Abstract: A dual purpose loudspeaker incorporating at least two audio signal inputs and at least one switch is adapted to receive and reproduce either a single audio signal input or two audio signal inputs according to the position of the switch. A switch slide mechanically linked to the switch is provided which blocks one of the audio signal inputs when the switch is in the single-channel position, thereby preventing erroneous connection to the blocked audio signal input.

Abstract: An all-in-one headset worn on the user's head, comprises a head band, a first housing arranged at the head band, a communication unit having selectable at least two communication channels and selectable at least two communication modes, an operating panel mounted on the outside wall of the first housing for selecting one communication channel and one communication mode, an ear-speaker mounted on the inside wall of the first housing, a microphone, a second housing for housing the microphone, a boom member for supporting the second housing to the first housing, an indicating unit for indicating the indication indicative of the selected communication channel and the selected communication mode.

Abstract: An audio compensation system (20) for producing a sound compensated output signal (So). The system comprises a volume control circuit (24) for producing a volume-adjusted signal (Sic) by applying a volume adjustment to an audio signal in response to a volume setting, wherein the sound compensated output signal is responsive to the volume-adjusted signal. The system also comprises circuitry (26, 30, 32) for producing an amplified signal (Sifa) by amplifying a selected bandwith of signals in response to the volume setting. The sound compensated output signal is also responsive to the amplified signal.

Abstract: Noise reduction arrangement including: a plurality of actuators (3(n)) for generating secondary noise (ps) to reduce primary noise (pp) and being located in a first surface; a plurality of errors sensors (2(m)) located in a second surface parallel to the first surface for sensing a total amount of noise resulting from the primary noise after being reduced by the secondary noise; a plurality of control elements (5(i)) for controlling the actuators (3(n)) based on the sensor outputs, wherein the distance (d) between the first and second surfaces is such that reduction in power RP of the total amount of noise relative to the primary noise within a predetermined frequency band is within the following range: 0.9×RPmax?RP?RPmax in which RPmax is the maximum obtainable reduction in power of the total amount of noise relative to the primary noise, both RP and RPmax being expressed in decibel.

Abstract: An apparatus and method for remotely and automatically adjusting the volume of a remotely controlled audio device. In one embodiment, the apparatus comprises a sensor circuit for continuously detecting audio signals generated by the audio device, a difference circuit for determining the difference between the amplitude of the detected audio signals and a reference audio signal amplitude and for outputting a signal that represents this difference, a difference signal transfer circuit having an input for receiving the difference signal and an output wherein the difference signal is coupled to the output when the sensor circuit outputs a signal that indicates an audio signal has been detected, and a control circuit for generating a control signal that effects attenuation, augmentation or maintenance of the amplitude of the audio signals generated by the audio device in accordance with the difference signal when the sensor circuit detects an audio signal.

Abstract: The circuit for adaptive suppression of noise is a component part of a digital-hearing aid, consists of two microphones (1, 2), two AD—converters (3, 4), two compensating filters (5, 6), two retarding elements (7, 8), two subtractors (9, 10), a processing unit (11), a DA—converter (13), an earphone (15) as well as the two filters (17, 18). The method for adaptive suppression of noise can be implemented with the indicated circuit. The two microphones (1, 2), provide two differing electric signals (d1(t), d2(t)), which are digitalized in the two AD—converters (3, 4) and pre-processed together with the two fixed compensation filters (5, 6). Downstream the compensation filters are arranged the two filters (17, 18) symmetrically crosswise in a forward direction and having adaptive filter coefficients (w1, w2). The filter coefficients (w1, w2) are calculated by a stochastic gradient procedure and updated in real time while minimizing a quadratic cost function consisting of cross-correlation terms.

Abstract: An audio signal processing device is described for deriving auxiliary audio signals, such as audio direction sensing signals or a center audio signal from first and second audio signals through first and second filter paths, each of which comprises a first adaptive filter, and a first summing means is provided for coupled to the first adaptive filters for providing a summed audio signal at its summing output. Each filter path further comprises a second adaptive filter coupled to said summing output, whose respective adaptive filter coefficients are transferred to the first adaptive filters and are adapted in response to respective comparisons of the first and second audio signals with filtered sums of the first and second audio signals. Therewith correlated and uncorrelated parts of the input audio signals are processed effectively.

Abstract: Two output signals (O1a and O1b) of a microphone system (1) depend in different manner on the angle of incidence (?) of acoustic signals and are divided one by the other (7). A mathematical product of the ratio (A7) and a weighting factor (?) is saturated (12) and subtracted from a signal value (A) which can be fed into the system. The subtraction remainder is multiplied (13) by that output signal from the microphone system (1) which also generates the denominator signal for the division (7). Depending on the weighting factor (a) of the saturation value (B) and on the subtraction value (A), a desired directional characteristic is implemented between the resultant signal (Sout) of the said multiplication and the angle of incidence (?) of acoustic signals impacting the microphone system (1).

Abstract: An active noise control system is provided to specify an input transducer, an error transducer, an output transducer and an active noise controller for generating an anti-phase canceling acoustic signal to attenuate an input noise and to output a reduced noise. The active noise control system also performs on-line secondary path modeling. For this purpose the active noise control system comprises, in addition to known systems, a secondary path change detection circuitry (6), a signal distinguishing circuitry (7) and an auxiliary noise control circuitry (8), all of them being used to model said secondary path.

Abstract: A low-pass filter of over-sampling type oversamples an input digital audio signal T1 and filters and removes the low frequency components of the produced aliasing noise. A spectrum analyzer circuit calculates the spectrum intensity of a predetermined band of an output signal from the low-pass filter. An expanded signal generating circuit generates an expanded signal having frequency components of the output signal from the low-pass filter. A level control circuit controls the level of the expanded signal according to the spectrum analyzer circuit. An adder adds the level-controlled expanded signal to the output signal from the low-pass filter thereby to generate a digital audio signal T2.

Abstract: An impulse noise reducer detects impulse noise in an audio signal by detecting and smoothing the high-frequency amplitude of the audio signal, attenuating the non-smoothed amplitude according to the smoothed amplitude, and comparing the attenuated amplitude with a threshold. Impulse noise is discriminated from high-frequency audio components because the latter tend to occur in longer-lasting bursts and are therefore attenuated more strongly. The impulse noise reducer is simplified because it does not have to perform intermediate-frequency signal processing, and its sensitivity is not affected by adjacent-channel signals because these signals are substantially absent from the audio signal. The impulse noise reducer can be implemented by digital signal processing, and is suitable for use in a medium-wave AM audio broadcast receiver.

Abstract: In a method for adjusting the phase of a sound signal from multiple speaker units connected to a computer (e.g. a PC), a microphone is installed at a user's listening position, a reference sound is generated from a single particular speaker unit based on a reference sound data having a certain waveform, a phase difference between the reference sound data and a sound detected by the microphone is calculated for each of the speakers, an average value qF of the phase differences for front speakers and an average value qR of the phase differences for rear speakers are respectively calculated, and if |qF−qR|≧90°, phases of the sound data output to the rear speakers are inverted.