摘要:
A sampling rate converting apparatus that performs sampling frequency conversion includes: a sampling-phase detecting unit (1) configured to calculate a sampling phase based on sampling timing after the sampling frequency conversion for each operation reference clock, based on a ratio of a frequency of an operation reference clock and a sampling frequency after the sampling frequency conversion; a sampling-clock detecting unit (2) configured to detect a sampling period based on the sampling phase; a poliphase filter (3) configured to apply filtering to an input signal and generate a signal after the sampling frequency conversion, based on the sampling period; and a filter-coefficient control unit (4) configured to set a filter coefficient in the poliphase filter (3), based on the sampling phase.
摘要:
The invention relates to a filter device (200) for filtering an input signal (s(t)). The filter device (200) has a plurality of taps (210-216) having a respective filter coefficient (w0-w6) and a plurality of delay elements (221-226), wherein at least two delay elements (221-226) have different delays.
摘要:
An audio signal output device for outputting on the basis of an digital audio signal. A decision unit decides the inversion frequency of the polarities of the digital audio signal. A select unit switches the output of the audio signal output device in accordance with the decision result by the decision unit between an output based on a first interpolated digital audio signal interpolated by a first interpolation of the digital audio signal and an output based on a second interpolated digital audio signal interpolated by a second interpolation of the digital audio signal.
摘要:
The present invention relates to an audio signal output device for outputting based on a digital audio signal. A judging unit judges the frequency of reversals in polarity of the digital audio signal. A selecting unit switches an output of the audio signal output device between an output based on a first interpolated digital audio signal obtained by interpolating the digital audio signal by way of a first interpolation processing and another output based on a second interpolated digital audio signal obtained by interpolating the digital audio signal by way of a second interpolation processing in response to a judgment result made by the judging unit.
摘要:
A linearity corrector is provided that reduces distortion in a signal processing system, such as an ADC. The linearity corrector provides a first order signal path having distortion components connected to an adder, and a filter product circuit that is also connected to the adder. A method is provided for reducing distortion by calculating a filter product and adding the filter product to a first order signal having a relative delay such that the filter product reduces, or eliminates, the order of distortions corresponding to the order of the filter product.
摘要:
A digital signal processing method for further improving the reproducibility of the waveform of a digital signal, a learning method, apparatuses for them, and a program storage medium are disclosed. Segments are cut out from a digital signal (D10) by means of widows different in size to calculate their autocorrelation coefficients (D40, D41). The class is determined on the basis of the result of calculation (D15) of the autocorrelation coefficients (D40, D41). The digital signal (D10) is converted by a prediction method corresponding to the class. Therefore, conversion further adapted to the feature of the digital signal (D10) is carried out.
摘要:
A sampling rate converter able to obtain an amplitude characteristic that passes any frequency and able to achieve a high precision conversion without depending upon a cutoff frequency, having an up sampler 103 for inserting (U-1) zero points between signals and raising a sampling frequency Fsi U-fold, a convolution processing unit 104 including an FIR filter and interpolating a value by convolution with respect to output signals of the up sampler, and a linear interpolation block 105 for selecting two points of samples from the output signal of the convolution processing unit 104 having a sampling frequency UFsi and finding the value at a required position from the linear interpolation, wherein the FIR filter has an impulse response becoming a filter coefficient, having a transmission function H(z) associated with a transmission function Z(z) of a pre-filter, and having a filter coefficient set by performing weighted approximation with respect to a desired characteristic associated with the frequency response of the pre-filter.