134 results about "Bit allocation" patented technology

A system for adaptive data compression uses content-aware analysis and dynamic feedback to optimize compression quality. An adaptive quantization subsystem analyzes content characteristics of input data and determines appropriate quantization parameters. A bit allocation engine distributes available bits across different portions of the input data based on the analyzed characteristics. A quality assessment subsystem monitors the compressed output and generates parameter adjustment signals based on measured quality metrics. A feedback control subsystem then modifies the quantization parameters in response to these signals. The modified parameters are used to create optimized compressed output data from the input dataset. This dynamic, content-aware approach enables improved compression quality while maintaining efficient data reduction.

The invention discloses a reconfigurable systolic array tensor IP core based on an FPGA. The architecture comprises a core controller, a shift distributor, a data processing array and a four-stage data cache. The tensor core can adapt to general tensor multiplication and addition operations in Prefill and Decode stages of the generative neural network at the same time through reconfigurable design. The core controller receives the tensor multiplication instruction and the configuration information and decouples the tensor multiplication instruction and the configuration information according to a data stream to generate a plurality of block tensor multiplication instructions. The shift distributor decodes a block tensor multiplication instruction, decoded mode information controls data communication between a first-level cache and a second-level cache (Distrided RAM and Bram), a control signal of a data processing array is generated, and the control signal enters the array together with data to flow. And the data processing array is connected with different secondary BRAM caches according to the received configuration information and is used for processing data in a bubble-free pulsating flow manner. The four-stage data cache realizes the multi-stage pinpong operation of the loading of the operation data of the tensor operation core, and realizes the multi-stage pipeline design of the data from the off-chip HBM to the on-chip URAM, from the URAM to the BRAM and from the BRAM to the operation unit. Aiming at a single-user edge end generation type network reasoning task, a core controller generates a data processing unit mode configuration signal in a fine-grained (tensor element level) manner, and the data processing unit mode configuration signal is transmitted by a shift distributor and flows in parallel with operation data, so that the mode configuration efficiency and the hardware resource utilization rate are greatly improved, the problem of high fan-out caused by broadcast control is reduced, and the user experience is improved. A control flow is designed by means of the structural characteristics of the systolic array, and a complex control algorithm is realized with low resource occupation.

The invention relates to the technical field of quantum compilation, and discloses an error suppression compilation optimization method for a quantum computer, which comprises the following steps of: capturing a physical quantum gate operation error rate and a hardware topological structure of a quantum processor for executing a mapping task, and constructing a hardware feature matrix; capturing gate interaction attributes of a to-be-compiled quantum logic circuit, and constructing a circuit characteristic matrix; according to the constructed hardware feature matrix and circuit feature matrix, establishing a mapping scheme representation model and a corresponding overhead function; solving the overhead function by using a genetic algorithm, and searching a mapping scheme which minimizes the overhead function value under the condition of meeting the resource constraint of the quantum processor; applying the optimized mapping scheme to the quantum logic circuit to be compiled, and compiling and optimizing the quantum logic circuit to be compiled; according to the method, the problems of poor adaptability, insufficient error rate optimization and the like of a quantum bit allocation scheme in the prior art are solved, and low-error and reliable quantum circuit mapping processing is realized.

The invention relates to the technical field of speech recognition, and discloses a real-time speech recognition method based on a Bluetooth audio stream, which comprises the following steps: analyzing bit allocation parameters to calculate quantized bit distribution and generate a frequency domain confidence mask, monitoring a packet loss concealment state flag bit of a decoder, forcibly setting the mask as a blocking threshold when an algorithm is activated, and generating a real-time speech recognition result. According to the method, a cross-level feature purification mechanism based on protocol priori and link states is constructed, acoustic model illusion caused by forged waveforms is blocked through targeted arbitration while deterministic quantization noise is eliminated, and the acoustic model recognition accuracy is improved. And the identification accuracy under a severe channel is ensured.

