237 results about "Decimation" patented technology

The invention relates to a method for realizing multi-channel ADC (Analog to Digital Converter) data synchronization processing based on an FPGA (Field Programmable Gate Array), which comprises the following steps that: the FPGA receives a clock frequency generated by an external clock generator and takes the clock frequency as a working clock of the FPGA; a plurality of groups of same-frequency clocks and same-phase clocks are generated in the FPGA and are respectively used as working clocks of the ADC chips; the FPGA receives sampling data of each ADC, an enable signal is set, and when the enable signal is effective, digital down-conversion, decimation filtering and FIR filtering are carried out on each sampling data; and after FIR filtering is completed, another enable signal ENend is generated, and data are further stored or transmitted to an upper computer. According to the method and the device for realizing the multi-channel ADC data synchronous processing based on the FPGA, the processor and the computer readable storage medium thereof, synchronous start and stop of the processing flow are controlled through the enable signals ENstart and ENend, synchronous processing of the multi-channel ADC data is ensured, accuracy and effectiveness are improved, and the method and the device have a wide application range.

The invention relates to the technical field of digital demodulation, in particular to a digital demodulation method suitable for deep ultraviolet light communication, which comprises the following steps of: adopting multiple delay processing with half code element width, and respectively carrying out double extraction processing on a discrete sampling signal which is not subjected to delay processing and a plurality of delay sampling signals to obtain a plurality of delay extraction signals; in this way, the key sampling time in the coverage code element period can be ensured, and a basis is provided for subsequent accurate synchronization; a plurality of delay extraction signals are respectively input into corresponding equalization processing units, a preset local training sequence is loaded to each equalization processing unit to obtain a plurality of equalization filtering signals, and the plurality of equalization filtering signals and the preset local training sequence are respectively subjected to related judgment processing, so that the inter-symbol interference is inhibited, and meanwhile, the equalization processing efficiency is improved. The synchronization signal can be quickly screened out, so that the problem that inter-symbol interference and synchronization are difficult to consider in a traditional demodulation method is solved, and the demodulation efficiency and reliability are remarkably improved.

The invention belongs to the technical field of cognitive radio, and particularly relates to an open set modulation identification method based on time-frequency domain feature learning and fusion. The method comprises the following steps of: receiving a signal by adopting a receiving antenna through electromagnetic spectrum monitoring equipment, capturing a space electromagnetic radiation signal, converting the space electromagnetic radiation signal into an electric signal, amplifying the signal through a low-noise amplifier, performing frequency conversion and filtering through an analog receiver, and converting the received signal into an intermediate-frequency signal. Then, the signal is digitalized through AD sampling, and the intermediate frequency signal is converted into a baseband complex signal through frequency mixing and digital filtering in a digital receiver. Then performing time domain characterization calculation and frequency domain characterization calculation on the signal sampling data to form a plurality of time domain characterization vectors and a plurality of frequency domain characterization vectors; and then the vectors are sent to a trained deep neural network model, reasoning is carried out by using the network model, a reasoning result is post-processed, and a signal modulation identification result is finally obtained. According to the invention, the recognition accuracy of unknown signals is improved.

The invention relates to a full-dynamic exponential increment type low-delay high-precision analog-to-digital converter, which comprises a loop integrator used for receiving an ith-round input signal, and sequentially performing addition processing and integration processing on the ith-round input signal and a feedback signal calibrated by an (i-1) th round to obtain an ith-round integration signal; the first-order exponential increment error feedback noise shaping SAR quantizer is used for carrying out addition processing and multi-bit quantization on the ith round integral signal and the double (i-1) th round residual signal to obtain an ith round quantization signal; the digital-to-analog conversion module is used for sequentially performing format conversion, mismatch calibration and digital-to-analog conversion on the (i-1) th round of quantized signals in the process of processing the ith round of input signals to obtain feedback signals after the (i-1) th round of calibration; and the digital decimation filter module is used for carrying out digital filtering and down-sampling processing on the ith-round quantized signal to obtain an ith-round Nyquist rate output signal. The system can realize higher significant digits in shorter conversion time.

