Abstract: The present invention addresses the problem of increasing the lifetimes of an oscillatory wave motor and a sound generation device using the oscillatory wave motor as a drive source, and proposed the structure and mechanism for integrally increasing the lifetimes of the oscillatory wave motor and the sound generation device using the oscillatory wave motor as the drive source. To maintain the drive performance of a drive unit, the drive unit is provided with a core sheathing structure, with the result that the drive unit is prevented from breaking and being damaged, and even if the entire drive unit is worn down, an intrinsic drive unit member serving as a core portion keeps the same drive area, so an initial drive feature is maintained. The descending order of a wear resistance of the members is the drive unit core material, a moving unit, and a drive unit sheathing material.

Abstract: An active noise control apparatus for transmitting, from a loud speaker, a secondary noise synthesized so as to have the same amplitude as and the opposite phase to a primary noise and for canceling the noise by acoustically overlapping the secondary noise. An overall system filter for simulating a characteristic of an overall system leading to an error detecting microphone from a noise detecting microphone is provided for the first and the second overall system filter. A second and a first noise control filter are connected to the first and the second overall system filter in cascade to form a noise control filter. A coefficient of an estimating noise transfer system filter, obtained when the difference between the differential output of both obtained when a white noise is applied to the circuit of cascade connections and the response difference of the first and the second overall system filter from a differential overall system filter becomes a minimum, is made the coefficient of the noise control filter.

Abstract: A method of updating reflection coefficients of a lattice filter is provided, which method includes the step of updating backward reflection coefficients by employing forward reflection coefficients which are updated so that a forward prediction error of a last stage of the lattice filter is minimized.

Abstract: A noise suppression apparatus is realized by at least one linear prediction analyzing circuit. Each linear prediction analyzing circuit includes: an adaptive filter which produces a linear prediction signal based on a first speech signal on which noise is superimposed, and outputs the linear prediction signal as a second speech signal in which the noise is suppressed; a subtraction unit which obtains a difference between the linear prediction signal and the first speech signal, and outputs the difference as a prediction error; and a coefficient updating unit which updates coefficients of the adaptive filter based on the first speech signal and the prediction error so as to minimize the prediction error. The noise suppression apparatus may includes a cascade connection of a plurality of linear prediction analyzing circuits each having the above construction.

Abstract: A voice switch in a hands-free communications system performs selective attenuation with respect to respective voice signals being transmitted and received in respective transmitting and receiving paths, the transmitted voice signal having been converted from an audible voice by a microphone connected to the transmitting path and the received voice signal having been converted by a loudspeaker to an audible voice output. A detector selectively detects a currently transmitted voice signal at a normal level, a currently received voice signal at a normal level, and drops in the respective levels thereof to nil levels and provides corresponding outputs to a controller. When one of the transmitted and received voice signals is of a normal level and the detector newly detects the other thereof at a normal level, the controller selectively attenuates that other, newly-detected normal level voice signal.

Abstract: An apparatus for estimating filter coefficients operates in a system which includes a filter simulating characteristics of an unknown signal transmission system based on a signal provided to the unknown signal transmission system and a response signal produced from the unknown signal transmission system. The apparatus includes an accumulation calculating section for accumulating a product of the signal provided to the unknown signal transmission system and a difference between the response signal from the unknown signal transmission system and an output signal of the filter for a given time period. The apparatus further includes a square-sum calculating section for accumulating a square of the signal provided to the unknown signal transmission system for the given time period.

Abstract: An estimation apparatus predicts filter coefficients for an adaptive filter, the response of which emulates the signal transmission characteristics of a known signal. The response thereto is sent to a signal transmission system of unknown characteristics, enabling execution of calculations without invalidating coefficient updating, even when there is a limit on the word length for processings. To achieve this, a sum of products calculation unit calculates the sum of products of the residual difference in response and the signal which is sent to the signal transmission system. A sum of squares calculating unit calculates the sum of the squares of the signal sent to the signal transmission system over a prescribed period of time. An updating amount calculation unit calculates the filter coefficient updating amounts from the ratio of the results from the sum of the products calculating unit to the results of the sum of the squares calculating unit.

Abstract: A method and apparatus of determining the transfer characteristic in an active-noise-control system, which involves generating white noise at an end of a one-dimensional sound field that is defined by a linear ventilating system in which sound travels essentially parallel to the extended direction of the system; equalizing the transfer characteristic of the one-dimensional sound field and generating cancelling sound, according to an inverse of the transfer characteristic, to cancel the white noise and prevent noise being output from the other end of the one-dimensional sound field; continuously preventing the noise output and measuring the characteristic data of the one-dimensional sound field at, at least, one measuring point in the one-dimensional sound field; and calculating the transfer function of the one-dimensional sound field in the noise-output-prevented state, according to the characteristic data of the sound field.