The invention relates to an underground video coding method and device, and the method comprises the steps: obtaining to-be-coded video data based on the perception capability evaluation and video reconstruction of an underground video stream; obtaining a time-space domain saliency model based on the human eye visual perception structure; and carrying out bit allocation and code rate allocation on the to-be-coded video data based on the time-space domain saliency model, and determining coded data corresponding to the underground video stream. The underground video is subjected to lossless reconstruction based on a human visual perception mechanism, so that the visual redundancy is removed, the code rate is reduced, and the underground bandwidth is relieved; bit allocation and code rate allocation are performed on the to-be-coded video data through the time-space domain saliency model, so that the coding efficiency and the visual experience degree are remarkably improved, and the method has wide application value.

A zero-delay panoramic video bit rate control method that considers distortion temporal propagation aims to optimize target bit allocation and achieve global rate-distortion optimization in coding. It encompasses coding tree unit (CTU)-level bit rate control and temporal global rate-distortion optimization. CTU-level bit rate control involves optimizing target bit allocation and updating bit rate control parameters. Temporal global rate-distortion optimization utilizes the reconstruction error and motion compensation prediction error information of the previous coded frame to estimate the temporal dependence between CTUs in the current coding frame and those in the previous coded frame to adjust the coding parameters of the current CTU. The coding parameters are further fine-tuned according to the area stretching ratio encountered during the projection of the panoramic video from a 3D spherical surface to a 2D plane, taking into consideration the detrimental impact of interpolated redundant pixels on the coding process.

The invention provides a region-of-interest protection-oriented power image compression method and system and a code stream transmission method, and relates to the technical field of power image compression. According to the method, the region of interest and the background region are distinguished, the background region is smoothed through bilinear downsampling, spatial redundant information of the background region of the power image is removed, and bit allocation of the background region is effectively reduced, so that lower compressed code streams are consumed under the condition that the fidelity of the region of interest and the same background quality are achieved. Meanwhile, through a content weighted attention module, adaptive bit distribution of different spatial positions is realized, and bits can be reasonably distributed according to the importance difference of image contents, so that the coding quality of the region of interest and the overall compression effect are improved. According to the code stream transmission method provided by the invention, the code stream required by each path of image is forwarded in a staggered and ordered manner, so that transmission bandwidth resources are better utilized.

The invention relates to the technical field of artificial intelligence and automatic driving, and discloses a space-time constraint end-to-end automatic driving system and method based on a unified VLA model, and the system comprises a unified encoder, a computing power management unit, a core neural network and an output decoder. A gating operator embedded into a core neural network is used for calculating characteristic information entropy and computing power consumption in real time, and through topology reconstruction logic based on entropy and computing power joint constraint, computational graph collapse operation is dynamically triggered at an instruction level, so that network reasoning depth is adaptively expanded along with physical time constraint, and the network reasoning accuracy is improved. According to the method, isomorphic mapping of the topological structure of the calculation model and the physical time constraint is established, the deterministic delay upper limit under the extreme working condition is determined, the problem of dimension mismatching of an existing static calculation graph under the hard real-time constraint is solved, and optimal bit allocation of calculation resources in the time dimension is achieved.

The invention discloses a soft decoding method and device, a medium and a product, and relates to the technical field of error correction coding, and the method is applied to a low-density parity check code, and comprises the steps: obtaining a target low-density parity check code with an information node and a check node containing an erasure bit; and allocating a preset non-zero log-likelihood ratio to the erasure bit on the check node, and executing first soft decoding. And if the first soft decoding fails, judging whether to carry out second soft decoding or not based on the weight of the syndrome generated by the first soft decoding and the number of the erasure bits on the information node. If it is judged that the second soft decoding needs to be carried out, a preset non-zero log likelihood ratio is distributed to the erasure bit located on the information node, and the second soft decoding is carried out. According to the method and the device, the erasing bits on different nodes are conditionally processed in stages, so that the success rate and the efficiency of decoding are effectively improved.