The invention provides a frequency spectrum splicing method and device, equipment, a storage medium and a program product, and is applied to the technical field of signal processing. The method comprises the following steps: determining a target number of analog-to-digital converters according to a target sampling bandwidth and a sampling rate of a single analog-to-digital converter, and collecting initial broadband signals by adopting the analog-to-digital converters with the target number in different frequency bands; dividing the frequency spectrum of the initial broadband signal into a plurality of sub-bands, and performing target processing on the sampling rate of each sub-band in the plurality of sub-bands to obtain a target sampling rate; performing fast Fourier transform processing on the target sampling rate based on fast Fourier transform points to obtain frequency domain data; performing frequency spectrum splicing on the frequency domain data to obtain a target frequency spectrum; wherein the target processing comprises the steps of sequentially carrying out time domain filtering processing, and adjusting the sampling rate through decimation filtering or interpolation filtering according to the target frequency resolution.

The present invention relates to a method for repopulating a 3D design. The method for compressing a 3D model is also described. A system for performing the methods is also provided.The method for repopulating a digital design of a building is provided comprises exporting one or more model files from a first project, importing the exported one or more files into a second project, and uploading one or more object files to the second project, wherein the one or more object files are matched to the exported one or more object files.The method for compressing a 3D model comprises determining a decimation range for a number of objects, and removing polygons for the objects based on the decimation range.

An apparatus for calibrating a digital filter to replicate a transfer function of a signal processing device comprising a detector with a first input to receive a first signal; a second input to receive a response signal of the signal processing device to the first signal; a signal modification block; a comparison block and a decimation block; wherein the comparison-block compares a phase and amplitude of the first signal after a correction has been applied by the signal modification block and decimation has been applied by the decimation block; and a feedback loop; wherein detector is configured to determine at least a feedback control signal at a first frequency and a second frequency, different to the first frequency and determine calibration information for programming of the transfer function of said digital filter.

A device may encode a first set of pictures of the video data to generate first encoded video data; generate first error correction data based on the first encoded video data; transmit the first error correction data and the first encoded video data to a receiving device; receive, from the receiving device, a decimation pattern indication that indicates a decimation pattern determined based on the first set of pictures, the decimation pattern being a pattern of encoded video data non-transmission; encode a second set of pictures of the video data to generate second encoded video data; generate second error correction data based on the second encoded video data; apply the decimation pattern to the second encoded video data to generate decimated video data; and transmit the second error correction data and the decimated video data to the receiving device.

The invention discloses an SAR ADC nonlinear calibration algorithm and device based on harmonic distortion component extraction. The SAR ADC nonlinear calibration algorithm comprises the steps of obtaining a digital output matrix of an SAR ADC; separating the digital output matrix according to columns to construct a new digital output matrix; calculating a digital summation vector according to the ideal weight vector and the new digital output matrix; calculating a normalized frequency of the plurality of odd harmonics; for each odd harmonic, the method comprises the following steps: carrying out decimation filtering on a digital summation vector according to a normalized frequency to obtain a harmonic vector; constructing a new linear combination expression of a digital output matrix and a harmonic vector according to the to-be-estimated harmonic weight vector; solving the linear combination expression to obtain a to-be-estimated harmonic weight vector; calculating a total to-be-estimated harmonic weight vector according to the to-be-estimated harmonic weight vectors of all odd harmonics; and calculating a calibration weight vector according to the ideal weight vector and the total to-be-estimated harmonic weight vector. According to the invention, no circuit needs to be embedded, the portability is high, the hardware overhead is small, the calibration time is short, and the calibration precision is high.

