Abstract: An active noise attenuation system (10) comprises a speaker (26) in communication with a flow body (78). The speaker (26) emits an output for attenuating noise from the flow body (78). A control unit (31) communicates with the speaker (26) and controls the output. A housing (14) has one volume (18) acting as a liquid storage chamber and another volume (22) acting as a speaker chamber.

Abstract: A multi-channel filter (20) includes S inputs for receiving S input signals (x1 . . . xS) and at least S filters (14, 28) coupled to the S inputs for filtering the input signals (x1 . . . xS) or auxiliary signals (y1 . . . yS) derived therefrom. The multi-channel filter (20) further includes a signal combiner (16) for combining the filtered signals into an output signal (ê) and a further signal combiner (22) coupled in between at least part of the inputs and at least part of the filters (14, 28). The further signal combiner (22) is arranged for combining at least part of the input signals (x1 . . . xS) into at least one auxiliary signal (y1 . . . yS) The multi-channel filter (20) is asymmetric in that at least one of the filters (28) has a filter length which is substantially larger than the filter lengths of the other filters (14).

Abstract: A method and audio converter for generating further audio signals (u, ul, ur, uc, us) from initial audio signal (x, xl, xr), wherein optionally an information signal (in means 23) is derived from said initial audio signals (x). On basis of the initial audio signal (x, xr, xl), a dominant signal y(k) and a residue signal (or signals) q(k), substantially transverse to each other, are determined (means 21 and 22). In at least two frequency ranges frequency components of the dominant signal are analysed (means 24), and a difference signal yr ({y(k)?yb(k)) corresponding to the dominant signal minus a frequency range component of the dominant signal in one or more frequency ranges (yb(k)) is formed. The difference audio signal yr and the residue signal q(k) are transformed into said further audio signal u(means 25), i.e. u = T ? ( y r ? ( k ) q ? ( k ) ) .

Abstract: A musical garment has a plurality of interior pockets disposed on two interior portions of the garment for generating extemporaneous music by hand-impact with a selective pocket portion. Each interior pocket is secured via hook and loop fasteners and includes a sound generating piezoelectric transducer electrically connected thereto for producing multiple signals to a sound generating module. The piezoelectric transducers are sensitized or calibrated to produce a sound having a different sound characteristic compared with other piezoelectric transducers. The sound generating module is removably housed within an interior auxiliary pocket of the garment and utilizes a multiplexer to provide serial output or music for subsequent recording or listening. An acoustic screen is disposed on an exterior surface portion of the garment pocket for acoustically transmitting sound generated from the module.

Abstract: An adaptive resonance canceller system and method for attenuating narrowband noise signals within an input signal, where the narrowband noise signals may vary significantly in frequency. The sensor comprises a notch filter, an error reference and gradient generator, and a complex correlator circuit for attenuating the narrowband noise component of the input signal. The input signal is applied to the notch filter and to the error reference and gradient generator. The latter component generates an error reference signal and an error gradient signal. These two signals are applied to the complex correlator circuit which, in turn, generates a tuning parameter signal which is applied to the notch filter to tune the filter to the center frequency of the noise component of the input signal.

Abstract: An audio signal processing circuit reduces noise in an incoming audio signal and is particularly useful in telephone communication systems utilizing one or more hands-free microphones. A preferred embodiment of the audio signal processing circuit includes a pre-emphasis circuit receiving the audio signal from a microphone or other transducer, an amplifier circuit receiving the pre-emphasized audio signal from the pre-emphasis circuit and a de-emphasis circuit receiving the amplified signal from the amplifier circuit. An output of the de-emphasis circuit provides the processed audio signal having an improved signal to noise ratio with minimum audible distortion. A preferred embodiment of the amplification circuit includes an amplifier defining a non-linear transfer function therethrough which provides low gain to the lower amplitude noise signals and higher gain to the higher amplitude audio signals.

Abstract: The invention concerns a method for continuously controlling the quality of distributed digital sounds broadcast by radio or television on a digital channel. The method consists in temporally breaking down the digital signal into sequences of samples; carrying out a spectral analysis of each sequence to observe the variations in energy and envelope of the digital signal and calculating a global quality index; and in calculating on the basis of the global quality index, a final gated and continuous quality index representing the quality of the digital signals. The invention is applicable to the continuous control of the quality of distributed sounds.

Abstract: A signal processor for reducing undesirable signal content reduces the undesirable signal content by exaggerating the undesirable signal content and then using this exaggerated undesirable signal and adaptive filter means to estimate the undesirable content in the signal and then substantially removing it from the signal. The signal processor includes a signal mapping means for exaggerating the undesirable signal content; and an adaptive filter means for reducing the undesirable signal content using the exaggerated undesirable signal content.

Abstract: The disclosure relates to localization of an acoustic image out of the head in hearing a reproduced sound via headphone, and includes the steps of: with audio signals S1-S11 of left, right channels reproduced by an appropriate audio appliances as input signals, branching the input signals of the left and right channels to at least two systems; to form signals of each systems corresponding to the left, right channels with left, right speaker sounds imagined in an appropriate sound space with respect to the head of a listener wearing a headphone Hp and virtual reflected sound in the virtual sound space SS caused from a sound generated from the left and the right virtual speaker SPL, SPR, creating a virtual speaker sound signal by processing so that the virtual speaker sounds from the left and the right speakers are expressed by direct sound signals, and virtual reflected sound signals by processing so that the virtual reflected sound is expressed by reflected sound signals; mixing the direct sound signal and re

Abstract: A key click masker and method for disguising key clicks protect sensitive information by randomly varying key click echo generated by a computer system. The key click masker and method disguise the actual keystrokes by generating sounds that would confuse a listener trying to determine the actual keystrokes entered by a user while the user is inputting sensitive information on an input device.