The invention discloses a screen content video coding perception code rate control method and device based on DCT domain JND, and relates to the field of video coding, and the method comprises the steps: obtaining a screen content video, and obtaining a DCT coefficient of the video through discrete cosine transform; guiding frame-level and CTU-level target bit allocation by using the texture energy weight; obtaining a JND threshold value of the DCT domain according to a product based on the spatial contrast sensitivity threshold value, the brightness masking modulation factor and the contrast masking adjustment factor; and deducing a Lagrange multiplier combined with a JND regulation factor by using a JND threshold value to realize code rate control. According to the method, the JND model suitable for the screen content video is established in the DCT domain, bit distribution is guided through the texture energy weight, the new rate distortion model is established, the coding rate distortion performance can be improved, and the code rate control precision of the screen content video is improved.

Disclosed are a method and device for allocating bits of an audio object, which relates to the technical field of audio coding and decoding, and is helpful to improve the overall quality and coding efficiency of reconstructed audio objects. The method comprises: pre-rendering a plurality of audio objects to be pre-rendered in an audio frame respectively to obtain a plurality of pre-rendered audio objects; obtaining a perceptual importance parameter value of each of the plurality of pre-rendered audio objects, wherein the perceptual importance parameter value of the current pre-rendered audio object is used to indicate the degree of perceptual importance of the current pre-rendered audio object among the plurality of pre-rendered audio objects; obtaining a bit allocation parameter value of the current audio object to be pre-rendered based on the perceptual importance parameter values ​​of each of the plurality of pre-rendered audio objects; and determining a target number of bits to be allocated for the current audio object to be pre-rendered based on the bit allocation parameter value of the current audio object to be pre-rendered and the total number of bits to be allocated corresponding to the plurality of audio objects to be pre-rendered. The method can be applied to a stereo encoder or a multi-channel encoder.

The invention provides a multi-satellite cooperative beam position distribution method, system and device, and an electronic device, which can solve the technical problem of low operation efficiency in a multi-satellite multi-beam position scene in the prior art. The method comprises the following steps: determining a coverage wave position set of a to-be-distributed satellite at a first moment; determining a to-be-transmitted service priority of each coverage wave position in the coverage wave position set; according to the priority of the service to be transmitted, based on a first preset condition, determining a first target satellite of each coverage wave position; the remaining irradiation time of the first target satellite is greater than or equal to a preset beam position distribution period; and according to the first target satellite and the coverage wave position, obtaining a wave position distribution result of each to-be-distributed satellite. And invalid calculation is reduced through priority screening of to-be-transmitted services. And screening the first target satellites of which the residual irradiation time is greater than or equal to the preset period, excluding satellites of which the time does not reach the standard, simplifying a matching link and ensuring continuity, thereby preventing multiple satellites from competing for resources, reducing interference and reducing energy loss.

The invention provides a quantization method and a quantization device, which can perform singular value decomposition on a first matrix formed by word embedding vectors to obtain the maximum characteristic value of each block matrix in M block matrixes, and can determine the quantity of quantization bits of each block matrix according to the maximum characteristic value of each block matrix. The method comprises the following steps of: dividing each block matrix into a plurality of block matrixes, quantifying each block matrix to obtain each quantized block matrix, combining each quantized block matrix to obtain a quantized first matrix, and representing the amount of semantic information carried in each block matrix by a maximum characteristic value of each block matrix to a certain extent, according to the embodiment of the invention, the quantized bit number can be allocated to each block matrix according to the amount of semantic information carried by each block matrix, so that different bit numbers can be allocated to the block matrixes carrying different semantic information, differential bit number allocation can be realized, semantic information loss can be reduced, and user experience can be improved.

The invention relates to the technical field of oral images, and discloses an oral panoramic image real-time compression transmission and 3D reconstruction cloud system. The system comprises an image acquisition module, a compression processing module, a cloud transmission module and a 3D reconstruction module. The image acquisition module acquires initial image data and pre-processes the initial image data to obtain target image data; the compression processing module generates a compression strategy through block coding, and when conditions are met, sub-compression units are divided, and dynamic bit allocation is optimized; the cloud transmission module extracts the compressed code streams, constructs real-time and delayed transmission queues in a classified manner, ranks the queues, verifies the integrity of the code streams to be verified and counts the complete number; and the 3D reconstruction module determines an effective data size according to the complete number and the compressed code streams which do not pass the verification, and starts a cloud reconstruction engine. The system improves the oral image processing efficiency and the 3D reconstruction quality through dynamic compression, intelligent transmission and precise verification, and is suitable for oral medical diagnosis scenes.