The present document describes a method (200) for determining N coefficients of an asymmetric prototype filter p0 for use in a low delay M-channel analysis and / or synthesis filter bank (101, 102) comprising M analysis filters hk (103) and / or M synthesis filters fk(106), k=0, . . . , M−1, wherein M is greater than 1, and wherein subband signals which are processed by the analysis and / or synthesis filter bank (101, 102) are decimated by a decimation factor S, with S<M. The method (200) comprises determining (201) a target transfer function of the analysis and / or synthesis filler bank (101, 102) comprising a target delay D; wherein D is smaller or equal to N. Furthermore. the method (200) comprises determining (202) a composite objective function etot comprising a transfer function error term e and an aliasing error term ea; wherein the transfer function error term et is associated with a deviation between a transfer function of the analysis and / or synthesis filter bank (101, 102) and the target transfer function; and wherein the aliasing error term ea is associated with errors incurred

The invention relates to the technical field of broadband signal synthesis and high-speed digital-to-analog conversion, in particular to an oversampling-based real-time frequency interleaving digital-to-analog conversion system aliasing suppression method and system, and aims to solve the problem of spectrum aliasing caused by non-ideal characteristics of a digital sub-band decomposition filter and an analog filter under a real-time architecture of the system. In the invention, the sub-band decomposition is completed by adopting the linear phase complementary filter bank, so that distortion introduced by a transition band of a digital filter is avoided; by restraining system parameters such as passband and stopband frequencies and local oscillation frequency of a sub-band decomposition filter, a protection band is set by an oversampling mechanism, a frequency margin is reserved for a transition band of the filter, and aliasing components are suppressed. According to the method, extraction aliasing of a digital domain and mirror image aliasing of an analog domain are avoided at the same time from the system architecture, the frequency spectrum purity of output signals is improved, the method is suitable for a real-time system for processing continuous data streams, and technical support is provided for achieving a high-performance broadband signal source.

The invention discloses a software radio digital signal frequency conversion method and system and a medium. Relates to the technical field of digital signal processing. Comprises: acquiring an input signal; performing complex multiplication on the input signal and the carrier signal to obtain a frequency conversion signal; based on an adjustable filtering extraction module, performing integral multiple or rational multiple filtering extraction on the frequency conversion signal to obtain an output signal; wherein rational multiple filtering extraction is realized by a numerical control oscillator and an interpolation unit; according to the scheme, on the basis of the traditional technology of realizing the digital down converter on the FPGA, the method and the structure are improved, and on the basis of an adjustable filtering extraction module, integral multiple or rational multiple filtering extraction is carried out on a frequency conversion signal to obtain an output signal; free adjustment of the digital down-conversion multiple is achieved, and rational number multiple configuration of digital down-conversion is achieved based on the numerically controlled oscillator and the interpolation unit.

Systems and methods for compensating a transmit signal for charge trapping effects of a power amplifier are provided. In certain embodiments, a non-linear filter is trained based on time aligning a first set of observations taken from digital transmit data prior to conversion to a radio frequency transmit signal, and a second set of observations taken from an output of a power amplifier that amplifies the radio frequency transmit signal. In certain implementations, the first set of observations and the second set of observations are obtained without decimation. Rather, decimation is provided after timing alignment. By implementing the DPD system in this manner, signal data is not lost by decimation and more accurate timing alignment between the sets of observations is achieved.

The invention discloses a high-order digital decimation filter with a variable decimation multiple, which comprises a cascade comb filter used for downsampling and anti-aliasing filtering, filtering high-frequency components and obtaining low-frequency density signals, and a compensation filter connected to the output end of the cascade comb filter and used for compensating passband attenuation and performing double decimation on the signals, and the output end of the compensation filter is connected with a half-band filter for realizing third-stage signal extraction. The invention further discloses a filtering method of the high-order digital decimation filter with the variable decimation multiple. Aiming at the defects that a traditional single-stage FIR filter is high in hardware requirement, complex in design, high in filter order and not easy to implement and the like, and aiming at the defect that a CIC filter is insufficient in passband flatness, a CFIR compensation filter is used for compensation, and finally a half-band filter is used for frequency reduction.

