Provided is a digital filter circuit in which a filter coefficient can be easily changed, for which circuit scale and power consumption can be reduced, and which carries out digital filter processing in a frequency domain. This digital filter circuit includes: a separating circuit for separating a first complex number signal, of a frequency domain that was subjected to Fourier transform, into a real number portion and an imaginary number portion; a filter coefficient generating circuit for generating a first frequency domain filter coefficient from a first input filter coefficient and a third input filter coefficient, and for generating a second frequency domain filter coefficient from a second input filter coefficient and the third input filter coefficient; a first filter that filters the separated real number portion using the first frequency domain filter coefficient; a second filter that filters the separated imaginary number portion using the second frequency domain filter coefficient; and a combining circuit for combining the output from the two filters.

Technologies are disclosed for improving the efficiency of real-time audio processing, and specifically for improving the efficiency of continuously modifying a real-time audio signal. Efficiency is improved by reducing memory bandwidth requirements and by reducing the amount of processing used to modify the real-time audio signal. In some configurations, memory bandwidth requirements are reduced by selectively transferring active samples in the frequency domain—e.g. avoiding the transfer samples with amplitudes of zero or near-zero. This has particular importance when the specialized hardware retrieves samples from main memory in real-time. In some configurations, the amount of processing needed to modify the audio signal is reduced by omitting operations that do not meaningfully affect the output audio signal. For example, a multiplication of samples may be avoided when at least one of the samples has an amplitude of zero or near-zero.

Utilizing a fast Fourier transform (FFT) to cancel a non-liner phase response of a digital infinite impulse response (IIR) lowpass filter is presented herein. An apparatus generates, via the digital IIR lowpass filter, respective discrete time domain orthogonal frequency-division multiplexing (OFDM) based digital output values comprising non-linear phase distortion; in response to removing respective cyclic prefix values from the respective discrete time domain OFDM based digital output values to obtain a group of discrete time domain OFDM based digital output values, generates, based on such values via a digital FFT, respective frequency domain OFDM based digital output values comprising a non-linear phase response of the digital FFT; and based on the non-linear phase response of the digital IIR lowpass filter, applies phase compensation to the respective frequency domain OFDM based digital output values to obtain frequency compensated frequency domain OFDM based digital output values comprising a linear phase response.

Technologies are disclosed for improving the efficiency of real-time audio processing, and specifically for improving the efficiency of continuously modifying a real-time audio signal. Efficiency is improved by reducing memory bandwidth requirements and by reducing the amount of processing used to modify the real-time audio signal. In some configurations, memory bandwidth requirements are reduced by selectively transferring active samples in the frequency domain—e.g. avoiding the transfer samples with amplitudes of zero or near-zero. This has particular importance when the specialized hardware retrieves samples from main memory in real-time. In some configurations, the amount of processing needed to modify the audio signal is reduced by omitting operations that do not meaningfully affect the output audio signal. For example, a multiplication of samples may be avoided when at least one of the samples has an amplitude of zero or near-zero.

An analysis filter bank corresponding to a plurality of sub-bands, comprising: multiple sub-filters with different center frequencies which perform multiple complex-type first-order infinite impulse response filtering operations on an audio input signal to generate multiple sub-filter signals; a first set of binomial combiners, each of which performs a weighted-sum operation on a first number of the sub-filter signals with a first set of binomial weights to generate one of multiple sub-band signals; a second set of binomial combiners, each of which performs a weighted-sum operation on a second number of the sub-filter signals with a second set of binomial weights to generate one of multiple lower sub-band-edge signals or one of multiple higher sub-band-edge signals; and multiple envelope detection with decimation devices, which perform multiple envelope detection with decimation operations on the sub-band signals, the lower sub-band-edge signals, and the higher sub-band-edge signals to generate multiple fine spectrums.

Determining a thickness of a layer on a wafer during a semiconductor process may include executing the process on the layer on the wafer; monitoring the wafer during the process with an in-situ spectrographic monitoring system to generate spectral data reflected from the wafer; applying a bandpass filter operation to the spectral data to generate filtered spectral data, where the bandpass filter may be configured to pass a frequency range corresponding to the layer on the wafer; and matching the filtered spectral data to a reference filtered spectral data, where the reference filtered spectral data may have been filtered using the bandpass filter operation, and the reference filtered spectral data may be associated with a thickness of the layer.

The invention relates to a system for detecting an input and controlling at least one downstream device, wherein the system comprises at least one sensor device and an evaluation and control device connected to the sensor device for signalling purposes, wherein the at least one sensor device detects an input signal which varies at least over time in the form of a movement, wherein the evaluation and control device evaluates at least the time profile of the detected input signal, wherein at least one evaluation condition is specified, and wherein the downstream device is activated when the at least one evaluation condition is met.

The present disclosure relates to a digital filter arrangement (DFA) for compensating group velocity dispersion (GVD) in an optical transmission system (OTS) wherein the DFA is configured to receive a sequence of samples of a digital input signal in the time domain in the form of consecutive blocks of size L. The DFA is configured to generate M discrete Fourier transforms of a current overlap block of a size N greater than the size L and of M−1 delayed versions of the current overlap block. The DFA is configured to filter the entries of the generated M discrete Fourier transforms to generate an output discrete Fourier transform with N entries, wherein the compensation filter is implemented by a delay network and a linear combination algorithm.

An electric digital received signal obtained from a received optical signal is segmented into blocks of a certain length with an overlap of a length determined in advance with an adjacent block. Fourier transformation is performed for each of the blocks. The blocks subjected to the Fourier transformation are stored consecutively in time series, a coefficient determined based on a wavelength dispersion compensation amount according to one of frequency positions and a delay amount according to one of the frequency positions and one of time positions is applied to each of frequency component values included in a plurality of the stored blocks, and the blocks to which the coefficient has been applied and which are obtained by adding up the frequency component values to which the coefficient has been applied for each of the frequency positions are generated. Inverse Fourier transformation is performed on the generated blocks to which the coefficient has been applied. A part of the overlap subjected to the inverse Fourier transformation is removed.

A method for signal processing, receiving a time domain signal having a sample-rate Fs and generating N time domain signal bands, each having a bandwidth equal to Fs/N. Receiving the N signal bands and transforming a first time domain signal band to a frequency domain at a first resolution and a second time domain signal band to the frequency domain at a second resolution, where the first resolution may be different from the second resolution. Determining one or more first filter coefficients using the frequency domain components from the first signal band and one or more second filter coefficients using the frequency domain components from the second signal band. Transforming the first and second filter coefficients from the frequency domain to a time domain. Applying the first and second time domain filter coefficients to the first and second time domain signals, respectively.