The invention relates to the technical field of image compression, in particular to a plane image compression method, and provides the following scheme: obtaining a to-be-compressed plane image; carrying out feature extraction on the plane image to obtain image features, and calculating a task value density map and uncertainty corresponding to the task value density map; generating a quantization step size and a bit allocation weight according to the task value density map and the uncertainty, and performing compression transformation on the plane image to obtain a main code stream; further generating invariant feature side channel and task consistency indication information according to the image features, and performing compression to obtain an auxiliary code stream and a constraint code stream; and sending the main code stream, the auxiliary code stream and the constraint code stream to a decoding end. According to the method, the readability and testability of key structures such as character strokes, table grids and boundary curvatures can be guaranteed under the condition of limited code rates, and the consistency of subjective visual quality and machine recognition performance is realized.

In communication devices and methods configured to communicate with each other, bits of a continuous bit stream are allocated to resource units of different sizes. Various options are disclosed that differ in both implementation complexity and diversity achieved. Furthermore, exemplary implementations are provided that consider the definition of resource units of IEEE 802.11ax as a baseline and outline the operation of the bit allocation for combination of resource units of various sizes.

The invention discloses a high-definition video low-delay compression method based on self-adaptive adjustment, and aims to solve the problems of insufficient self-adaptive adjustment capability, too high encoding and decoding delay and low code rate and network bandwidth matching degree in the existing high-definition video compression technology. The method comprises the following steps: firstly, acquiring a high-definition video stream and a real-time network bandwidth parameter, extracting intra-frame texture complexity, inter-frame motion intensity and scene switching frequency characteristics, and constructing a two-dimensional adaptive compression strategy in combination with a real-time bandwidth; low-delay coding is executed by adopting parallel frame fragmentation and simplified adaptive arithmetic coding, adaptive code rate control is realized by combining code rate overflow early warning and intra-frame key region priority bit allocation, and finally, fast inverse transformation, quantization recovery and motion compensation optimization are completed at a decoding end based on strategy parameters synchronously transmitted at a coding end so as to reconstruct a high-definition video frame. According to the invention, the adaptive adjustment capability of video compression is improved, low time delay of coding and decoding is realized, and the method is suitable for real-time high-definition video transmission scenes such as remote monitoring and video conferences.

Methods, systems, and devices for wireless communications are described. In some examples, a device may compute the coding rate associated with each coding level of coded modulation scheme based on one or more channel conditions of a wireless channel. The device may allocate multiple information bits among multiple coding schemes associated with a multi-level coding scheme. The quantity of information bits allocated to each coding scheme may be based on the coding rates associated with one or more coding levels corresponding to the coding scheme. The device may encode an information payload using the multiple coding schemes in accordance with the allocation of the multiple information bits. The device may transmit the encoded information payload via the wireless channel. The receiving device may compute the coding rate associated with each coding level of the multi-level coding scheme and allocate the information bits accordingly. The receiving device may decode the payload.

The present invention belongs to the field of image processing technology, and specifically relates to a zero-delay panoramic video rate control method considering the propagation of time domain distortion. The method of the present invention is mainly to optimize the target bit allocation and global rate distortion optimization coding, including rate control at the coding tree unit (CTU) level and time domain global rate distortion optimization. CTU-level rate control mainly includes optimizing target bit allocation and updating rate control parameters; time domain global rate distortion optimization mainly uses the reconstruction error and motion compensation prediction error information of the encoded frame to estimate the time domain dependency of each CTU in the current encoded frame and adjust the rate distortion optimization parameters of the CTU accordingly, and at the same time, according to the process of projecting the panoramic spherical graphic onto the 2D plane (ERP projection), the characteristics of oversampling and increasing pixel redundancy will be generated at the two poles, and the coding parameters are adjusted according to the projection area change ratio. The present invention can effectively shorten the encoding time and improve the video encoding quality under the condition of a given bandwidth.

