A method of mass spectrometry is disclosed comprising digitising a signal output from a detector to provide a first digitised signal. A finite impulse response (FIR) filter, a digital filter or an echo cancellation filter is applied to the first digitised signal in order to reduce the effect of baseline perturbations, echoes or ringing effects. Alternatively, an analogue signal output from a detector is passed to one or more first power splitters or dividers, wherein one or more first transmission lines are attached to one or more ports of one more said first power splitters or dividers in order to reduce the effect of baseline perturbations, echoes or ringing effects.

An audio processor (1) includes a first filter coefficient calculator (31) that calculates a first filter coefficient so as to correspond to first gains for respective bands set by a user, a second filter coefficient calculator (32) that if values of third gains for respective bands of the first filter coefficient are greater than an absolute value of a second gain set by the user, calculates a second filter coefficient by limiting the values of the third gains for the respective bands to the amplitude value of the second gain, and a filtering unit (35) that filters an audio signal that has been transformed into a frequency-domain signal, using the second filter coefficient.

An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes n analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting.

The embodiments of the present disclosure provide a method and system for adjusting a passband width of a filter. The method includes determining an initial passband width, controlling a filter according to the initial passband width to filter signals to be processed, and correcting the initial passband width according to a first frequency at which a peak spectrum line corresponding to the filtered signals is located.

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.

A method of mass spectrometry is disclosed comprising digitizing a signal output from a detector to provide a first digitized signal. A finite impulse response (FIR) filter, a digital filter or an echo cancellation filter is applied to the first digitized signal in order to reduce the effect of baseline perturbations, echoes or ringing effects. Alternatively, an analog signal output from a detector is passed to one or more first power splitters or dividers, wherein one or more first transmission lines are attached to one or more ports of one more said first power splitters or dividers in order to reduce the effect of baseline perturbations, echoes or ringing effects.

A method for compensating the bandedge ripple of an analog filter, using a circuit comprising a low pass filter is described. The method comprises receiving, at the analog filter, a plurality of tones of different frequencies from a tone generator, measuring, an amplitude of each tone in the plurality of tones after each tone is processed by the analog filter, storing the measured amplitudes and frequencies in a database, measuring a bandedge ripple by measuring a difference in amplitude between a first tone and a second tone from the plurality of tones, and selecting a low pass filter, from a plurality of low pass filters, based on the measured difference.

Example embodiments disclosed herein relate to filter coefficient updating in time domain filtering. A method of processing an audio signal is disclosed. The method includes obtaining a predetermined number of target gains for a first portion of the audio signal by analyzing the first portion of the audio signal. Each of the target gains is corresponding to a linear subband of the audio signal. The method also includes determining a filter coefficients for time domain filtering the first portion of the audio signal so as to approximate a frequency response given by the target gains. The filter coefficients are determined by iteratively selecting at least one target gain from the target gains and updating the filter coefficient based on the selected at least one target gain. Corresponding system and computer program product for processing an audio signal are also disclosed.

In some embodiments, a multiplier-based programmable filter comprises a pre-scaling circuit, a first multiplier circuit coupled to a first output of the pre-scaling circuit and a second output of the pre-scaling circuit, and a second multiplier circuit coupled to the first output of the pre-scaling circuit and the second output of the pre-scaling circuit. In some embodiments, the multiplier-based programmable filter also comprises a first adder coupled to a first output of the first multiplier circuit and a second output of the first multiplier circuit, a second adder coupled to a first output of the second multiplier circuit and a second output of the second multiplier circuit, first register coupled to an output of the first adder and an input of the second adder, and a second register coupled to an output of the second adder.

An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes n analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting.