Icom Incorporated

Weimin Sun

Shigeki Kajimoto

Michio Nakajima

Hirofumi Yamauchi

An input analog signal is sampled at a sampling frequency f.sub.s, which is four times a carrier frequency f.sub.c. A demultiplexer 11 outputs supplied data to four systems sequentially, and generates four zero carriers whose frequencies and amplitudes are equal to each other, with only the phases being different. The delay circuits 12 to 15 delay each one of the zero carriers, and supply delayed signals to adaptive filters 16 to 19. Data output from the four adaptive filters 16 to 19 are selected by a multiplexer 22, and the multiplexer 22 outputs the selected signals as a single signal. The difference between a first zero carrier and an output signals of the first adaptive filter 16 is obtained, and tap coefficients of the adaptive filters 16 to 19 are controlled in accordance with an LMS algorithm based on the difference. A signal output by the multiplexer 22 is output as a reproduced carrier.

Digital filter system, carrier reproduction circuit using the digital

Constants {P.sub.b00i, Q.sub.b00j, . . . , P.sub.ak2i, Q.sub.ak2j, P.sub.ci, Q.sub.cj, P.sub.di, and Q.sub.dj } for calculating each of filter coefficients {a.sub.k1, a.sub.k2, b.sub.00, b.sub.k1, b.sub.k2, c and d} for a digital filter 15 by using non-linear polynomials for pass band position data x and pass band width data y are stored in a memory 13. The constants {P.sub.b00i, Q.sub.b00j, . . . , P.sub.ak2i, Q.sub.ak2j, P.sub.ci, Q.sub.cj, P.sub.di, and Q.sub.dj } are determined by using the least square method so that the sum of the square of the errors between the filter coefficients calculated by using non-liner polynomials and the filter coefficients of digital filter having known characteristics becomes the least or the minimum. A CPU 11 calculates the filter coefficients non-linearly by using the constants stored in the memory 13, and sets the results to the digital filter 15 when new pass band position data x or pass band width data y is input at knobs SC and SW.

Digital filter system

Data from an A/D converter is supplied to a fixed-point DSP. The fixed-point DSP adjusts the level of the data x so that computed results do not exceed .+-.1. In succession, for the i-th data x.sub.i, the fixed-point DSP computes cos .pi..multidot.x.sub.i when i=1, 5, . . . , -sin .pi..multidot.x.sub.i when i=2, 6, . . . , -cos .pi..multidot.x.sub.i when i=3, 7, . . . and sin .pi..multidot.x.sub.i when i=4, 8, . . . and outputs a digital phase modulated signal y(t) by outputting the computed results in order. Trigonometrical functions are computed by expanding them to a series so that the intermediate computed results do not exceed .+-.1.

Digital phase modulator formed with a fixed-point digital signal

An adaptive noise canceling system for improving a S/N ratio and clarity of a output signal includes a delay for delaying the input signal X(j) by a fixed time delay a, a filter for filtering the delayed signal D(j), a subtracter for subtracting the filtered signal Y(j) from the input signal X(j), a coefficients adjuster for adaptively adjusting coefficients of the filter using a least-means-square (LMS) algorithm, a level adjuster for adjusting a level of an error signal .epsilon.(j) which is obtained by subtracting the filtered signal Y(j) from the input signal X(j), and an adder for adding the level adjusted signal .epsilon.(j) to the filtered signal Y(j).

Adaptive noise canceling system and FM radio communication receiver

A 90-degree digital phase shift network includes at least two all-pass digital filters which have a phase difference of 90 degrees therebetween. These digital filters are adapted to receive the same signal. Each of the all-pass digital filters has phase characteristics .theta. (.omega.) given by ##EQU1## where T represents a sampling period, P.sub.i a real number indicative of a filter coefficient, .omega. the angular frequency, m a weight variable, n'=n/2 (n=even number), n'=(n+1)/2 (n=odd number), n an order of a 90-degree digital PSN, m being a positive value other than 1, and satisfying .vertline.2m.omega..vertline. / .vertline.P.sub.i .vertline.&lt;1 when .omega. T is sufficiently lower than 1, while satisfying .vertline.-P.sub.i .omega. T.sup.2 .vertline. / 2m&lt;1 when .omega. T is substantially equal to .pi..