The invention provides a digital filter circuit and method of a sigma-delta analog-to-digital converter. Relates to the technical field of digital decimation filters, and solves the problems that in the prior art, the line decimation rate is single, and output of a first data is delayed. The input data of the sigma-delta analog-to-digital converter is sent to one of the integrator comb filter and the low-delay filter, and the filter receiving the input data is used as a decimation filter; and the decimation filter is used for opening the enabling of the decimation filter according to the configuration information, adjusting the decimation rate of the decimation filter according to the configuration information, and filtering and sampling the input data according to the decimation rate to obtain filter output data. The digital decimation filtering circuit is suitable for the requirements of low time delay and multiple decimation rates, the time delay for outputting the first data can be reduced by selecting the filter path, and the digital decimation filtering with multiple decimation rates and low time delay can be realized.

This invention provides a phase code digital low-pass filter circuit and a re-timer based on a PI-based architecture. The phase code digital low-pass filter circuit includes a first accumulation unit and a downsampling decimation module. The first accumulation unit accumulates the initial frequency control information generated by the clock data recovery module of the re-timer to generate a phase signal. The downsampling decimation module downsamples the phase signal to generate a filtered phase code. The filtered phase code is used to generate the transmit channel clock. This invention effectively solves the problems of high implementation cost, large footprint, and easy introduction of additional analog non-ideal factors in existing digital filtering solutions.

The invention discloses a piecewise linear quantization mixed structure analog-to-digital converter with a noise shaping function, and belongs to the field of analog-to-digital converters, the analog-to-digital converter comprises an input control circuit, a nonlinear quantization circuit, a noise shaping modulation circuit, an output control circuit and a digital decimation filter circuit, the control circuit samples an input signal; the non-linear quantization circuit determines a non-linear section where an input signal is located and provides reference voltage for the noise shaping modulation circuit; the noise shaping modulation circuit is used for performing noise shaping and fine quantization on an input signal; the output control circuit combines the two parts of quantization output and generates a high-speed low bit code stream; the digital decimation filter circuit receives a high-speed code stream and generates a high-precision digital code through decimation filtering; according to the technology provided by the invention, suitable nonlinear quantization characteristics are provided according to different signal characteristics, and meanwhile, higher quantization precision and a wider dynamic range can be realized.

This invention relates to the field of downsampling filter technology, and more particularly to a cascaded digital filtering device for large-scale downsampling. It includes: inputting initial information at a high sampling rate into a first-stage variable decimation factor cascaded integral-comb filter bank; after preliminary downsampling, the output is sent to a second-stage fixed decimation factor cascaded integral-comb filter bank; the second-stage fixed decimation factor cascaded integral-comb filter bank comprises three sequentially connected 16x decimation cascaded integral-comb filters; after high-fold downsampling, the data is output to a third-stage compensation filter to compensate for the large passband attenuation generated by the previous cascaded integral-comb filter bank and increase stopband attenuation; then the data is output to a fourth-stage decimation filter bank for multi-stage decimation and further downsampling, finally outputting a low-frequency sampling rate signal. Its advantages are: it can achieve downsampling of over a thousand times while maintaining passband flatness, suppressing high-frequency aliasing noise, and ensuring the integrity and accuracy of the downsampled data.

An analog to digital conversion circuit of a touch screen computing device includes a plurality of analog to digital converter circuits operable to convert a plurality of analog signals into a plurality of digital signals at an oversampling rate, a digital decimation filtering module operable to convert the plurality of digital signals into a plurality of digital filtered signals at a first output rate, a coefficient processor operable to generate real component coefficients and imaginary component coefficients of a filtering function at a plurality of frequencies, a first bandpass filter circuit operable to produce first affect values at known frequencies of the plurality of frequencies, and a second bandpass filter circuit operable to produce second affect values at selected frequencies of the plurality of frequencies.

