Patent classifications
H03H17/02
Automatic gain control system for processing of clipped signal samples
Techniques are provided for automatic gain control processing to reduce adverse effects associated with clipped samples resulting from conversion of analog signals to digital signals. A methodology according to an embodiment includes identifying a clipped sample of the digital signal, for example by comparison of the digitized sample values to a threshold value associated with a full scale value of the converter. The method also includes applying a window function to portions of the digital signal. The window function is configured to attenuate samples of the digital signal within a region centered on the identified clipped sample. A Hilbert finite impulse response (FIR) filter may be applied to the digital signal prior to applying the window function. Parameters of the window function are selected based on frequency response characteristics of the FIR filter and on signal to noise ratio requirements of an application that receives the windowed digital signal.
Automatic gain control system for processing of clipped signal samples
Techniques are provided for automatic gain control processing to reduce adverse effects associated with clipped samples resulting from conversion of analog signals to digital signals. A methodology according to an embodiment includes identifying a clipped sample of the digital signal, for example by comparison of the digitized sample values to a threshold value associated with a full scale value of the converter. The method also includes applying a window function to portions of the digital signal. The window function is configured to attenuate samples of the digital signal within a region centered on the identified clipped sample. A Hilbert finite impulse response (FIR) filter may be applied to the digital signal prior to applying the window function. Parameters of the window function are selected based on frequency response characteristics of the FIR filter and on signal to noise ratio requirements of an application that receives the windowed digital signal.
ACTUALLY-MEASURED MARINE ENVIRONMENT DATA ASSIMILATION METHOD BASED ON SEQUENCE RECURSIVE FILTERING THREE-DIMENSIONAL VARIATION
The present invention provides an actually-measured marine environment data assimilation method based on sequence recursive filtering three-dimensional variation. The method includes: preprocessing actually-measured marine environment data; calculating a target function value; calculating a gradient value of a target function; calculating a minimum value of the target function; extracting space multi-scale information from the actually-measured data; and updating background field data to form a final data assimilation analysis field. The present invention improves the traditional recursive filtering three-dimensional variation method, and sequentially assimilates information with different scales, thereby effectively overcoming the problem that multi-scale information cannot be effectively extracted by a traditional three-dimensional variation method. A high-order recursive Gaussian filter is used, and a cascaded form of the high-order recursive filter is converted into a parallel structure, so that the recursive filtering process of the recursive Gaussian filter can be executed in parallel, and many problems caused by a cascaded filter are overcome.
Filter for a Brushless DC Motor
A filter for use with a brushless DC motor to filter a signal received from a floating terminal of the brushless DC motor, wherein the filter is configured such that a time delay introduced by the filter to the signal received from the floating terminal is equal to the time taken for a rotor of the motor to rotate through an angle equal to half of a commutation step of the motor.
DIGITAL CONTROLLER FOR A MEMS GYROSCOPE
A digital control circuitry for a MEMS gyroscope is provided. The digital control circuitry comprises a digital primary loop circuitry configured to process a digitized primary signal, a digital secondary loop circuitry configured to process a digitized secondary signal and a digital phase shifting filter circuitry configured to generate two phase shifted demodulation signals from the digitized primary signal. The digital secondary loop is configured to demodulate the digitized secondary signal using the two phase shifted demodulation signals.
DEVICE AND METHOD OF MEDIAN FILTERING
A median filter device is provided with a reordered circuit, a comparison circuit and a data refresh circuit on the basis of the conventional data buffer circuit and data register circuit. The reorder circuit re-sorts the signal data stored in the data buffer circuit in a preceding clock cycle according to their numerical values. The comparison circuit compares the new signal datum entered in the current clock cycle with the signal data already stored to generate a median. The data refresh circuit updates the signal codes stored in the data register circuit with the signal codes corresponding to the new signal data, for calculation of the median in a following clock cycle. The length of the data buffer circuit and data register circuit can be reduced from N signal data to N-1 signal data, which achieves less data storage capacity, smaller circuit area, easier data processing and higher operation efficiency.
DEVICE AND METHOD OF MEDIAN FILTERING
A median filter device is provided with a reordered circuit, a comparison circuit and a data refresh circuit on the basis of the conventional data buffer circuit and data register circuit. The reorder circuit re-sorts the signal data stored in the data buffer circuit in a preceding clock cycle according to their numerical values. The comparison circuit compares the new signal datum entered in the current clock cycle with the signal data already stored to generate a median. The data refresh circuit updates the signal codes stored in the data register circuit with the signal codes corresponding to the new signal data, for calculation of the median in a following clock cycle. The length of the data buffer circuit and data register circuit can be reduced from N signal data to N-1 signal data, which achieves less data storage capacity, smaller circuit area, easier data processing and higher operation efficiency.
Digital Filterbank for Spectral Envelope Adjustment
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.
Signal Processing Method and Apparatus
Embodiments of the present invention provide a signal processing method and apparatus. The method includes: performing M-way filtering on an input signal to obtain M filtered signals, performing extraction on M filtered signals separately to obtain M extracted signals, performing fast Fourier transform (FFT) on the M extracted signals separately to obtain M frequency-domain signals, and finally determining output signals according to the M frequency-domain signals. According to the embodiments of the present invention, signal filtering and extraction are performed and then FFT is performed.
Phase aligned interleaved sampling of multiple data channels
Provided is a method for processing data samples from a plurality of data channels. The method may include obtaining a plurality of data samples from the plurality of data channels. Obtaining the plurality of data samples may involve successively obtaining a data sample from each data channel of the plurality of data channels. Successively obtaining a data sample from each data channel may be performed a plurality of times during a specified time period. Each data sample of the plurality of data samples may be associated with a respective sample time, and each respective sample time may be relative to a single specified reference point in time. The method may further include, for each data sample of the plurality of data samples, determining a time-dependent coefficient value that may correspond to the sample time associated with the data sample, and applying the determined time-dependent coefficient value to the data sample.