sensing system and the like. A first communication device generates a PPDU and transmits the PPDU. The second communication device receives the PPDU and analyzes the PPDU. The PPDU comprises first indication information, the first indication information can be used for indicating a first modulation mode set and a coding rate, and the first modulation mode set is one of N modulation mode sets divided based on W modulation modes. The Q modulation modes in the first modulation mode set are used for modulating the PPDU.

The invention discloses a packet loss error correction method and system based on a sliding window, and belongs to the field of packet loss error correction, and the method comprises the steps: setting an original data packet number N, a redundant data packet number M, FEC coding time and FEC decoding time, and determining a preliminary window size according to the FEC coding and decoding time, a packet loss rate, a code rate and jitter rate information; carrying out redundancy coding on the data packets by adopting forward error correction codes, and sequentially sending N + M data packets by a sending end based on a sliding window; the receiving end maintains a receiving window, stores the received data packet, and performs error correction by using the redundant packet; detecting the lost data packet by adopting an acknowledgement mechanism, and if the ACK or the NACK is not received in the Ttimeout, retransmitting the lost data packet by the transmitting end; and dynamically adjusting the window size according to the packet loss rate. According to the method, the window size and the FEC redundancy ratio are dynamically adjusted according to the network packet loss rate, adaptive adjustment is achieved in different network environments, bandwidth waste is reduced, higher error correction capacity is provided in a high packet loss environment, and data transmission efficiency is improved.

Methods and apparatuses for transmitting and receiving information of different priorities. A method of operating a UE includes receiving a first PDCCH providing a first DCI format. The first DCI format schedules a reception of a PDSCH, includes an MCS field, and includes a priority indicator field. The method further includes determining an MCS table, from a first predetermined MCS table or a second predetermined MCS table, based on a value of the priority indicator field in the first DCI format; determining a modulation order and a code rate from the MCS table based on a value of the MCS field; receiving a TB in the PDSCH according to the modulation order and the code rate; and determining a priority of HARQ-ACK information in response to the TB reception based on the value of the priority indicator field in the first DCI format.

The invention provides a video monitoring platform adaptive play support optimization method, which comprises the following steps that: a server acquires equipment information and user habits of a terminal, and correspondingly configures a plurality of types of video stream preprocessing strategies; and the server selects a video stream preprocessing strategy based on the prediction operation, and pulls the preprocessed video stream to the corresponding terminal in advance. Wherein when the video picture proportion of the stream pushed by the server is not matched with the picture proportion of the terminal player, the server carries out cutting and scaling on the pushed video stream based on a black edge removing algorithm. According to the method, efficient self-adaptive stream pushing is realized by predicting user habits and combining the preset strategy, on one hand, adaptive coding video streams and code rates can be pushed according to requirements, so that bandwidth occupation is reduced, playing delay is reduced, and terminal types are expanded; and on the other hand, the video stream matched with the picture proportion is pushed by using the black edge removing algorithm in a feedback mode on the server side, so that the effective information of the picture is reserved to the maximum extent, and the playing pressure of the terminal can be reduced.

The invention provides a quantum key generation method and device, equipment and a medium, and relates to the technical field of quantum communication. The method comprises the following steps: negotiating with a first communication party through a classical channel, and determining a coding strategy; receiving a photon sequence sent by the first communication party, randomly selecting a base vector to measure photons of the order recorded in the coding strategy to obtain a first measurement result, and storing the photons which are not measured; after the first communication party publishes the measurement base vector of the photon sequence and the bit string corresponding to the photon of the order recorded in the coding strategy, the bit error rate is calculated based on the bit string and the first measurement result; under the condition that the bit error rate is smaller than or equal to a preset threshold value, measuring unmeasured photons through a base vector published by the first communication party to obtain a second measurement result; a quantum key is generated based on the second measurement. According to the embodiment of the invention, the number of keys generated in unit time can be increased, and the code rate of the quantum key is improved.

The embodiment of the invention provides a video coding method and device, and the method comprises the steps: determining a target coding parameter of a to-be-coded video according to a total code rate corresponding to a plurality of to-be-coded videos and the video features of the to-be-coded videos after the video features of the to-be-coded videos are extracted from the to-be-coded videos; and then, coding the video to be coded by adopting the target coding parameter to obtain a coded video. In the process, the target coding parameter corresponding to the to-be-coded video can be determined through the total code rate corresponding to the plurality of to-be-coded videos and the video feature of the to-be-coded video, and the coding feature of the preprocessed video does not need to be acquired after the to-be-coded video is pre-coded, so that the coding efficiency of the to-be-coded video is improved. The efficiency of determining the target coding parameter of the to-be-coded video is improved, and the video coding quality is ensured.

