Kokusai Denshin Denwa Co., Ltd.

Kokusai Denshin Denwa Kabushiki Kaisha

Sotokichi Shintani

Hiroshi Sakaki

Michitoshi Tamori

Hidetaka Yanagidaira

Hideo Kobayashi

Akira Sato

Katsuhiko Furuya

Kazuo Kawai

Hidemi Kuroda

Masamichi Niiya

Masaru Fukai

Nobuo Nagai

Masayuki Takeuchi

An improved Kalman equalizer for providing a tap weight of a transversal filter for equalizing a transmission line has been found. A complex number fast Kalman equalizer (FIG. 27), which is the improvement of the prior fast real number Kalman equalizer (see; 'Fast calculation of gain matrices for recursive estimation schemes' by Lennart Ljung, et al, INT. J. Control, 1978, vol. 27, No. 1, 1-19), has a complex conjugate generator (C.sub.2) for processing a complex input signal so that an equalization can be performed on the bandwidth division concept, and a transmission line having a complex impulse response can be equalized. Then, a bandwidth division equalization in which a whole bandwidth is separated into a plurality of sub-bandwidth each being equalized by a related sub-equalizer becomes possible. Also, a QAM signal (Quadrature Amplitude Modulation) which has a complex impulse response can be equalized. A bandwidth division fast Kalman equalizer (FIG. 68) is also possible as the modification of the present invention.

Kalman equalizer

An adaptive control type echo suppressor which compares the levels of voice signals of sending and receiving paths by a first comparator with each other and cuts off the sending path to suppress an echo when the level of the voice signal of the receiving path is higher than the level of the voice signal of the sending path. A control loop is provided for applying the voice signal from the receiving path to a second comparator via a variable loss circuit for comparing the peak of the integrated value of the same with the peak level of the integrated value of the voice signal of the sending path to effect adaptive control of the variable loss circuit by the output of the second comparator in a manner to reduce the output of the second comparator to zero. Each of the integrated values are obtained under a relatively long time constant characteristic to prevent influence of noise of the sending and the receiving paths, and the output of the variable loss circuit is applied as the voice signal of the receiving path to the first comparator.

Adaptive control type echo suppressor

A smear filter and/or a de-smear filter having the characteristics that the delay time is proportional to the frequency comprising (a) a transversal filter having a plurality of delay elements connected in series with one another, the delay time (T) of each delay element being the same as the sampling interval of the input signal, the first delay element being connected to the input terminal, a plurality of potentiometers each connected to the junction points of said delay elements and the output of the last delay element providing the tap weight to each of the tap output, and an adder for providing the sum of the outputs of said potentiometers to provide an output of the present delay circuit, (b) a digital memory storing the precalculated value for each desired delay characteristics for providing the tap weights of said potentiometers, and (c) means for providing the content of said digital memory selectively to the potentiometers according to the desired delay characteristics.

Delay circuit

An apparatus for reducing a sampling frequency which derives information with the bandwidth f.sub.B /N (N is an integer) having a sampling frequency of 2f.sub.B /N from an input FDM signal with a bandwidth up to f.sub.B having the sampling frequency 2f.sub.B has been found. The present apparatus comprises a pair of partial frequency reduction systems and a switch for alternately selecting the outputs of said partial frequency reduction systems to provide the output signal. Each of said partial frequency reduction systems comprises a DFT means for performing the discrete Fourier transform for every G sampling points (G is the multiple of N) to said input signal, a multiplication means for multiplying the predetermined frequency characteristics with the output of said DFT means, linear conversion means for linear conversion of the output of said multiplication means and for providing G/N number of data, and means for performing the inverse discrete Fourier transform to the output of said linear conversion means. The present invention reduces sampling frequency by performing a smaller number of multiplications than a prior art, thus, the structure of an apparatus can be simple.

Apparatus for reducing a sampling frequency

A method and apparatus is provided for adjusting the tap coefficient of an adaptive automatic equalizer having a plurality of tap registers. An input signal is repetitively applied to a train of tapped delay elements and the sum of the weighted output of each tap output is obtained. An error output is obtained by taking the difference between the sum of the weighted output of each tap output and a reference signal, and a correlation value is obtained between the error output and each tap output of the delay elements. The improvement comprises storing the tap weights of each of the tapped registers and obtaining the total sum of each tap weight of one of the registers wherein the tap weight of a tap is the correlation value. This step is repeated M times, where M is an integer so that the total sum of each tap weight in another of the tapped registers is obtained and the error output is added to the total sum, determining a value of tap coefficient by the final total sum after M adjustments of the most rapid equalization. The optimum coefficient is obtained by using the value of the tap coefficient for the most rapid equalization of the tapped registers and the weight of the taps of the delay element is revised by the optimum coefficient-weight-sum of the contents of the tapped register and the tapped delay elements, thereby adjusting the weight of each tap of the delay elements.

Method for establishing a tap coefficient of an adaptive automatic

A Bessel function type automatic delay equalizer comprising a plurality of transversal filters each of which has an independent delay cosine equalization component having a predetermined period, and the superposition of those filters providing the desired characteristics, characterized in that, said equalizer further comprises a Bessel function generator for controlling the tap gain of said transversal filters.

Bessel function type automatic delay equalizer

In an automatic equalizer for regenerating a received digital signal, comprising a delay element train with a plurality of taps, a coefficient matrix circuit and a group of variable coefficient units, the structure of said coefficient matrix circuit being considerably simplified by introducing a simple adder and/or subtractor at the input and/or the output of said coefficient matrix circuit.

Automatic equalizer having a coefficient matrix circuit

An equalizer having improved amplitude characteristics and phase characteristics of more than several hundred MHz has been found by using a four port directional coupler. The equalizer consists of coupled lines, and two far end ports of the directional coupler are grounded through two resistances R.sub.1, respectively; while interconnecting the two far end ports through a resistance r, an input port is formed at one of the near end ports of the directional coupler, and an output port is formed at the other end thereof, said resistances substantially satisfying the relation of

Constant-resistance coupled-line type equalizer

A system for detecting the signal quality of a phase-modulated wave, in which the phase positions of the signal elements of the digital phase-modulated wave are each detected with respect to a reference phase carrier by a phase detector for developing digital output signals composed of higher digits representing the ideal quantized phase positions of the phase-modulated wave signal elements and at least one lower digit representative of the direction of phase deviation of the signal elements from the ideal quantized phase positions so that the lower digit is taken out as signal quality information of the phase-modulated wave.

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Tokyo, Japan

Sotokichi Shintani