13 results about "Prototype filter" patented technology

The invention belongs to the technical field of digital communication, and particularly relates to a multi-phase filtering channelized ultra-wideband signal detection and fusion method and system.The multi-phase filtering channelized ultra-wideband signal detection and fusion method comprises the steps that channelization is conducted, specifically, a receiving bandwidth is evenly divided into K sub-channels, 50% overlapping is adopted between every two adjacent sub-channels, the method comprises the following steps: designing an appropriate prototype filter according to frequency response, determining a multiphase filter coefficient according to the number of sub-channels and the prototype filter, carrying out D-time extraction at the front end of the multiphase filter, enabling the extracted signal to enter the multiphase filter for filtering, and then carrying out fast Fourier inverse transform on K points output each time to obtain output signals of K sub-channels, time domain amplitude detection: acquiring arrival time and pulse width information of the signal in a time domain; according to the method, the problems of missing detection and feature detection errors of multiple time domain overlapped signals in a single channel are solved, and accurate broadband signal parameters are obtained by using the corresponding relationship between the signal frequency domain and the channel passband.

This invention discloses an optimal pairwise pilot design method for a filter bank multicarrier communication system. First, based on the pairwise pilot channel estimation method, a pilot optimization model is established, and expressions for inherent interference and prototype filters are given in the model. Then, using differentiation analysis, the optimal pilot values ​​are determined under constraint boundary conditions, thus transforming the constraints of the pilot optimization model into equality constraints. Finally, optimal pairwise pilot structures applicable to different prototype filters are provided, and these structures are used as pilot designs for the pairwise pilot channel estimation method. Using this method, only two columns of pilot symbols are needed for channel estimation, and different pilot structures are adopted according to the type of prototype filter, thereby reducing pilot load while adapting to different communication scenarios.

A decoding method and an encoding method applying filter coefficient update are provided. The decoding method includes: determining a plurality of reference samples for a prediction mode; determining a prototype filter; acquiring an updated filter based on the prototype filter by trimming and / or truncating one or more coefficients of the prototype filter; and filtering the plurality of reference samples with the updated filter to generate predicted samples of a current block.

A method of processing audio is provided. The method includes determining modulated filter bank, MFB, domain broad band gains for fading an audio signal in accordance with a time domain target gain, so that application of the broad band gains in the MFB domain emulates application of the target gain in the time domain. Determining the broad band gains includes computing the broad band gains using the target gain, an MFB analysis prototype filter, and an MFB synthesis prototype filter. Also provided are corresponding apparatus, programs, and computer-readable storage media.

The invention discloses a method for realizing asynchronous sampling rate conversion in a video signal digital-to-analog conversion process, which comprises the following steps of: designing an FIR (finite impulse response) type asynchronous sampling rate conversion filter (as shown in an attached drawing) with a frequency response function of (kT / 2) / sin (kT / 2) type; and the frequency component signal gain of the video signal subjected to digital-to-analog conversion and long-distance transmission is kept constant. Therefore, various technical indexes such as a frequency response curve, a chromaticity and brightness gain difference, group delay and a signal-to-noise ratio of video signals subjected to digital-to-analog conversion and long-distance transmission can meet requirements. The FIR type asynchronous sampling rate conversion filter is realized through an FPGA, prototype filter coefficients of 24 basic phases are stored in the FPGA, filter coefficients on finer phases and optimization of the filter coefficients are realized through 3-order Lagrange interpolation, and filtering processing of 16 paths of video signals is simultaneously realized through one FPGA module, so that the performance of the asynchronous sampling rate conversion filter is effectively improved, and the filtering efficiency is improved. And the storage space of the coefficient is saved.

The present invention relates to a cascaded filter and a fundamental wave extraction method thereof, belonging to the technical field of digital signal processing. The method includes: determining an initial value of an interpolation factor M; substituting the interpolation factor into the cascaded filter to obtain a filtering result corresponding to the initial value; the cascaded filter includes a shaping filter and a masking filter, the shaping filter is an FIR filter, and the shaping filter is obtained by interpolating M - 1 zeros between adjacent impulse response coefficients of a prototype filter; the masking filter is a recursive filter; adjusting the magnitude of the interpolation factor, comparing the filtering results corresponding to different interpolation factors, and taking the value of the interpolation factor with the best filtering result as the final value of the interpolation factor; the cascaded filter performs fundamental wave extraction according to the final value of the interpolation factor. In the present invention, the increase of the M value does not affect the computational amount of the recursive filter, and thus does not affect the computational amount of the overall cascaded filter. Therefore, when the M value is relatively large, the extraction efficiency of the fundamental wave is improved.

