Provided are a filter unit and a filter array, and the filter unit includes: a control module (10) configured to analyze configuration information of the filter unit, and control selection of internal functions and flow direction of data in the filter unit; a filtering module (20) connected to the control module (10), and configured to perform an operation on internal data of the filter unit and perform a sampling rate conversion function under control of the control module (10); and an output selection module (30) connected to the control module (10) and the filtering module (20), and configured to perform a selection function on output data of the filtering module (20) under control of the control module.

Filters are discussed where a first window function and a second window function are applied to a digital input signal, wherein a window length of the first window function is longer than a window length of the second window function. The results of this windowing are integrated.

A crest factor reduction (CRF) circuit may include a scaler configured to receive the input signal and generate a scaled input signal. A clipping circuit may be configured to receive the input signal and generate a clipped input signal. A negator circuit may be configured to receive the clipped input signal and generate a negated clipped input signal. A first summer may be configured to sum the scaled input signal and the negated clipped input signal to generate a summed signal. A first digital filter may be configured to receive the summed signal and provide a first digital filter output. A second digital filter may be configured to receive the clipped input signal and provide a second digital filter output. A multiplexer may be configured to receive the first digital filter output and the second digital filter output and generate an output signal.

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.

Filters are discussed where a first window function and a second window function are applied to a digital input signal, wherein a window length of the first window function is longer than a window length of the second window function. The results of this windowing are integrated.

A digital filter includes integrator circuits configured to operate based on a clock of a sampling frequency f.sub.S that is equal to a sampling frequency of input data and determine a sum of the input data on a sample-by-sample basis, a frequency converter circuit configured to perform decimation on data of the sampling frequency f.sub.S to reduce the sampling frequency f.sub.S to a sampling frequency f.sub.D=f.sub.S/N, one or more differentiator circuits configured to operate based on a clock of the sampling frequency f.sub.D and subtract data of an immediately preceding sample from the input data, a differentiator circuit for removal of 50 Hz configured to operate based on the clock of the sampling frequency f.sub.D and subtract, from the input data, data preceding the input data by a plurality of samples, and a differentiator circuit for removal of 60 Hz configured to operate based on a clock of the sampling frequency f.sub.D and subtract, from the input data, data preceding the input data by a plurality of samples.

The present invention relates broadly to a method of digitally filtering an audio signal by applying a composite audio filter. The composite audio filter may be obtained by applying one audio filter to another audio filter each having the same predetermined sample rate including neighboring sample points. The other audio filter may also include one or more intervening sample points between adjacent of its neighboring sample points. The one audio filter may be applied to the other audio filter at an adjusted sampling rate relative to the other audio filter. The adjusted sampling rate may be inversely proportional to the number of intervening sample points relative to the number of neighboring sample points for the other filter. The frequency response curve for the composite filter derived using the adjusted sampling rate may be more indicative of an idealized lowpass filter. The frequency response with the adjusted sampling rate may display a more bell-shaped characteristic compared with the frequency response without an adjusted sampling rate (shown in broken line detail).

The present invention relates broadly to a method of digitally filtering an audio signal at a predetermined sample rate by applying a composite audio filter derived at an increase sample rate. The composite audio filter is obtained by combining one audio filter with another audio filter at the increased sample rate. The sample rate of the audio filters may be increased from their predetermined to the increased sample rate using various weighting techniques. The composite filter may provide a frequency response curve with a corner frequency as it approaches the Nyquist frequency whereas the frequency response of a conventional filter (shown in broken line detail) is flat with no effect.

A signal processing circuit has a first signal loop with a first signal processing block and a first feedback path that extends around the first signal processing block, the first signal processing block having a frequency dependence that causes the first signal loop to generate a passband. A second signal processing block is downstream of the first signal loop. A second feedback path extends from downstream of the second signal processing block to upstream of the first signal processing block. In operation, the first feedback path reinforces a signal in the passband and the second feedback path conditions the signal at an output downstream of the first signal processing block.

A digital filter includes integrator circuits configured to operate based on a clock of a sampling frequency f.sub.S that is equal to a sampling frequency of input data and determine a sum of the input data on a sample-by-sample basis, a frequency converter circuit configured to perform decimation on data of the sampling frequency f.sub.S to reduce the sampling frequency f.sub.S to a sampling frequency f.sub.D=f.sub.S/N, one or more differentiator circuits configured to operate based on a clock of the sampling frequency f.sub.D and subtract data of an immediately preceding sample from the input data, a differentiator circuit for removal of 50 Hz configured to operate based on the clock of the sampling frequency f.sub.D and subtract, from the input data, data preceding the input data by a plurality of samples, and a differentiator circuit for removal of 60 Hz configured to operate based on a clock of the sampling frequency f.sub.D and subtract, from the input data, data preceding the input data by a plurality of samples.