A streaming engine employed in a digital data processor specifies a fixed read only data stream defined by plural nested loops. An address generator produces address of data elements. A steam head register stores data elements next to be supplied to operational units for use as operands. A promotion unit optionally increases date element data size by an integral power of 2 either zero filing or sign filling the additional bits. A decimation unit optionally decimates data elements by an integral factor of 2. For ease of implementation the promotion factor must be greater than or equal to the decimation factor.

The invention discloses a noise suppression method for time-frequency electromagnetic detection data. Corresponding signal processing work is carried out on analog signals through a front-end signal conditioning circuit and digital signals obtained after AD sampling on an FPGA platform so as to improve the signal-to-noise ratio of input signals. And in a time domain electromagnetic detection mode, denoising processing is carried out on the transient sampling signal by using a superposition average algorithm. In a frequency domain electromagnetic detection mode, a harmonic sampling signal is subjected to digital orthogonal locking amplification algorithm processing, frequency mixing is carried out on the harmonic sampling signal, and the frequency spectrum of the harmonic sampling signal is shifted to be near zero frequency; low-pass filtering is realized through a narrowband filtering extraction module, out-of-band noise is suppressed, and the signal-to-noise ratio of the signal is improved; and further improving the signal-to-noise ratio by using an FFT algorithm, and obtaining the amplitude and phase of the target signal through a CORDIC algorithm.

The application discloses a high-speed sampling data real-time extraction method based on parallel digital filters, which comprises the following steps: collecting a measured signal through a collection system of a mixed domain oscilloscope to obtain parallel multi-channel signals; realizing large-multiple real-time extraction filtering through a parallel CIC filter and a cascaded parallel half-band filter to obtain serial data; and finally performing serial extraction on the serial data to obtain output data, so that the frequency resolution of the spectrum analysis function of the oscilloscope can be significantly improved without increasing the FFT length.

Embodiments of the present disclosure disclose processing of Narrow Band-Internet of Things (NB-IoT) Physical Resource Block (PRB). The method comprises aligning center of Physical Resource Block (PRB) in received time domain samples of Long-Term Evolution (LTE) channel bandwidth to center of LTE channel bandwidth; upon aligning center of PRB with center of LTE channel bandwidth, performing decimation on received time domain samples at predefined number of decimation stages sequentially to obtain decimated time domain samples of predefined sample rate; performing Fast Fourier Transform (FFT) operation on decimated time domain samples using FFT of predefined point to obtain corresponding frequency domain samples related to decimated time domain samples; and extracting valid tones from frequency domain samples based on valid tone indices to obtain modified PRB. The modified PRB is transmitted to processing unit for decoding data from modified PRB.

A method of encoding a dynamic mesh includes creating a base mesh through mesh decimation, subdividing the base mesh, extracting displacement information for the subdivided mesh, and encoding the base mesh and the displacement information. In this instance, mesh subdivision information for subdivision of the base mesh is encoded and signaled.

A method and system for bandwidth extension of a critical sampling circuit, belonging to the field of signal sampling technology. The implementation method is as follows: S1, estimating the model coefficients of the squared amplitude response of the simulated channel of the critical sampling circuit through ultra-relaxed primordial duality, constructing the simulated channel response matrix, and obtaining the system response matrix by row decimation; S2, establishing the high-dimensional response matrix of the critical sampling circuit and completing parameter initialization; S3, performing iterative calculation of the extension features through elastic network regression, checking the convergence of the extension residual or the number of iterations, if satisfied, proceeding to S4, otherwise continuing iteration; S4, reconstructing the high-dimensional signal using the extension features calculated in S3, and obtaining the reconstructed signal after bandwidth extension of the critical sampling circuit through sampling rate conversion. This invention, by iteratively calculating the spectral characteristics within the target extended bandwidth of the critical sampling circuit, can recover out-of-band lost spectral information without increasing the sampling clock frequency and analog input bandwidth, achieving low-cost high-frequency signal sampling.