An apparatus and method are disclosed for processing an audio signal. The apparatus includes an input interface, a digital filterbank having an analysis part and a synthesis part, a first phase shifter, a spectral envelope adjuster, a second phase shifter, and an output interface. The first phase shifter and the second phase shifter reduce a complexity of the digital filterbank, which includes both analysis and synthesis filters that are complex-exponential modulated versions of a prototype filter.

The present invention provides a dynamic digital channelization receiving method, including: establishing a mathematical model of a WOLA analysis filter bank; establishing a mathematical model of a WOLA synthesis filter bank; determining the passband cut-off frequency and the stopband start frequency of a prototype filter according to a channel division method; designing the prototype filter using the FRM technique; performing spectrum detection on the output signal of the analysis filter bank using energy detection, and inputting the detected cross-channel signal into the synthesis filter bank for reconstruction; The present invention considers the entire implementation process of dynamic channelization. By improving the FRM technique, a novel FRM is proposed for designing filters with power complementary characteristics, which are then used as the prototype filters of the dynamic channelization structure of the WOLA filter bank, reducing the computational complexity of directly designing such filters and overlapping channels to reduce signal distortion caused by blind spots.

The invention discloses a method for calculating short-time flicker and a computer device, and belongs to the technical field of power grid power. The method comprises the following steps: 1) sampling an instantaneous flicker signal to obtain a standard signal, and performing compensation calculation on the standard signal by using a correction filter to obtain a compensated signal; wherein the correction filter adopts an IIR (Infinite Impulse Response) structure as a prototype filter structure, and the coefficient of the correction filter is calculated by taking the sum of squares or the minimum sum of absolute values of deviations of all test frequency points in a flicker passband range as an objective function; and 2) calculating a cumulative probability density curve of the compensated signal under the cumulative duration of the short-time flicker, and determining the short-time flicker by using the cumulative probability density curve. Compensation calculation is carried out on the standard signal through the correction filter so as to compensate calculation result precision deviation caused by a calculation method of an IEC recommendation scheme and digitization, the accuracy of the flicker value is effectively improved, and the error between the short-time flicker value and the actual flicker value is reduced.

The invention discloses a sound field control method, a measurement device, electronic equipment and a readable storage medium, and relates to the technical field of spatial audio, and the method comprises the steps: obtaining the propagation time delay from a plurality of sound sources to at least one target listening point in a target listening area; performing pulse response modeling based on each propagation time delay to obtain a prototype filter model; performing optimization solution on the prototype filter model according to a preset target response function to obtain a target filter coefficient; wherein the objective of the optimization solution is to minimize an error between a system response function of the prototype filter model after the prototype filter model is processed by a target filter coefficient and a target response function; the target filter coefficient is used for filtering the audio input signal so as to reconstruct or form a sound field in the target listening area. According to the invention, the robustness of sound field control is improved.

The invention provides an FBMC (Filter Bank Management Controller) waveform design method for determining intelligent parameter values of a design mode, which is used for a DFT-preread FBMC communication system. According to the method, mapping from system performance indexes to prototype filter design parameters is achieved through the neural network, and the problem of parameter exhaustion search in a traditional iterative least square method is avoided. A neural network is trained through an offline data set, performance requirements are converted into design parameters, a prototype filter is obtained through an iterative least square (ILS) method, and the efficiency and flexibility of FBMC waveform design are improved. Waveform design can be realized according to system performance requirements, and the design cost is reduced.

A method of processing an audio signal is disclosed. According to embodiments of the method, magnitude response information of a prototype filter is determined. The magnitude response information includes a plurality of gain values, at least one of which includes a first gain corresponding to a first frequency. The magnitude response information of the prototype filter is stored. The magnitude response information of the prototype filter at the first frequency is retrieved. Gains are computed for a plurality of control frequencies based on the retrieved magnitude response information of the prototype filter at the first frequency, and the computed gains are applied to the audio signal.

To provide a system and a method, for a capture, a processing, and a reproducing of an audio signal.SOLUTION: The present invention provides a method for processing an audio signal. According to an embodiment of the present method, prototype filter size response information is determined. The size response information contains a plurality of gain values, and at least one of them contains a first gain corresponded to a first frequency. The size response information of the prototype filter is stored. The size response information of the prototype filter in the first frequency is read out. Each gain is calculated in regard to a plurality of control frequencies on the basis of the size response information of the prototype filter in the first frequency in regard to the plurality of control frequencies. The gain to be calculated is adopted to the audio signal.SELECTED DRAWING: None