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). The device eliminates the need of calculating a coefficient upon interpolation calculation and the need of use of a large image memory for performing an interpolation calculation.
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
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 ΔΣ-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 ΔΣ-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.
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 four D flip-flops 10-1 through 10-4, four multipliers 12-1 through 12-4, three adders 14-1 through 14-3, a D/A converter 16, and two integrating circuits 18-1 and 18-2. 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 16 and integrated twice by means of the two integrating circuits 18-1 and 18-2.
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:
An interpolation function generation circuit is formed by cascade connecting a first FIR filter (10) having a numerical value string composed of a ratio “−α, α, β, β, α, −α” (α is an emphasis coefficient and β 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:
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 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 ΔΣ-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 ΔΣ-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.