31results about How to "Reduce the number of encoded bits" patented technology

The invention discloses a method and system for speech or audio signal bandwidth expansion, which comprises: A. to simulate spectral envelope of the high-frequency signal components in the speech or audio signal. B. to make a synthesis of the said spectrum envelope and the low-frequency signal components corresponding to the high-frequency signal components in the frequency and spatial domain to obtain the reset high-frequency signal components. The invention also discloses the method and system to realize the said bandwidth expansion, the technical scheme offered by which has the advantage of less bit number of coding that can be adaptively adjusted based on the type features of the signals. Besides, by extracting spectrum envelope of the high-frequency signal components, the invention makes the fine structure acted on the low-frequency signal components corresponding frequency and spatial domain to guarantee the correlation between the reset high-frequency signal spectrum and the harmonization of the high-frequency signal spectrum lopped during coding.

The invention discloses a derivable motion compensation prediction method of a decoding terminal. The method comprises the steps of firstly, obtaining motion vectors and coordinates of encoding blocks on the left side, the upper side and the upper right side of an adjacent encoding block, and working out model parameters; secondly, determining the initial value of a candidate motion vector of the current encoding block according to the model parameters obtained by computing; thirdly, modifying the initial value of the candidate motion vector to obtain candidate motion vector set, and selecting an optimum candidate motion vector from the candidate motion vector muster; fourthly, performing compensation prediction according to the optimum candidate motion vector, and computing a rate-distortion cost when the optimum candidate motion vector is adopted to perform the compensation prediction; and fifthly, performing prediction on the current encoding block by adopting a motion compensation prediction method in H.264/AVC, and selecting a motion compensation prediction method with the lowest rate-distortion cost to perform prediction. The method can save motion vector information, and improves the compression efficiency of a zoom video; and simultaneously, when an encoding terminal is not required to encode the model parameters, the decoding terminal can directly determine the model parameters.

The invention discloses an MMVD mode-based inter-frame prediction method, a video coding method and related devices and equipment. The MMVD mode-based inter-frame prediction method comprises the following steps: constructing a first candidate list of a current coding block in an MMVD mode, the first candidate list comprising a first preset number of first candidate motion vectors; determining a second preset number of motion search amplitudes and a third preset number of motion search directions; for each first candidate motion vector, performing searching by taking the first candidate motionvector as a starting point and respectively utilizing different motion combinations of the motion search amplitude and the motion search direction to obtain motion vector offset of the first candidatemotion vector under different motion combinations; and determining the optimal motion vector of the current coding block based on the motion vector offset of each first candidate motion vector underdifferent motion combinations. According to the method, the accuracy of inter-frame prediction can be improved or the coding amount of the inter-frame prediction result can be reduced with a large probability, and the coding compression ratio is improved.

The invention discloses a method for realizing AMR-WB (adaptive multi-rate wideband) voice coding or decoding by adopting an information hiding technology, capable of reducing coding parameter length and saving voice transmission bandwidth. The AMR-WB voice coder comprises a linear prediction (LPC) analysis module, a fundamental tone search module and a fixed codebook search module (comprising an information hiding sub module). The method comprises the following steps of: (1) extracting ISF (information system finishing) parameter coding by virtue of an LPC analysis module according to a coding rate mode, and dividing the parameter coding into hidden information ST to be embedded in a step (2); and (2) searching a second best codebook according with constraint conditions (I) in a local codebook space by adopting a first depth tree search algorithm by virtue of the fixed codebook search module, and embedding the ST into a fixed codebook index to be transmitted, wherein T is a number of a codebook track and is selected from a codebook track 0, a codebook track 1, a codebook track 2 and a codebook track 3; ST is ISF index information to be embedded in a codebook track T; NT is the number of nonzero pulses contained in the codebook track T; PTk is the kth pulse in the codebook track T; and M is an embedded bit control bit, and log2M is maximum bit number of embeddable information in each track. By applying the method, the coding speed is reduced, the resource utilization factor is improved, the method is simple and feasible, and no additional computation complexity is increased.

The invention discloses a coding method of an intra-frame division mark. The coding method is used for solving the technical problem that an existing coding method is low in compressed coding efficiency. The coding method conducts a line controller unit (LCU) address with 0 depth and 0 coding division mark or address difference, and difference in coding adjacent image division manner with 1 depth, effectively reduces the number of depth 0 and 1 division mark coding bits, and improves compressed coding efficiency.