Methods, systems, and devices for wireless communications are described. In some examples, a device may compute the coding rate associated with each coding level of coded modulation scheme based on one or more channel conditions of a wireless channel. The device may allocate multiple information bits among multiple coding schemes associated with a multi-level coding scheme. The quantity of information bits allocated to each coding scheme may be based on the coding rates associated with one or more coding levels corresponding to the coding scheme. The device may encode an information payload using the multiple coding schemes in accordance with the allocation of the multiple information bits. The device may transmit the encoded information payload via the wireless channel. The receiving device may compute the coding rate associated with each coding level of the multi-level coding scheme and allocate the information bits accordingly. The receiving device may decode the payload.

This invention discloses a method for UAV communication and synchronization control based on a multi-timescale dual-channel architecture, comprising the following steps: S1, establishing a nonlinear singular perturbation system model for each UAV node; S2, configuring dual channels for each UAV node; S3, selecting to transmit either the original measurement value or the differential value; S4, generating quantized symbols and transmitting them to a remote controller via the corresponding channel; S5, decoding the received quantized symbols to obtain the reconstructed measurement value; S6, solving for the controller gain that meets stability requirements; S7, designing a controller based on the reconstructed measurement value and the controller gain, generating control inputs, and applying them to the UAV nodes to achieve high-precision synchronization control of the UAV system network nodes. This invention achieves high-precision synchronization control of UAV system network nodes and ensures system stability under conditions of limited total bit rate through dual-channel event-triggered differential transmission and dynamic bit allocation.

A method, device, equipment and storage medium for extending the frequency band of an audio signal, belonging to the field of communication technology. The method comprises: receiving a bit stream sent by an encoding device, decoding the bit stream to obtain a decoded audio frequency domain signal; in response to the highest frequency point of the bit allocation of the audio frequency domain signal being lower than the starting frequency point of the preset bandwidth extension band, or the frequency band of the bit allocation of the audio frequency domain signal being smaller than the preset bandwidth extension starting band, predicting the spectrum signal between the highest frequency point of the bit allocation and the highest frequency point of the preset bandwidth extension band based on the spectrum signal within the predetermined frequency band range or the predetermined frequency point range in the audio frequency domain signal. The method can avoid the situation where "there is no spectrum signal between the highest frequency point of the bit allocation and the starting frequency point of the preset bandwidth extension band", ensure the balance of high and low frequency energy within the frame, avoid the mechanical feeling caused by spectrum holes, and improve the quality of reconstructed audio.

The invention provides a beam position resource allocation method and device and computer equipment, which can be applied to the technical field of wireless communication. Based on the method, a first scheduling weight of each service in M services corresponding to each user in N users can be determined, N * M first scheduling weights are obtained, the first scheduling weights are demand intensities of the services corresponding to the users for time slot resources in wave positions to which the services belong, and N and M are positive integers; according to N * M first scheduling weights corresponding to each beam position in B beam positions in a target adjustment period, a second scheduling weight of each beam position in the B beam positions is determined, the second scheduling weight is the demand intensity of the beam position for time slot resources in a beam, and B is a positive integer; determining an air interface load of each wave position in the B wave positions in the target adjustment period; and according to the second scheduling weight and the air interface load of each beam position in the B beam positions, allocating time slot resources to each beam position in the B beam positions.

The application relates to the technical field of signal processing, in particular to a bit allocation method, device and equipment for bit pool optimization of an encoder. The method comprises the following steps: obtaining basic information of an input audio of the encoder; determining a current encoding mode of the encoder according to the basic information; if the current encoding mode is a constant bit encoding mode, performing constant bit encoding according to a total number of frames in the basic information, and outputting a code stream subjected to the constant bit encoding; and if the current encoding mode is a dynamic bit encoding mode, performing dynamic bit encoding according to the total number of frames in the basic information, and outputting a code stream subjected to the dynamic bit encoding, wherein the dynamic bit encoding comprises dynamically adjusting a bit allocation strategy according to a window length of a current frame. Thus, the problem that a fixed bit encoding mode is adopted in a low code rate mode in the prior art, resource requirements of long window frames and short window frames cannot be met at the same time, and audio quality is poor is solved.

