Abstract: An original filter is connected to an adjustment filter in the longitudinal way. The original filter has a first filter coefficient of a symmetric numeric string. The adjustment filter has a contact point at a position where the maximum value is acquired in the original filter frequency amplitude characteristic A and has a symmetric second filter coefficient realizing the frequency amplitude characteristic B having the minimum value at the contact point. By executing a convolution calculation of the first filter coefficient and the second filter coefficient, it is possible to design a desired filter coefficient.

Abstract: A direct conversion receiver wherein even when signals are continuously received, the automatic gain control can be implemented in accordance with the signal levels from which DC offset voltages have been removed. The direct conversion receiver comprises a low-noise amplifier 14, a mixer 16, a local oscillator (LO) 20, a lowpass filter (LPF) 23, a baseband amplifier (second amplifier) 24, an analog-to-digital converter (ADC) 26, digital-to-analog converters (DAC) 28, 32, a signal processing section 30, a speaker 34, a DC component extracting filter 100, an average value calculating circuit 200 and a subtractor 210. The average value calculating circuit 200 calculates an average value of the signal levels of the baseband signals. The DC offset voltage extracted by the DC component filter 100 is subtracted from the average value, thereby generating a control voltage, which then controls the gain of the input circuit 10 or low-noise amplifier 14.

Abstract: A digital filter is designed by combining unit filters (L10?, H10?) having a predetermined asymmetric numerical sequence as filter coefficients (H1 to H3). Thus, it is possible to automatically obtain a desired digital filter coefficient only by combining the unit filter. Moreover, a symmetric numerical sequence {?1, 0, 9, 16, 9, 0, ?1}/32 is divided at the center into two parts and one of them is used as the asymmetric filter coefficients (H1 to H3). This reduces the number of taps required for the digital filter designed, eliminates use of a window function, and prevents generation of a discretization error in the filter characteristic obtained.

Abstract: A screen enlargement and reduction device includes: a coefficient calculation unit (2) for obtaining a coefficient corresponding to a pixel value of each pixel of an original image and storing the coefficient in a coefficient RAM (4) for each interpolation position set in accordance with an enlargement and reduction ratio “s”; and a coefficient multiplication unit (6) for obtaining a pixel value of each pixel at the interpolation position by multiplying/adding a pixel value of each pixel of the original image extracted by a matrix decomposition unit (5) in accordance with a second process clock (cs2) generated by a clock generation unit (1) and a coefficient read out from the coefficient memory (4) in accordance with a first process clock (cs1). When actually performing an interpolation calculation, the device can be used only by reading out a necessary coefficient from the coefficient RAM (4).

Abstract: An interpolation function generation circuit is formed by cascade connecting a first FIR filter (10) having a numerical value string composed of a ratio “??, ?, ?, ?, ?, ??” (? is an emphasis coefficient and ? is a fixed value) as a filter coefficient and a second FIR filter (20) having a numerical value string composed of a ratio “1, 3, 5, . . . , m?1, m?1, . . . , 5, 3, 1” when the tap length is an odd number and “1, 3, 5, . . . , n?2, n?1, n?2, . . . , 5, 3, 1” if the tap length is an odd number (m and n are multiples of the oversampling). With only the two FIR filter (10, 20), it is possible to easily realize an interpolation function having a variable emphasis.

Abstract: There are included a three-tap FIR calculating part (2) that multiples data outputted from three taps on a tapped delay line by respective filter factors comprising a ratio value sequence of “?1, m, ?1”; and an n-tap FIR calculating part (3) that multiples data outputted from n taps on a tapped delay line by respective filter factors comprising a predetermined value sequence. Interpolation values can be determined by use of sum-of-products calculations using various factor sequences comprising various values of m and n. The three-tap FIR calculating part (2) is adapted to determine interpolation values by use of the sum-of-products calculations that always use only three values. In this way, the circuit scale can be reduced and further the calculation process can be simplified, thereby achieving a high-rate interpolation process.