The invention discloses a voice communication method and equipment. The method comprises the following steps that a voice signal sending end determines a voice frequency band range; the voice signal sending end determinates frequency band limitation data in the audio frequency band range; the voice signal sending end uses the frequency band limitation data for carrying out coding processing on the voice signals, and sends the voice signals subjected to coding processing to the voice signal receiving end. According to the embodiment of the invention, on the basis of the characteristic of high-degree concentration of the user voice frequency range, the voice frequency band range is determined, in addition, the frequency band limitation data in the voice frequency band range is determined, and in addition, the frequency band limitation data is used for carrying out coding processing on the voice signals, so that the coding bits are put onto the most common frequency band of the user, the coding bit number is saved, the coding efficiency is improved, the redundancy in the coding range is avoided, and the network flow rate of the voice communication is reduced.

The invention provides an image prediction method, a coding tree node division method and devices thereof, and the image prediction method comprises the steps: determining the type of a current codingunit which is a brightness and chrominance coding unit, a brightness coding unit or a chrominance coding unit; determining a prediction mode of the current coding unit according to the type of the current coding unit and/or the prediction mode of an adjacent image block, wherein the image block in the current coding unit and the adjacent image block are spatially adjacent image blocks, and the adjacent image block comprises an adjacent luminance block and/or an adjacent chrominance block; and predicting the image block in the current coding unit according to the prediction mode of the currentcoding unit. According to the method in the embodiment of the invention, the video encoding and decoding efficiency can be improved.

An encoding method for an intra-frame prediction mode of a chrominance component provided by the invention comprises the following steps: determining chrominance component intra-frame prediction attribute information of a current image block; for any sub-block in the current image block, selecting a target intra-frame prediction mode for the chrominance component of the sub-block from intra-frameprediction modes matched with the chrominance component intra-frame prediction attribute information, and encoding the target intra-frame prediction mode to obtain encoding information of each sub-block; and according to the encoding information of each sub-block in the current image block, sending an encoding bit stream carrying indication information to a decoding end, the indication informationbeing used for indicating the chrominance component intra-frame prediction attribute information of the current image block. According to the encoding and decoding method, device and system for the intra-frame prediction mode of the chrominance component, the encoding bit number can be reduced, and the decoding throughput can be improved.

The invention relates to an audio arithmetic coding method based on time frequency plane context and belongs to the field of coding, wherein probability of an audio coding parameter can be self adapted. The audio arithmetic coding method based on the time frequency plane context is characterized in that firstly modified discrete cosine transformation (MDCT) and vector quantization are carried out on sound signals of each frame, and then context distribution and probability modeling are carried out on MDCT factors which are quantified. During the process, frequency domain contexts are firstly used for quantifying the MDCT factors, namely, statistics of first order condition frequency of the MDCT factors to be coded and quantified is carried out, after statistics of the first order condition frequency of each MDCT factor is carried out by the utilization of each frequency domain context, and final first order condition frequency is sent to an arithmetic encoder to be coded. The audio arithmetic coding method based on the time frequency plane context can keep good audio coding quality, meanwhile, reduce average coding rate of audio signals, and then improve coding efficiency of the audio encoder.

The embodiment of the invention provides an image filling method, a device and equipment. The method includes the steps of: selecting a prediction model for an image block comprising a filling region pixel point; according to the prediction model, determining the prediction value of the filling region pixel point in the image block; and using a decoding restore value of the predication value as afilling value of the filling region pixel point in a corresponding image block of a subsequent image frame. The embodiment uses the decoding restore value as the filling value of the filling region pixel point in the corresponding image block of the subsequent image frame, thus leading the prediction differences of the pixel points in the filling region to be zero and consequently decreasing the coding bit numbers occupied by the filling region. In addition, the embodiment also applies the prediction model for the filling region and fully utilizes the interframe or intra-frame correlation of the filling region, thus improving the interframe or intra-frame predication efficiency.

The invention discloses a beam measurement reporting method and system in 5G communication, and relates to the technical field of mobile communication, and the method comprises the steps: controllinga terminal to carry out the beam reporting in a single-branch differential mode when a corresponding indication bit in a UE-level high-level signaling is equal to 0; when the corresponding indicationbit in the UE-level high-level signaling is equal to 1, controlling the terminal to perform beam reporting by adopting a multi-branch sorting differential mode; when a single-branch differential modeis adopted for beam reporting, dynamically configuring at least one bit of signaling or semi-statically configuring at least one bit of signaling through UE-level high-level signaling to indicate a first quantization step length delta 1 in the single-branch differential mode, wherein delta 1 is larger than or equal to 1; when a multi-branch sorting differential mode is adopted for beam reporting,dynamically configuring or semi-statically configuring at least one bit of signaling through UE-level high-level signaling to indicate the branch number M and the second quantization step length delta2 in the multi-branch sorting differential mode, wherein M is greater than or equal to 2. Error propagation is reduced while beam reporting overhead is reduced.