A digital filter includes: integration calculation units (10) that are cascade-connected, are fed time-division-multiplexed data, the time-division-multiplexed data being formed of pieces of data on M channels that are time-division multiplexed, the pieces of data on the respective channels being updated at a rate equal to a sampling frequency f.sub.s, operate in accordance with a clock having a frequency f.sub.s×M, and integrate the time-division-multiplexed data for every M samples; a frequency conversion unit (11) that operates in accordance with a clock having a frequency f.sub.D×M, decimates data at the sampling frequency f.sub.s input from the integration calculation unit (10) in the last stage at a sampling frequency f.sub.D, and delays data obtained as a result of decimation by (M−1) samples; and difference calculation units (12) that operate in accordance with the clock having the frequency f.sub.D×M, are cascade-connected to the output of the frequency conversion unit (11), and each subtract, from data input thereto, data M samples before.

A multi-stage demultiplexing circuit in which a plurality of circuits each combining a selector and a frequency decimation circuit are connected is included. The selector selects one of input signals based on a control signal, and generates a plurality of output signals. The plurality of output signals output from the selector are input to the frequency decimation circuit, and the frequency decimation circuit performs frequency conversion processing, low pass filter processing, and down-sampling processing based on a control signal to generate an output signal. Two or more reception signals are input to the multi-stage demultiplexing circuit, and the multi-stage demultiplexing circuit executes demultiplexing processing based on a control signal so that an output signal that includes an unused band portion is prevented from being output downstream.

A multi-stage demultiplexing circuit in which a plurality of circuits each combining a selector and a frequency decimation circuit are connected is included. The selector selects one of input signals based on a control signal, and generates a plurality of output signals. The plurality of output signals output from the selector are input to the frequency decimation circuit, and the frequency decimation circuit performs frequency conversion processing, low pass filter processing, and down-sampling processing based on a control signal to generate an output signal. Two or more reception signals are input to the multi-stage demultiplexing circuit, and the multi-stage demultiplexing circuit executes demultiplexing processing based on a control signal so that an output signal that includes an unused band portion is prevented from being output downstream.

Method for carrying out a morphing process, wherein an output parameter relating to the output of an audio signal outputted into an interior via an audio output device is changed.

Method for carrying out a morphing process, wherein an output parameter relating to the output of an audio signal outputted into an interior via an audio output device is changed.

An encoder includes a low-pass filter to filter input audio signals. The low-pass filter has fixed filter coefficients. The encoder generates quantized signals based on a difference signal. The encoder includes an adaptive quantizer and a decoder to generate feedback signals. The decoder has an inverse quantizer and a predictor. The predictor has fixed control parameters which are based on a frequency response of the low-pass filter. The predictor may include a finite impulse response filter having fixed filter coefficients. The decoder may include an adaptive noise shaping filter coupled between the low-pass filter and the encoder. The adaptive noise shaping filter flattens signals within a frequency spectrum corresponding to a frequency spectrum of the low-pass filter.

A digital filter for filtering an input signal to form an output signal containing a coefficient multiplier and a moving-average filter. The coefficient multiplier is embodied to multiply values of the input signal by coefficients of the filter to form an intermediate signal. The moving-average filter is embodied to generate the output signal as a moving average of the intermediate signal.

In order to at least approximately determine statistical properties of raw measured values, e.g., measurement noise and/or average errors, without knowledge of a filter applied to raw measured values and only with the aid of a useful signal obtained from the filtering, i.e., in order to make statements regarding measurement conditions by assuming a few frequently encountered boundary conditions, statistical properties of raw measured values are determined from the useful signal, which is composed of a temporal sequence of filter output values. A filter characteristic of the filter is ascertained from the temporal sequence of output values obtained under stable measurement conditions, the ascertained filter characteristic is inverted, raw measured values are reconstructed from the inverse of the filter characteristic and from the useful signal, and the statistical properties are ascertained from the reconstructed raw measured values and/or from the inverse of the filter characteristic and from the useful signal.

A method of performing quadrature mirror filter (QMF) synthesis filtering includes recording new samples corresponding to a current time slot at positions of samples to be discarded in a first array that includes modulated QMF sub-band samples. The method further includes extracting samples from the first array to remove aliasing between adjacent sub-bands, determining filter coefficients corresponding to the extracted samples by using modulo operation, and synthesizing a time domain sample where aliasing is removed by using the extracted samples and the filter coefficients.

A method for processing a bitstream starts by shifting a bitstream of a first sample of a signal into a buffer. The buffer also holds bits of one or more additional bitstreams for one or more additional samples of the signal. Bits of a first half of the buffer are incrementally compared to corresponding bits of a second half of the buffer. Each bit of the first half of the buffer is compared to a corresponding bit of the second half of the buffer. A computation is performed on each bit of the first half of the buffer that is equal to a corresponding bit of the second half of the buffer. The results of the computations are summed to determine an output value for the first sample of the signal.