Abstract: A cutoff frequency adjusting circuit includes a filter circuit (1) provided with a plurality of resister elements, and a switch to one of the resister elements, and a capacitor. A cutoff frequency of the filter circuit (1) is determined by a resistor value of the resister element selected by the switch and capacitive value of the capacitor. The cutoff frequency adjusting circuit further includes a clock signal generator (2) that generates first and second frequency clock signals (CK1) and (CK2), and a DSP (3) that compares a level of an output signal output from the filter circuit (1) when the first frequency clock signal (CK 1) is input to the filter circuit (1) and that of an output signal output from the filter circuit (1) when the second frequency clock signal (CK2) is input to the filter circuit (1) and that controls the switch in response to its comparing result.

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: An LNA (2) is directly connected to a loop antenna (1), and a variable BPF (3), the passed frequency band of which is adapted to be variable, is connected in a stage following the LNA (2). In this way, the variable BPF (3) is used to configure a variable tuning circuit exhibiting a high Q-value without using any impedance conversion transformer between the loop antenna (1) and the LNA (2), whereby almost all of the constituent elements of the tuner part can be integrated in an IC chip (10) and further the variable tuning can maintain an excellent selectivity of a target frequency.

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: 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 object of the invention is to provide an adaptor device for a car radio capable of improving the receiving state. The adaptor device 20 for a car radio is connected to a vehicle-mounted antenna connection terminal 12 provided in an FM radio receiver 10 installed within a vehicle room, and includes a signal line inserted between the vehicle-mounted antenna connection terminal 12 and a vehicle-mounted antenna 110; and an in-vehicle antenna 22 branching out from the signal line. A radio wave transmitted from an FM transmitter 40 disposed within the vehicle room via an antenna 42 is received by the radio receiver 10 via the in-vehicle antenna 22.

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: An object of the invention is to provide an FM radio receiver capable of improving the receiving state. The radio receiver 10 is disposed within a vehicle room and provided with a vehicle-mounted antenna connection terminal 12. The radio receiver 10 includes a front-end unit converting to an intermediate frequency signal an FM broadcast signal received via a vehicle-mounted antenna 110, a signal line inserted between the vehicle-mounted antenna connection terminal 12 and the front-end unit, and an in-vehicle antenna 14 branching out from the signal line. A radio wave transmitted from an FM transmitter 40 disposed within the vehicle room via an antenna 42 is received by the radio receiver 10 via the in-vehicle antenna 14.

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: An object of the present invention is to provide an FM transmitter in which the degree of freedom in parts selection is improved.

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: A numerical string consisting of a ratio of “?1, m, ?1” or “1, m, 1” is subjected to a predetermined moving average calculation n times. A numerical string thus obtained is used as filter coefficients of a basic filter and at least one basic filter is combined in an arbitrary way for cascade connection, thereby calculating the filter coefficients of the digital filter to be obtained. This significantly reduces the number of taps and the number of multipliers used as compared to the conventional FIR filter. Moreover, by using the numerical strings “?1, m, ?1” and “1, m, 1” so that the filter impulse response becomes a finite-base function, it is possible to obtain a preferable frequency characteristic having no discretization error and having a great attenuation amount out of band.