Abstract: A signal suppressing apparatus for suppressing undesired signal components, in which a first filter electrically simulating a propagation system A of a physical phenomena such as a sound, vibration, electricity, etc. is mounted; a signal having a correlation to the signal to be input to the propagation system A is input to this first filter; and the output of the first filter exerts an influence upon the signal passing the propagation system A to make the passed signal approximate the intended value, wherein the first filter is split into a plurality of filters, each of which is composed by a discrete DSP, outputs are individually calculated in the split individual filters, and the individual calculation results are tabulated, whereby a value the same as the value obtained where the calculation is made without splitting of the first filter is obtained. This enables high speed signal processing even if a high speed DSP is not used.

Abstract: A speaking apparatus having a handfree conversation function, provided with an echo canceler which can cancel the echo caused by direct acoustic coupling between a speaker and a microphone positioned in a system having a casing, a ground surface and a speech switching circuit. In the apparatus, the amount of insertion attenuation of the transmitted and received signal is set with an upper limit of the amount of attenuation sufficient for cancelling the echo caused by the indirect acoustic coupling determined by the location where the speaking apparatus is used. In addition, the echo canceler and the speech switching circuit share optimum functions to obtain an excellent speaking quality apparatus with an inexpensive and small sized processing circuit.

Abstract: An acoustic echo canceler and a side-tone echo canceler provided in a hands-free telephone suppress an acoustic echo and a side-tone echo respectively with few error by providing automatic gain controllers and/or limiters in the hands-free telephone so that the acoustic echo canceler and the side-tone echo canceler operate in linear, and a directional characteristic of a microphone used in the hands-free telephone for reducing the acoustic echo is controlled by a microphone direction controller so that the microphone operates as an omnidirectional microphone when a level of a received signal of the hands-free telephone is less than a designated level and as a bidirectional microphone when the level exceeds the designated level, further gains of an output of the microphone increases in a low frequency range.

Abstract: A method for detecting a transition of an echo path used for an estimation of the transfer function of a system by using an adaptive filter, from an echo caused by a signal transmitted through an input terminal of the system at which the response of the transmitted signal is received. The method includes the steps of calculating the amount of the whole or the first delay portion of an impulse response of the system, calculating the amount of the remaining delay portion of the impulse response of the system, calculating the ratio between the amount of the whole or the first delay portion of the impulse response and the amount of the remaining delay portion of the impulse response, and deciding an occurrence of a transition of the characteristics of the system based on the calculated ratio. The transition of the characteristics of the system is discriminated from a hindering signal produced in the system in the decision making process.

Abstract: An adaptive digital filter for estimating a response characteristic of a signal path by monitoring a sampled output signal of the signal path, estimating a predetermined number of filter coefficients which represent the response characteristic of the signal path, and generating an estimated output signal of the signal path using a plurality of successive sampled input signals of the signal path and the estimated filter coefficients, where the estimation is carried out so that a difference between the output signal and the estimated output signal is reduced. Each of the filter coefficients is extracted by a low-pass filter where the low-pass filter coefficient in the low-pass filter can be set to a constant. Normalization can be carried out in either the input side or the output side of the low-pass filter. Otherwise, the low-pass filter coefficient may be set to 1-K.multidot.X.sub.j (m).sup.

Abstract: An electronic telephone terminal having a transmitter and a receiver, both having an approximately linear acoustic-to-electric transduction characteristics, and having a surrounding noise suppression function, and including a variable attenuator for controlling a gain of a transmission system; a noise detection device for detecting surrounding noise; and a control device for controlling the variable attenuator in such a manner that when a sound pressure level input to the transmitter exceeds a predetermined threshold value, the gain is fixedly set to a constant value, and when the input sound pressure level is equal to or below the predetermined threshold level, the gain is controlled in response to a change in the surrounding noise level detected by the noise detection device.

Abstract: An inverse control system is disclosed, which comprises FIR filters provided between transmitting elements at n (n=2, 3, . . . ) input points of a linear FIR system and a common signal source, for an inverse control such as to provide desired impulse responses between the signal source and m (n>m) output points of the linear FIR system. A j-th (j=1, 2, . . . , n) one of the FIR filters has a number L.sub.j of taps which satisfies the relationships represented by ##EQU1## for all i=1, 2, . . . , m and j=1, 2 . . . , n where w.sub.ij is the number of discrete signals representing the impulse response g.sub.ij (k) between the j-th output point and i-output point and P.sub.i is the number of discrete signals representing the desired impulse response r.sub.i (k) between the signal source and i-th output point. The j-th FIR filters has a filter coefficient h.sub.j (k) satisfying the relationship ##EQU2## where .circle.* is a discrete convolution.

Abstract: An acoustic signal is received by a plurality of microphone elements and their outputs are delayed by delay means and weighted and summed up by weighted summation means, obtaining a noise-reduced output. A fictitious desired signal is electrically generated and the weighting values of the weighted summation means is determined based on the fictitious desired signal and the outputs of the microphone elements when receiving substantially only noises.