Embodiments of this application disclose a three-dimensional audio signal processing method and apparatus, to implement bit allocation of a signal. The method includes: performing spatial coding on a to-be-coded three-dimensional audio signal, to obtain a transmission channel signal and transmission channel attribute information, where the transmission channel signal includes at least one virtual speaker signal group and at least one residual signal group; and determining a bit allocation ratio of the virtual speaker signal group and a bit allocation ratio of the residual signal group based on the transmission channel attribute information.

This invention provides a power image compression method, system, and bitstream transmission method for region-of-interest (ROI) protection, relating to the field of power image compression technology. This invention distinguishes between the ROI and the background region, smoothing the background region through bilinear downsampling to remove spatial redundancy information in the background region of the power image, effectively reducing bit allocation in the background region. This achieves lower bitstream consumption while maintaining ROI fidelity and the same background quality. Simultaneously, through a content-weighted attention module, adaptive bit allocation at different spatial locations is implemented, rationally allocating bits according to the importance of image content to improve ROI encoding quality and overall compression performance. The bitstream transmission method proposed in this invention forwards the bitstream required for each image in a staggered and orderly manner to better utilize transmission bandwidth resources.

The present application discloses a VVC code rate control optimization method based on HVS characteristics from the two problems of bit allocation in VVC not matching human visual characteristics and the convergence step being too small when updating parameters and not considering absolute convergence. First, the spatial domain information is obtained by using the Sobel operator; the temporal domain information is adaptively obtained by using the MV information of the reference frame; a special case of temporal domain information extraction method is proposed; then, the frame target bits are divided into two parts, and the CTU level bit allocation is carried out by using the spatial domain information of each CTU; after the completion of a CTU or a frame coding, the quasi-Newton convergence algorithm is used for parameter updating, and the corresponding method is proposed for the convergence of special cases. The video sequence obtained based on the present application can better meet the needs of the human visual system, and the objective quality index and subjective quality can be greatly optimized, and the fluctuation degree of the output coded video is more stable.

An improved gain-shape vector quantization is achieved by determining a number of bits to be allocated to a gain adjustment-and shape-quantizer for a plurality of combinations of a current bit rate and a first signal property. The bit allocation is derived by using an average of optimal bit allocations for a training data set. A number of bits to the gain adjustment and the shape quantizers for a plurality of combinations of the bit rate and a first signal are pre-calculated, and a table indicating the number of bits to be allocated to the gain adjustment-and the shape-quantizers for a plurality of combinations of the bit rate and a first signal property is created. In this way, the table can be used for achieving an improved bit allocation.

The application discloses a high-definition video low-delay compression method based on adaptive adjustment, and aims to solve the problems of insufficient adaptive adjustment capability, high coding and decoding delay and low matching degree of code rate and network bandwidth in the prior high-definition video compression technology. First, high-definition video stream and real-time network bandwidth parameters are acquired, and features of intra-frame texture complexity, inter-frame motion intensity and scene switching frequency are extracted, and a double-dimension adaptive compression strategy is constructed in combination with real-time bandwidth; low-delay coding is executed by adopting parallel frame slicing and simplified adaptive arithmetic coding, adaptive code rate control is realized by combining code rate overflow early warning and intra-frame key area priority bit allocation, and finally, fast inverse transformation, quantization recovery and motion compensation optimization are completed based on the strategy parameters synchronously transmitted by the encoding end at the decoding end to reconstruct high-definition video frames. The application improves the adaptive adjustment capability of video compression, realizes low coding and decoding delay, and is suitable for real-time high-definition video transmission scenes such as remote monitoring and video conference.

An apparatus comprising means for: obtaining for a plurality of audio objects within an audio environment, respective audio signals and directional metadata values; obtaining at least one ratio parameter with respect to a respective audio object of the plurality of audio objects and with respect to a respective time element and a respective frequency element of a frame, the at least one ratio parameter configured to identify a distribution of the audio object within an object part of the total audio environment; obtaining a further ratio parameter associated with the distribution of the object part of the total audio environment; determining an object priority order based on the ratio parameter and the further ratio parameter; determining quantization bit allocations for the directional metadata values associated with the plurality of audio objects based on the determined object priority order, and quantizing the directional metadata values.