Abstract: A waveform of a desired frequency characteristic is input as a numeric value string and is subjected to a reverse FFT so as to obtain a filter coefficient group. Thus, without having any expert knowledge and only by inputting a waveform of a desired frequency characteristic as an image, it is possible to easily design a first FIR filter constituting a tone quality adjustment device. Moreover, by performing a predetermined calculation on the numeric value string input and performing a reverse FFT on the result, it is possible to easily design a second FIR filter having a frequency characteristic symmetric to the first FIR filter with respect to the gain reference value as an axis.

Abstract: A method for designing a digital filter for outputting a signal that is the sum of the products of multiplication of the signals at the taps of delay units (11-16) by the filter factors given by factor units (21-25), several times of the signals, wherein various filters from a low-pass filter to a high-pass filter can be designed by using, as the filter factors, the terms of a symmetrical sequence, e.g., {?1, 0, 9, 16, 9, 0, ?1} in which the sum of all the terms is not zero, and the sum of every other terms is equal to the sum of the other every other terms and has the same sign of that of the other every other terms and by simply changing the signs of the terms of the sequence. By applying a combination of cascade connection of filters, conversion of clock rate, and transfer of filter factors, a digital filter having desired frequency characteristics can be extremely simply designed.

Abstract: A range obtained by adding virtual pixels for “k” clocks on an outer side of effective pixels of an original image is set as a processing range. A representative value is calculated for each pixel block including 2×2 pixels adjacent to one another in the processing range. Pixel values of the respective pixels in an interpolation position corresponding to a predetermined scaling factor are calculated by applying an interpolation operation to the representative value calculated such that an effective range 22 of an image formed by the representative value is shifted further to the outer side than the effective range 21 of the original image. The interpolation operation is applied to the representative value, the effective range of which is shifted to the outer side in this way. Consequently, even if the effective range of the image formed by the interpolation value is shifted to an inner side, an effective range obtained as a result of the shift is the same as the effective range of the original image.

Abstract: An over-sampling unit 2 that over-samples inputted sequential respective sample values to be N sample values and an FIR filter unit 3 that applies filter processing to the over-sampled respective sample values using coefficients formed by a sequence of numerical values (1/2N2, 3/2N2, 5/2N2, . . . , (N?3)/2N2, (N?1)/2N2, (N?1)/2N2, (N?3)/2N2, . . . , 5/2N2, 3/2N2, 1/2N2} (when N is an even number) are provided. Consequently, original discrete sample points are smoothly interpolated along a spline curve according to over-sampling and FIR filter processing and, in a frequency characteristic of an output, a pass band is limited to 2/N of a sampling frequency.

Abstract: An original filter is connected to an adjustment filter in the longitudinal way. The original filter has a first filter coefficient of a symmetric numeric string. The adjustment filter has a second filter coefficient which realizing an invert frequency amplitude characteristic B contacting with a frequency amplitude characteristic A of the original filter at the amplitude maximum value “1”. By executing a product-sum calculation of the first filter coefficient and the second filter coefficient, it is possible to design a desired filter coefficient.

Abstract: A compressing device comprises plural stages of delay circuits (1?1 to 4?1) and multiplying/adding circuits (5?1 to 10?1) that performs weighted addition of output data from the delay circuits (1?1 to 4?1) according to the value of a digital basic function and thereby determines thinned-out data from sampling data sequentially inputted. Since the thinned-out data is determined by the compression part using a digital basic function serving as the original of a sampling function of infinite supports defferentiable once or more times over the whole range, a compression ratio of at lease 8 can be achieved only by the simple four operations. Further, since interpolation data is determined by the decompression part by using the same digital basic function, the original data before the compression can be reproduced with substantial fidelity by only the simple four operations.

Abstract: A standard function is inputted and an interpolation function of finite length is calculated therefrom. Then, the frequency characteristics of the interpolation function is shifted by a desired amount in the frequency axis direction, thereby determining input frequency characteristics based on specification. Filter coefficients are determined by performing inverse FFT of a numerical sequence representative of the input frequency characteristics and rounding is performed according to the coefficient values in order to obtain a smaller number of filter coefficients. Thus, it is possible to eliminate the need for windowing as an operation for decreasing the number of filter coefficients and easily design an FIR filter having desired frequency characteristics.

