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 waveform of a desired frequency characteristic is input as a numeric value string and is subjected to reverse FFT so as to obtain a filter coefficient group. Thus, without having any expert knowledge, 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 to 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:
An interpolation circuit capable of performing interpolation operation with a simple constitution. A 16-times oversampling from discrete data is performed by D frip-frops 4, 5. A first convolution operation is performed by D frip-frops 4 through 11 and an adder 12, on the result of which a second convolution operation is performed by D frip-frops 13 through 20 and an adder 21. Data interpolated along a quadratic function curve interpolating the discrete data is obtained from the adder 21.
Abstract:
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.
Abstract:
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 a multiplying section 1, four data holding sections 2-1 through 2-4, four data selectors 3-1 through 3-4, an adding section 4, a D/A converter 5, and two integrating circuits 6-1 and 6-2. Input data is multiplied by four multiplicators by the multiplying section 1, and the four multiplication results are 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 the step functions outputted from the four data selectors. Furthermore, a stepwise analog voltage corresponding to the sum is generated by the D/A converter 5 and integrated twice by means of the two integrating circuits 6-1 and 6-2.
Abstract:
A digital-to-analog converter for generating output waveforms with less distortion without the need for high-speed components. The digital-to-analog converter comprises four data holding sections, four step function generators, an adding section, a D/A converter, two integrators and a timing controller. Four digital data successively inputs are held in the data holding sections, respectively, and the step function generators generate step function whose values corresponding to the held data. The adding section sums the step functions generated in the step function generators, and the D/A converter generates the analog stepwise voltage corresponding to the summed value. The two integrators integrate this combined waveform two times, thus producing a continuous analog voltage that connects the input digital data smoothly.
Abstract:
A system for obviating the special operation needed for storing a program in memories in a channel selecting system used in television sets, etc., which is adapted to make the control for selecting specified channels at specified times, following a program which has been preliminarily stored in the memories, by writing into the memories from the channel control means the state of channel selection unmodified, as the program, which is based on the channel selecting operation actually performed by the user, using the commanding means during normal real time operation, as a way of storing the program in the memories.
Abstract:
A D-A converter for converting a digital signal to a duty factor of a pulse train signal and for averaging the pulse signal by a low-pass filter to convert it to an analog signal is disclosed. A plurality of pulses which are to be selected in accordance with the input digital signal comprise a plurality of basic pulses of different phases and pulse widths derived by frequency division and auxiliary basic pulses which occur at a cycle period which is at least twice as long as a repetition cycle period of the basic pulses. The basic pulses are selected by high order bits of the digital signal while the auxiliary basic pulses are selected by low order bits of the digital signal to produce an output pulse train signal a duty factor of which changes in accordance with the digital signal. The pulse train signal is then averaged to convert it to an analog signal.
Abstract:
Disclosed is a digital-to-analog converter wherein a counter comprising a number of n - m (where n > m) flip-flops is so arranged that an m-bit and (n - m)-bit outputs may be alternately derived, weighted and added to derive an n-bit output which in turn is smoothed to deliver a DC output.