Other

Neuro Solution Corp.

Yasue Sakai

Nsc Co., Ltd.

Neuro Solutions Corp.

Yukio Koyanagi

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.

Digital filter, design method thereof, design device, and digital filter design program

A digital filter is designed by combining unit filters (L', H&#x2032;) having a predetermined asymmetric numerical sequence as filter coefficients (Hto H). Thus, it is possible to automatically obtain a desired digital filter coefficient only by combining the unit filter. Moreover, a symmetric numerical sequence {&#x2212;1, 0, 9, 16, 9, 0, &#x2212;1}/32 is divided at the center into two parts and one of them is used as the asymmetric filter coefficients (Hto H). 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.

Digital filter design method and device, digital filter design program, and digital filter

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.

Tone quality adjustment device designing method and designing device, tone quality adjustment device designing program, and tone quality adjustment device

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 (-) by the filter factors given by factor units (-), 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., {&#x2212;1, 0, 9, 16, 9, 0, &#x2212;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.

Digital filter and its designing method

A compressing device comprises plural stages of delay circuits (to) and multiplying/adding circuits (to) that performs weighted addition of output data from the delay circuits (to) 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.

Compressing device and method, decompressing device and method, compressing/decompressing system, program, record medium

A digital-analog converter includes a counter () for performing counting operation according to clocks (CK, CK), a comparator () for comparing the count value to a digital input value and outputting the clocks (CK, CK) until the values coincide, switches (SW, SW) which turn ON/OFF according to the clocks (CK, CK), and a capacitor (C) which charges and discharges by utilizing constant current sources () when the switches (SW, SW) are ON. A digital portion () including the comparator () and the counter () is completely separated from an analog portion () including the capacitor (C) and the switches (SW, SW) and these portions are connected only by the clocks (CK, CK), so that the digital portion () and the analog portion () can be designed separately.

Digital-analog converter

An analog filter includes a first arithmetic operation sectionhaving a plurality of sets of processing circuit being cascade connected, each processing circuit having an S/H circuit of plural stages for holding a &#x394;&#x3a3;-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 circuitsanddecreases toward the end of cascade connection, and a second arithmetic operation sectionconfigured in the same way, which are cascade connected. By using such an analog filter, over sampling and convolution of a &#x394;&#x3a3;-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. Compared with a conventional circuit for over-sampling and convolution, the number of stages of the S/H circuits and the number of adders are small.

Analog filter suitable for smoothing a &#x394;&#x3a3;-modulated signal

A data interpolating device comprises plural stages of delay circuits () for delaying discrete data sequentially inputted and multiplication/addition circuits (to) that performs weighted addition of data outputted from the output stages of the plural stages of delay circuits () according to the value of a digital basic function (&#x2212;1, 1, 8, 8, 1, &#x2212;1) and thereby determine interpolation data. Since a sampling function of finite supports differentiable once or more times over the whole range.

Data interpolating device and method, sampling function generating device, data interpolating program, and recorded medium

On a compression side, from an inputted analog signal , points to where a differential absolute value is at a predetermined value or smaller are detected as sample points, and a pair of discrete amplitude data on the sample points and timing data indicative of a time interval between sample points is obtained as compressed data. On an expansion side, amplitude data and timing data that are included in the compressed data are used to obtain expansion data by determining interpolation data for interpolating two pieces of amplitude data, based on the two pieces of amplitude data on two successive sample points and timing data therebetween. Thus, when a signal on a time base is compressed and expanded, the operation can be performed on a time base without frequency conversion.

Method and apparatus for compression, method and apparatus for decompression, compression/decompression system, record medium

With respect to data having periodicity to be compressed, windows of the same size are set for every two sections according to an interval of peaks appearing substantially periodically and processing for sorting sample data alternately among the set windows of the same size is sequentially performed, whereby a frequency of data having periodicity is replaced with an approximately half frequency without damaging reproducibility to original data at all to make it possible to apply compression processing to data of the replaced low frequency. If this sorting processing is applied to compression processing having a characteristic that a compression ratio is not increased in a high-frequency region, it becomes possible to improve a compression ratio without damaging a quality of reproduced data by decompression at all.

Alternate window compression/decompression method, apparatus, and system

A digital input is 8-fold oversampled, delay circuits to and multipliers/adders to perform a convolution arithmetic while multipliers/adders to 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 , a subsequent process using the digital basic waveform and the convolution arithmetic can be performed in a very simple process.

Interpolating function generating apparatus and method, digital-analog converter, data interpolator, program, and record medium

There are provided: an offset adding section for converting signed digital data to unsigned digital data by adding an offset value to signed digital data to be compressed, a rounding section 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 and a compressing section 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.

Compressing method and device, decompression method and device, compression/decompression system, and recorded medium

A digital-analog converter and method for lowering the computational complexity and decreasing the circuit scale, and data interpolating device and method. The digital-analog converter has a fundamental digital waveform generating section for generating a fundamental waveform according to the values of inputted discrete data, a convolution calculating section for carrying out oversampling and convolution of the output from the digital waveform generating section , and a D/A converting section for D/A converting the output from the convolution calculating section . The amplitude of the fundamental digital waveform is modulated according to the discrete data and the data is combined by convolution. Only by carrying out oversampling and convolution, continuous interpolation values are obtained.

Digital-analog converter and method, and data interpolation device and method

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 by four types of multiplicators, and the multiplication results are cyclically held in data holding sections - through - . The data selectors - through - sequentially read out the four types of multiplication results held in the data holding sections - through - one to one corresponding respectively to the data selectors - through - 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 - through - 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.

Oversampling circuit and digital/analog converter

It is object to provide a digital-to-analog converter capable of generating an output waveform having less distortion without increasing the operating speed of components. A D/A converter comprises four D flip-flops - through - , four multipliers - through - , three adders - through - , a D/A converter , and two integrating circuits - and - . Input data is fed sequentially to the four D flip-flops and held therein. The multiplier multiplies the data held in the respective D flip-flops by different multiplicators in the first half and second half of one clock period and the multiplication results are added by the three adders. A stepwise analog voltage corresponding to the sum is generated by the D/A converter and integrated twice by means of the two integrating circuits - and -

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Saitama, Japan

Yukio Koyanagi