Abstract:
A symmetric unit filter (L10″) is designed by combining in series two unit filters (1L10, 2L10) having asymmetric numerical sequences as filter coefficients (H1 to H3 and H4 to H6). Thus, by designing a filter by series connection, it is possible to automatically obtain a desired digital filter coefficient only by combining one type of unit filter (L10″) in series. 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 and H4 to H6). This reduces the number of necessary taps, eliminates use of a window function, and prevents generation of a discretization error in the filter characteristic obtained.
Abstract:
A digital input is 8-fold oversampled, delay circuits 11−1 to 11−4 and multipliers/adders 12 to 15 perform a convolution arithmetic while multipliers/adders 4 to 10 process the oversampling data into predetermined digital basic waveforms, and continuous interpolation values can be obtained, thereby requiring no low pass filter which causes deterioration of phase characteristics, and suppressing an interpolation abortion error by setting finite interpolating functions. Furthermore, by obtaining apart of oversampling data as input data by using an AND gate 2, a subsequent process using the digital basic waveform and the convolution arithmetic can be performed in a very simple process.
Abstract:
There are provided: an offset adding section 3 for converting signed digital data to unsigned digital data by adding an offset value to signed digital data to be compressed, a rounding section 5 for reducing the number of bits per word by performing a rounding operation of a lower-order bit on the unsigned digital data, a timing synthesizer 6 and a compressing section 7 for sampling the digital data undergoing a rounding operation at a time interval of a point where a differential value varies in polarity and for obtaining as compressed data a pair of a discrete amplitude data value on each sample point and a timing data value indicative of a time interval between sample points. When a signal is compressed and expanded on a time base, the operation can be performed on the time base without frequency conversion.
Abstract:
It is object to provide an oversampling circuit and a digital-to-analog converter which have small circuit scales and component costs of which are reduced. The data input at predetermined intervals is multiplied by the multiplying section 1 by four types of multiplicators, and the multiplication results are cyclically held in data holding sections 2-1 through 2-4. The data selectors 3-1 through 3-4 sequentially read out the four types of multiplication results held in the data holding sections 2-1 through 2-4 one to one corresponding respectively to the data selectors 3-1 through 3-4 in a predetermined order, thereby outputting predetermined step functions. Each step function corresponding to each of the four pieces of data inputted sequentially is generated at a different timing, each of the integrating sections 4-1 through 4-4 performs a digitally integrating process two times on each step function, and the results are added up, thereby performing the oversampling process to give a pseudo-rise of a sampling frequency to each piece of input digital data.
Abstract:
It is object to provide an oversampling circuit and a digital to analog converter capable of realizing a smaller circuit and reducing a cost of parts. The oversampling circuit comprises multiplying section 1, four data holding sections 2-1 through 2-4, four data selectors 3-1 through 3-4, an adding section 4, and two integrating circuits 5-1 and 5-2. Input data is multiplied by four multiplicators by the multiplying section 1, and four multiplication results held, as one set, in the data holding sections. The data selectors read out the data held in the four data holding sections in a predetermined order and generate step function data. The adding section adds the values of four step functions outputted from the respective data selectors, and then digital integrating operations corresponding to the sum are carried out by means of two integrating circuits.
Abstract:
A CCD television camera has a head part separated from the rest of the camera. The camera head part contains a CCD imaging device and is connected by only a single line coaxial cable to a main part containing a DC power source, a line flag output circuit, and a one H-alternate switch. The coaxial cable may be up to several hundred meters in length. Respective ends of the coaxial cable are connected to the head and main parts through hydrid coils to facilitate sending of DC power current and an accurately controlled sinusoidal clock frequency signal from the main part to the camera head part, and the sending of video signals over the same single cable line from the camera head part to the main part. The hybrid coils are tuned for resonance with the clock frequency and a PLL circuit, including the coaxial cable, regulates the clock frequency to adjust for different lengths of cable.
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 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 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.