Abstract: A digital hearing aid according to the present invention is capable of measuring its own performance. The hearing aid includes a test signal generator for feeding a test signal into the hearing aid amplifier. The response to the test signal is acquired at a specific point in the hearing aid, depending upon what aspect of performance is to be measured. Various elements of the hearing aid and/or the hearing aid feedback may be bypassed. The hearing aid further includes the capability of initializing hearing aid parameters based upon the performance measurements. The measurement and initialization capability may be entirely integral to the hearing aid, or an external processor may be used to download the measurement program and the run time program, and assist in computing the parameters.

Abstract: The invention relates to an arrangement for improving leak tolerance in an earpiece. The invention can be applied preferably in teleterminals, particularly in mobile stations. One idea of the invention is that an acoustic return path (314, 313, 311, 312, 320) is directed from the back part of the earpiece capsule (301) to a volume (307) between the earpiece (300) and the user's ear. By means of the solution according to the invention an optimum, controlled load is achieved particularly for low frequencies, such that a change in the volume (307) between the earpiece and the ear only has a minor effect on the volume and quality of the sensed sound. By means of the solution a good leak tolerance is achieved, even though the volume to be arranged behind the earpiece capsule is small.

Abstract: N correlated signals are processed by N pre-filters whose transfer characteristics have different zero points, then the processed signals are input into an N-input M-output linear FIR system, and its transfer characteristics are estimated from its response outputs and the processed signals from the pre-filters.

Abstract: The air induction system comprises an air induction body with a mouth and a speaker forming an interface with the mouth. A fender portion of a vehicle at least partially covers the interface, protecting the interface from environmental conditions. A control unit controls with the speaker to thereby generate a noise attenuating sound.

Abstract: A method of converting between a sampling rate associated with a first audio format and a second audio format includes up-sampling an input signal sampled at the sample rate associated with the first audio format. Then, the up-sampled signal is filtered as a function of a fractional delay to generate an output signal sampled at the sample rate associated with the second audio format. The fractional delay is computed from the sample rates associated with the first and second audio formats. In one embodiment, the sample rates that are converted between are associated with a compact disc format having a sample rate of about 44.1 kHz and a digital audio tape format having a sample rate of about 48 kHz. In such case, the input samples are preferably up-sampled by a factor of two and the samples are then preferably filtered in accordance with a third order six taps coefficient finite impulse response filtering technique.

Abstract: A modular audio loudspeaker array system. The system includes a first speaker module, and a second speaker module, constructed and arranged to radiate sound waves responsive to audio signals. Sound waves radiated by the first speaker module and the sound waves radiated by the second speaker module combine to form a radiation pattern. The system further includes a positioning system, for positioning the first bass module relative to the second bass module at a plurality of fixed predetermined distances; an audio signal input terminal electronically for coupling the loudspeaker array to a source of the audio signals for receiving the audio signals; an audio signal processing device, electronically coupling the audio signal input terminal to the first bass module and to the second bass module, for processing the audio signals and transmitting the audio signals to the first speaker module and to the second loudspeaker module.

Abstract: A microphone array system includes a plurality of microphones and a sound signal processing part. The microphones are arranged in such a manner that at least three microphones are arranged in a first direction to form a microphone row, at least three rows of the microphones are arranged so that the microphone rows are not crossed each other so as to form a plane, and at least three layers of the planes are arranged three-dimensionally so that the planes are not crossed each other, so that the boundary conditions for the sound estimation at each plane of the planes constituting the three dimension can be obtained.

Abstract: A surround reproducing circuit includes a first adder for generating a difference signal of an L signal and an R signal which are inputted, a low-pass filter connected to an output side of the first adder, and second and third adders for mixing an output signal of the low-pass filter as a surround signal with the L signal and the R signal in an opposite phase relationship to each other. Change in a phase is lessened within a frequency band of 20 Hz to 20 KHz and the localization can become definite, and furthermore, a harsh high pass is also lessened. Consequently, it is possible to realize a surround effect having natural spread. Furthermore, the number of capacitors to be required can also be decreased.

Abstract: A method is described for recording an optical information carrier, in which a mark representing recorded data is written in the information carrier at different writing speeds by a sequence of radiation pulses The last pulse of a sequence is followed by a period of reduced radiation power, the duration of which is inversely proportional to the writing speed.

Abstract: Some workers wear headsets to protect their hearing from loud persistent noises, such as airplane engines and construction equipment. These headsets are generally passive or active, with the active ones including ear speakers and automatic noise-reduction (ANR) circuitry to cancel or suppress certain types of loud persistent noises. One problem with active headsets the concerns the difficulty of salvaging headsets that fail performance testing. Accordingly, the inventor devised a unique a unique cup-in-cup structure for the earcups of active headsets as well as related assembly and testing methods. The unique structure not only allows for pretesting of the ANR circuitry prior to assembly, but also enhances performance of the resulting headsets.