Abstract: For example, more than one FIR-type basic filters having a symmetric sequence of numbers having a predetermined characteristic as filter coefficient are combined and connected in cascade connection. The filter coefficients are calculated and for the y-bits data of the calculated filter coefficients, the lower bits are cut off for rounding so as to obtain filter coefficients of x-bits (x<y). Thus, it is possible to significantly reduce unnecessary filter coefficients without performing the conventional window multiplication. Moreover, it is possible to realize a digital filter having a desired frequency characteristic with a small circuit size and with a high accuracy without causing a truncation error attributed to window multiplication in the frequency characteristic.

Abstract: A digital-analog converter includes a counter (13) for performing counting operation according to clocks (CK1, CK2), a comparator (12) for comparing the count value to a digital input value and outputting the clocks (CK1, CK2) until the values coincide, switches (SW1, SW2) which turn ON/OFF according to the clocks (CK1, CK2), and a capacitor (C1) which charges and discharges by utilizing constant current sources (21, 22) when the switches (SW1, SW2) are ON. A digital portion (10) including the comparator (12) and the counter (13) is completely separated from an analog portion (20) including the capacitor (C1) and the switches (SW1, SW2) and these portions are connected only by the clocks (CK1, CK2), so that the digital portion (10) and the analog portion (20) can be designed separately.

Abstract: A digital filter having the desired frequency characteristic can be designed through an extremely simple processing, which comprises: generating a plurality of filters, by a frequency shift calculation to a basic filter having a passband width equal to a sampling frequency divided by an integer, from the frequency/amplitude characteristic of the basic filter being shifted by a prescribed frequency so that the adjacent filter banks are overlapped each other at the part of one-half amplitude; and obtaining the final filter coefficients by arbitrarily selecting one or more filters among the basic filter and a plurality of frequency-shifted filters and adding the final filter coefficients thereof.

Abstract: An original filter is connected to an adjustment filter in the longitudinal way. The original filter has a first filter coefficient of a symmetric numeric string. The adjustment filter has a contact point at a position where the maximum value is acquired in the original filter frequency amplitude characteristic A and has a symmetric second filter coefficient realizing the frequency amplitude characteristic B having the minimum value at the contact point. By executing a convolution calculation of the first filter coefficient and the second filter coefficient, it is possible to design a desired filter coefficient.

Abstract: An analog filter includes a first arithmetic operation section 2-1 having a plurality of sets of processing circuit being cascade connected, each processing circuit having an S/H circuit of plural stages for holding a ??-modulated signal and an analog adder for adding the input and output signals of the S/H circuit, in which the number of stages of the S/H circuits 11-1, 14-1, 17-1 and 20-1 decreases toward the end of cascade connection, and a second arithmetic operation section 2-2 configured in the same way, which are cascade connected. By using such an analog filter, over-sampling and convolution of a ??-modulated signal are conducted so that the envelope of the filter output may be a quadratic curve of finite carrier that converges to zero at finite sampling points to prevent phase distortion of an LPF and a discretization error due to a conventional function.

Abstract: An analog filter includes a first arithmetic operation section 2-1 having a plurality of sets of processing circuit being cascade connected, each processing circuit having an S/H circuit of plural stages for holding a ??-modulated signal and an analog adder for adding the input and output signals of the S/H circuit, in which the number of stages of the S/H circuits 11-1, 14-1, 17-1 and 20-1 decreases toward the end of cascade connection, and a second arithmetic operation section 2-2 configured in the same way, which are cascade connected. By using such an analog filter, over sampling and convolution of a ??-modulated signal are conducted so that the envelope of the filter output may be a quadratic curve of finite carrier that converges to zero at finite sampling points to prevent phase distortion of an LPF and a discretization error due to a conventional function.

Abstract: An image processing circuit capable of carrying out high-speed processing and improving the horizontal and vertical resolutions even with its simple structure. The pixel values a to i of a total of nine pixels, three pixels in the horizontal direction along a scanning line by three pixels in the vertical direction in which the line and adjacent lines are abreast, are extracted by a pixel value extracting section. The pixel values A1 to A4 of four pixels Q1 to Q4 generated additionally around the center pixel P5 are determined by calculation by a pixel value calculation section. These pixel values are outputted in units of one scanning line by correlating them with two scanning lines by a pixel value output section.