In one implementation, we propose to change the decoding process of blocks within V-PCC patches by directly decoding blocks at reduced resolutions, allowing a less computationally intensive rendering of the point cloud. In one example, based on the rendering information from the renderer, we evaluate for each patch, for each block component of the patches, or for each pixel of the patches, if the corresponding reconstructed point will be viewed by the user after the rendering process and which decimation level could be selected without impairing significantly the rendering quality. Based on the evaluation results, we decide whether to render or not the current area and at which resolution. In one approach, we compare the distance to the screen to a ranges of distance thresholds. In another approach, we compare the distance in the screen coordinate of the projection of two adjacent points of the block.

The invention relates to a method for determining frequency-dependent winding impedance, wherein, in succession for different first frequencies fn, a sinusoidal voltage is applied to at least one of the motor phases, the frequency of the sinusoidal voltage being the particular first frequency fn, wherein the values ​​of the motor current detected by a current sensor are converted by a sigma-delta modulator into a digital data stream which: - is supplied to a first decimation filter; - for a plurality of natural numbers k, is multiplied by a time-dependent sinusoidal signal in each case, the frequency of the sinusoidal signal being the particular first frequency fn multiplied by k in each case, and a moving average of the product thus created is formed to determine each first spectral component; and - for the natural numbers k, is multiplied by a time-dependent cosinusoidal signal in each case, the frequency of the cosinusoidal signal being the particular first frequency fn multiplied by k in each case, and a moving average of the product thus created is formed to determine each second spectral component, wherein the winding impedance Z is determined from the spectral components of the current and the voltage for each of the first frequencies and their k-multiplied multiples.

This video encoding device includes: a decimation means for decimating pictures in a video; an encoding means for encoding a video signal resulting from the decimation of the pictures; and an auxiliary information generation means for generating auxiliary information for deriving a maximum latency picture quantity.

The invention relates to the technical field of radio frequency signal processing, in particular to a broadband radio frequency signal processing method and system based on multi-stage frequency conversion, and the method comprises the following steps: S1, inputting a radio frequency signal, mixing the radio frequency signal into a first intermediate frequency signal through a first-stage mixer, and filtering the first intermediate frequency signal through a first filter and a second filter to obtain a second intermediate frequency signal; frequency mixing is performed through a second-stage frequency mixer to obtain a second intermediate frequency signal; s2, sampling the second intermediate frequency signal to obtain a modulation signal, and extracting an in-phase component and an orthogonal component of the modulation signal; s3, performing frequency mixing on the in-phase component and the orthogonal component of the modulation signal and an orthogonal signal, and performing multi-stage decimation filtering through a filter to obtain an orthogonal baseband signal; and S4, calculating an instantaneous amplitude, an instantaneous phase and an instantaneous frequency of the orthogonal baseband signal, and completing frequency modulation signal demodulation through a phase difference algorithm to obtain a modulation signal source. The method provided by the invention has the advantages of wide frequency band coverage, high signal quality and high demodulation efficiency.

The present application discloses a variable decimation rate digital filter and its test platform. The filter includes an input sample receiving module that receives input signal sample data; a buffer update module that provides a buffer area to store historical values of the signal sample data and updates them in a timely manner; a kernel function construction module that sets a cutoff frequency and determines different types of kernel functions through a sliding window based on a sinc function and the cutoff frequency. The different types of kernel functions serve as weight coefficients of the corresponding filters; a variable tap design module that sets a decimation rate to adjust the attenuation factor of the filter by adjusting the decimation rate, thereby adjusting different frequency responses; a filter calculation module that initializes the output variable value of the filter and traverses the buffer update module to calculate the output variable value of the current filter according to the decimation rate and weight coefficient based on the signal sample data, and returns the current output variable value. The filter attenuation factor can be adjusted to adapt to different application scenarios and signal processing requirements.