H03H17/0201

DIGITAL FILTER, FILTER PROCESSING METHOD, AND RECORDING MEDIUM
20200304107 · 2020-09-24 · ·

The present invention addresses the problem of increasing the likelihood of making it possible to reduce the consumption of power necessary for filter processing and the amount of heat generated during filter processing. In order to overcome this problem, a second complex signal and a third complex signal are generated from a first complex signal in a frequency domain, the third complex signal being a complex conjugate of the second complex signal. Signal selection is performed from the plurality of types of complex signals having different amounts of change in signal amplitude. Processing is performed on the complex signal selected as the signal using a first filter coefficient and a second filter coefficient. The complex signals after filter processing are synthesized to generate a complex signal, which is then outputted.

GENERATING A REPRESENTATION OF HIGH-FREQUENCY ELECTRIC POWER DELIVERY SYSTEM DATA USING DEVIATIONS FROM A TREND

A system, method, and computer program product are provided for representation of high-frequency signal data. In use, input data is received including high-frequency signals, wherein the input data is of a first width. Next, the input data is processed to manage display of the input data, where specifically the input data is divided into one or more segments based on first criteria including the first width, and from each segment of the one or more segments, a maximum value is identified and a minimum value is identified. The maximum and minimum may be trend maximum and minimum values. The input data is transformed to a visualizable representation of the high-frequency signals, the visualizable representation of the high-frequency signals including a plot of the maximum value and the minimum value for each segment of the one or more segments. Additionally, the plot is displayed.

RESOURCE CONSERVING WEIGHTED OVERLAP-ADD CHANNELIZER
20200274524 · 2020-08-27 · ·

Systems and methods are provided for channelizing. A first stage can provide a WOLA filter bank that can apply a single multiplier resource to perform window weighting for multiple WOLA filter banks. The first stage can remove mixer-based post FFT adjustment and provide equal functionality with a particular modification of tuning mixers at inputs of second stage FIR paths. The first stage can include a variable decimation, using a particular implementation of variable sample block size.

Multichannel, multirate, lattice wave filter systems and methods

Systems and methods for multichannel, multirate lattice wave filters receive digital signal channels at a first sample rate and include a first multiplexer to combine the digital signal channels into a first digital data stream, and a first lattice wave filter comprising a first delay elements and a first feedback path to the first multiplexer, the first lattice wave filter produces a first output digital data stream having a second sample rate that is different than the first sample rate. The first multiplexer is configured to receive a first feedback signal through the first feedback path and combine the first feedback signal with the digital signal channels to produce the first digital data stream. The system may include a first processing branch comprising the first multiplexer and the first lattice wave filter structure, and a second processing branch comprising a second multiplexer and a second lattice wave filter structure. This may enable the implementation of simplified filters of lower complexity by reuse of hardware.

Digital Filterbank for Spectral Envelope Adjustment
20200066292 · 2020-02-27 · ·

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.

Modified digital filtering with sample zoning
10536136 · 2020-01-14 ·

The present invention relates broadly to a method of digitally filtering a signal, such as an audio signal, using a digital filter. The digital filter includes a plurality of neighbouring sample points broken into zones having different frequency content or frequency ranges. The zones adjacent one another may have neighbouring sample points in common. Generally each zone has at least same distinct frequencies compared with other zones. That is, the zones are roughly dependent on the frequency content. The invention in its preferred form involves combining values for two or more of the neighbouring sample points for select of the zones depending on its frequency content. The values are combined so as to provide a modified zone having substantially the same number of sample points as the select zone. The modified zones together provide a modified filter to be applied to the signal.

DIGITAL FILTERBANK FOR SPECTRAL ENVELOPE ADJUSTMENT
20240055010 · 2024-02-15 · ·

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.

METHOD AND SYSTEM FOR TRACKING SINUSOIDAL WAVE PARAMETERS FROM A RECEIVED SIGNAL THAT INCLUDES NOISE
20190294649 · 2019-09-26 ·

A system for tracking selected wave parameters from a received sinusoidal wave with noise and methods for making and using the same. The method includes performing a multi-track double integral analysis of the sinusoidal wave with noise and creating time dependent outputs. These time dependent outputs may be analyzed mathematically to determine the amplitude, frequency and/or phase of the wave with reduced noise. In one embodiment, the method may employ multiple passes through double integral analysis. The method advantageously can measure output sinusoidal wave parameters with reduced noise, measurements that are close to theoretical noise reduction limits.

Method and system for signal decomposition, analysis and reconstruction
10367476 · 2019-07-30 · ·

A system and method for representing quasi-periodic waveforms, for example, representing a plurality of limited decompositions of the quasi-periodic waveform. Each decomposition includes a first and second amplitude value and at least one time value. In some embodiments, each of the decompositions is phase adjusted such that the arithmetic sum of the plurality of limited decompositions reconstructs the quasi-periodic waveform. Data-structure attributes are created and used to reconstruct the quasi-periodic waveform. Features of the quasi-periodic wave are tracked using pattern-recognition techniques. The fundamental rate of the signal (e.g., heartbeat) can vary widely, for example by a factor of 2-3 or more from the lowest to highest frequency. To get quarter-phase representations of a component (e.g., lowest frequency rate component) that varies over time (by a factor of two to three) many overlapping filters use bandpass and overlap parameters that allow tracking the component's frequency version on changing quarter-phase basis.

Digital filter and vehicle driving force control apparatus
10308250 · 2019-06-04 · ·

To suppress an unstable operation of a device caused by filtering, an output selection unit is configured to acquire an unfiltered value representing an input signal (signal before filtering) stored in an input holding unit, a filtered value representing a signal acquired by filtering the input signal by a filtering unit, and a previous output value representing an output signal output at a previous time, select a middle value out of the filtered value, the unfiltered value, and the previous output value, and set the selected middle value as a current output value of an output signal.