90-degree phase shift network, system for controlling a 90-degree phase

A 90-degree digital phase shift network includes at least two all-pass digital filters which have a phase difference of 90 degrees therebetween. These digital filters are adapted to receive the same signal. Each of the all-pass digital filters has phase characteristics .theta.(.omega.) given by ##EQU1## where T represents a sampling period, P.sub.i a real number indicative of a filter coefficient, .omega. the angular frequency, m a weight variable, n'=n/2 (n=even number), n'=(n+1)/2 (n=odd number), n an order of a 90-degree digital PSN, m being a positive value other than 1, and satisfying .vertline.2m.omega..vertline./.vertline.P.sub.i .vertline.&lt;1 when .omega.T is sufficiently lower than 1, while satisfying .vertline.-P.sub.i .omega.T.sup.2 .vertline./2m&lt;1 when .omega.T is substantially equal to .pi..

90-degree digital phase shift network and linearizer using all-pass

A DMUX 11 outputs data in a digital FM signal in a circulation to produce signals S1 to S4. An MUX 14 alternately outputs inverted signals of the signals S1 and S3, obtaining a signal Si.sub.(k). An MUX 15 alternately outputs inverted signals of the signals S2 and S4, obtaining a signal Sq.sub.(k). An interpolation circuit 17 inserts data having a value of zero between individual data in the output signal of the MUX 14, and cuts off the high-frequency component of the resultant signal, obtaining a signal Si.sub.(i). A second interpolation circuit 19 inserts data having a value of zero between individual data in the output signal of the MUX 15, and cuts off the high-frequency component of the resultant signal, obtaining a signal Sq.sub.(i). An arithmetic operation circuit 2 computes SI.sub.(i-1) .multidot.SQ.sub.(i) -SI.sub.(i) .multidot.Sq.sub.(i-1), and integrates it to obtain a digital demodulated signal.

Digital demodulator for a frequency modulated signal and an amplitude

A CW (Continuous Wave) radio frequency signal converted to an IF (Intermediate Frequency) signal is mixed with a local oscillator signal by an IF mixer, and the obtained signal is converted to a digital signal by an ADC (Analog-to-Digital Converter). A digital bandpass filter passes only the target component in the output of the ADC. The passed IF signal is demodulated by a digital demodulator, and the demodulated signal is converted to an analog signal before being output as a sound. When pitch control to increase the reception pitch by .DELTA.f is instructed, an MPU (Microprocessor Unit) alters the frequency of the local oscillator signal by .DELTA.f as well as the center frequency of the digital bandpass filter by .DELTA.f, so that the center frequency of the input signal to a narrow-band filter coincides with the center frequency of the digital bandpass filter.

Radio communication system

A demodulator receives a radio signal and causes a carrier signal reproducing circuit to reproduce the carrier signal of the received signal. One amplifier amplifies the amplitude of the reproduced carrier signal by K, and another amplifies the amplitude of the reproduced carrier signal by (K+2). An adder adds the reproduced carrier signal, amplified by K, and the received signal. A subtracter subtracts the received signal from the reproduced carrier signal amplified by (K+2). The output signal of the adder is demodulated by a first FM demodulator and the output signal of the subtracter is demodulated by a second FM demodulator. Another subtracter outputs the difference between the demodulated signals from the first and second FM demodulators as a demodulated signal of the received signal.

Demodulator for demodulating a single side band signal, a real zero

The filter coefficients of a digital band pass elliptic filter can be approximated to a linear function of the center frequency of the pass band width. The filter coefficients of a band pass digital filter having a predetermined center frequency and the slope of this linear function are stored in first and second memories. A central processing unit (CPU) obtains the difference between a desired center frequency and a predetermined center frequency. A multiplier multiplies the slope stored in the second memory by the difference obtained by the CPU. An adder adds each of the filter coefficients stored in the first memory to an associated one of the products output from the multiplier. The adding results become the filter coefficients for obtaining the desired center frequency. Those filter coefficients are set in a digital filter.

Digital band pass elliptic filter system

A Single-Sideband (SSB) modulator includes:

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Nara, Japan

Weimin Sun