The present invention discloses a data transmission method and a data transmission system for remote takeover of unmanned mining trucks, belonging to the technical field of data transmission. When the data transmission method is executed by the sending end, it includes: obtaining the data to be transmitted and its data type; in response to the data type of the data to be transmitted being video data, encoding and encapsulating the video data in sequence to obtain the original data packet and FEC redundant packet; sending the original data packet and FEC redundant packet to the receiving end; based on the retransmission request sent by the receiving end, sending the original data packet specified by the retransmission request to the receiving end; wherein, the encoding code rate and the transmission code rate sent to the receiving end are obtained by adjusting the code rate by the GCC congestion control algorithm according to the prediction result output by the pre-trained network prediction model, the actual delay sent by the receiving end, and the bandwidth detection result. The present invention can ensure the real-time performance of data transmission and improve the security and controllability of remote takeover.

The invention provides a user experience optimization method and system for an audio and video transmission system, and the method comprises the steps: firstly obtaining a transmission quality data set in the audio and video transmission process of a target terminal, the transmission quality data set comprises a plurality of transmission session sequences, and each transmission session sequence is composed of network fluctuation, coding and decoding parameters, user feedback and other features; performing transmission feature extraction on the data set to obtain audio and video synchronization, dynamic code rate and decoding delay features, calling a pre-trained reinforcement learning optimization model, performing dynamic aggregation processing on the features to generate a transmission quality evaluation result, and finally determining a dynamic transmission optimization strategy based on the transmission quality evaluation result. And the real-time transmission parameter adjustment operation is triggered, so that the user experience of the audio and video transmission system is optimized.

The embodiment of the invention provides a coding and decoding method and an end-cloud cooperation system. The coding and decoding method is realized based on the end-cloud collaborative system, and the total code rate of the end-cloud can be reduced under the condition of ensuring the end-side rendering picture. The encoding method comprises the steps that rendering processing is carried out on a three-dimensional scene to obtain a rendered image, a rendering intermediate result is generated in the rendering processing process, and the rendering intermediate result comprises a first rendering intermediate result. And performing target operation on the reconstructed image based on the first rendering intermediate result to generate a virtual reference block. And predicting the rendered image based on the virtual reference block to obtain a predicted image. And encoding a residual image between the predicted image and the rendered image to obtain a code stream. Wherein the target operation comprises at least one of reprojection, validity check, weighted averaging, hole filling or intra-frame noise reduction.

The invention discloses a layered voice coding and decoding method based on MDCT coefficient residual vector quantization, which comprises the following steps: preprocessing voice data for training, and training to generate a multi-level quantization codebook of MDCT; the method comprises the following steps of: enhancing a high-frequency component of voice to be coded through a pre-emphasis filter, windowing a pre-emphasis voice signal, generating an MDCT coefficient based on an FFT (Fast Fourier Transform) algorithm, determining a bit allocation mode according to a code rate to generate a coding control bit, compressing a codebook index and the coding control bit through Huffman coding, and transmitting a code stream to a channel; the method comprises the steps of reading a code stream in a channel, obtaining a coding control bit and an MDCT codebook index through Huffman decoding, determining a voice sampling rate and a bit distribution mode through the coding control bit, obtaining an MDCT vector based on MDCT codebook index retrieval, and restoring an original voice signal through de-emphasis filtering. Through the design of MDCT layered coding, the coder can dynamically adjust coding bit distribution according to the code rate condition, so that the method adapts to the actual communication environment and deals with the sudden network condition.

In some implementations, an apparatus may include a transmitter and one or more processors. The one or more processors may be configured to identify, according to a code rate of 1 / 2 and a code block size of 3888 bits, a first binary parity check matrix for a quasi-cyclic-low-density parity-check (QC-LDPC) code. The first binary parity check matrix may correspond to a first exponent matrix having 288 values. The one or more processors may be configured to encode data using the first binary parity check matrix. The one or more processors may be configured to transmit the encoded data.

This disclosure describes systems, methods, and devices for probabilistic constellation shaping in wireless transmissions may include a device configured to generate, using a first quadrature amplitude modulation (QAM) order shaping encoder associated with a first code rate, shaped amplitude bits; generate, using a forward error correcting (FEC) encoder and a second code rate smaller than the first code rate, parity bits for the shaped amplitude bits; cause to transmit, using a channel, a first portion of the parity bits as sign bits for the shaped amplitude bits; and cause to transmit, using the channel, a second portion of the parity bits.

The invention discloses an end-to-end video coding method based on a scale-spatial stream and hierarchical hyper-prior, and the method employs a hierarchical hyper-prior coding and decoding network to introduce hyper-prior information to assist the generation of entropy model parameters, thereby enabling an entropy model to obtain more accurate estimation probability distribution, and improving the compression efficiency. And meanwhile, the motion information of the video is better estimated in combination with a scale-spatial stream, and the video coding performance is improved by estimating a more accurate entropy model and motion information. Experimental results show that when SSIM and MS-SSIM are respectively adopted as evaluation standards, compared with an HEVC (High Efficiency Video Coding) standard, the coding method provided by the invention has the advantage that the average code rates are respectively saved by 16.40% and 15.63% under the same distortion condition for video sequences with different resolutions. Compared with a traditional video compression method, the quality of the compressed low-resolution video can be effectively improved, and a good visual effect can be obtained through the method provided by the invention.