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.

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 invention relates to a signal processing device, a control method thereof, a control procedure and a recording medium, which is unnecessary to increase a memory capacity for implementing sampling to determine a filter parameter. The signal processing device includes a data acquiring element (10), acquiring signals of a time series, i.e. time series signals from a sensor; a filtering element (21), performing a filtering operation according to frequencies; a transfer element (30), transferring the time series signals; and a filter parameter determination element (50), performing frequency analysis to the time series signals within a pre-specified interval, i.e. a specified interval to determine a filter parameter.

A finite impulse response (FIR) filter including a delay line and a plurality of arithmetic units. Each arithmetic unit is coupled to a different one of a plurality of tap points of the delay line, is configured to receive a respective signal value over the delay line, and is associated with a respective coefficient. Any given one of the arithmetic units is configured to receive a respective control word. The respective control word specifying: (i) a plurality of trivial multiplication operations, and (ii) a plurality of bit shift operations. Any given one of the arithmetic units is further configured to estimate or calculate a product of the respective signal of the arithmetic unit respective signal value and the respective coefficient of the arithmetic unit by performing the trivial multiplication operations and bit shift operations that are specified by the respective control word that is received at the given arithmetic unit.

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.

A finite impulse response (FIR) filter including a delay line and a plurality of arithmetic units. Each arithmetic unit is coupled to a different one of a plurality of tap points of the delay line, is configured to receive a respective signal value over the delay line, and is associated with a respective coefficient. Any given one of the arithmetic units is configured to receive a respective control word. The respective control word specifying: (i) a plurality of trivial multiplication operations, and (ii) a plurality of bit shift operations. Any given one of the arithmetic units is further configured to estimate or calculate a product of the respective signal of the arithmetic unit respective signal value and the respective coefficient of the arithmetic unit by performing the trivial multiplication operations and bit shift operations that are specified by the respective control word that is received at the given arithmetic unit.

In a control method of controlling a frequency response of a filter for processing an audio signal, the frequency response is defined by an amplitude characteristic and a phase characteristic. The control method includes: acquiring a target amplitude characteristic, a target phase characteristic, and a latency value; obtaining a first frequency response in accordance with the target amplitude characteristic and the target phase characteristic; obtaining a second frequency response by modifying the first frequency response so that a latency of the second frequency response satisfies the latency value; and obtaining a third frequency response to be set for the filter by correcting the second frequency response, the second frequency response being corrected so as to reduce a difference between (i) an amplitude characteristic of the second frequency response and (ii) the target amplitude characteristic.

A digital filter, an audio signal processing system, and a method of designing a digital filter are provided, where the digital filter reduces a DC component of a signal which corresponds to an input data sequence, and has a transfer function H(s) represented as

[00001]$H\left(s\right)=\frac{s^2+\frac{{\omega }_0}{Q}s}{s^2+\frac{{\omega }_0}{Q}s+{\omega }_0^2}$

where a cutoff frequency F.sub.0 is ω.sub.0/2π and a Q factor is Q.

A data storage device is disclosed comprising a head actuated over the disk. A first notch filter is calibrated at a first temperature, and a second notch filter is computed for a second temperature based on the first notch filter and a first delta temperature between the first temperature and the second temperature. The second notch filter is computed by transforming a continuous-time transfer function of the first notch filter adjusted by the first delta temperature into a discrete-time transfer function of the second notch filter.

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.