Abstract: The estimation of an HRTF for a given individual is accomplished by means of a coupled model, which identifies the dependencies between one or more images of readily observable characteristics of an individual, and the HRTF that is applicable to that individual. Since the HRTF is highly influenced by the shape of the listener's outer ear, as well as the shape of the listener's head, images of a listener which provides this type of information are preferably applied as an input to the coupled model. In addition, dimensional measurements of the listener can be applied to the model. In return, the model provides an estimate of the HRTF for the observed characteristics of the listener.

Abstract: The apparatus is a cross fader with dual optocouplers which communicate first and second audio signals. A manually controlled blade shutter travels along a linear path wherein at a first end of the path, a first edge of the blade shutter is inserted into the first optocoupler thereby attenuating or cutting the first audio signal. Likewise, at a second end of the path, a second edge of the blade shutter is inserted into the second optocoupler thereby attenuating or cutting the second audio signal.

Abstract: A system for programming hearing aids utilizing a multi-electrode programming strip. The strip is removably attached to a connector and inserted into the aid through a narrow slot in the faceplate; the slot is bounded by the faceplate and by one of the edges of the battery door. When the door is closed, the strip (and the electrodes thereon) is urged against corresponding terminals located inside the aid.

Abstract: According to the invention, a circuit arrangement for gaining a 38 kHz stereo subcarrier and a 57 kHz RDS carrier for decoding a stereo signal comprised in a demodulated FM signal and/or an RDS signal comprised in a demodulated FM signal, having a simple structure with only one PLL and particularly only one voltage-controlled oscillator (2) is characterized in that the arrangement comprises a phase-locked loop with a loop filter (1), a voltage-controlled oscillator (2), a first phase detector (3) which receives a reference signal having a reference frequency, and a second phase detector (4), which receives the FM signal, the output signal of the voltage-controlled oscillator (2) being coupled to both phase detectors (3, 4) in a form divided down by means of dividers (6, 9; 10, 11), and the signal fed back to the second phase detector having a frequency of 19 kHz, in that dividers (9, 10, 12) are provided, by means of which the output signal of the voltage-controlled oscillator (2) is divided down and which su

Abstract: The apparatus is a cross fader with a slide assembly which slidingly traverses a path defined by two parallel guide rods which pass through a partially open channels in the slider of the slide assembly. Tension plates are secured by self-tapping screws within recesses laterally adjacent to open portions of at least one of the channels. The tension plates are urged radially against the guide plates, so that the tighter the screws are secured, the tighter the tension plates are urged against the guide rods and the more frictional resistance is generated when the user moves the slider.

Abstract: An adapter for reducing echo and noise in internet telephony for use with personal computers. The device employs analog circuitry to provide signal filters and delay which act together to reduce line induced echo and background noise, thus providing a clear voice communication channel. The device is readily inserted between standard microphones and speakers employed in the personal computer.

Abstract: A virtual device control module for interfacing with a target device is disclosed. A DCM (device control module) is operable for interfacing with the target device, wherein the DCM includes a first FCM (function control module) and a second FCM operable for controlling respective first and second functional components of the target device. The first FCM can be a physical FCM operable for controlling the operation of the first functional component, and the second FCM can be a virtual FCM operable for controlling the operation of a second functional component.

Abstract: Computer-implemented techniques are provided for synthesizing sounds of an internal combustion engine vehicle using a physical model of the vehicle. In general terms, the method includes independently generating and/or synthesizing separate components of the vehicle sound, then combining these components to produce a final sound. Using a physical model of the vehicle, the separate components of the vehicle sound are independently generated from vehicle control parameters characterizing the operating conditions of the vehicle. The components are then combined using mixers and equalizers to produce a realistic vehicle sound. The present technique allows independent control of the separate components of the vehicle sound, is not limited to specific vehicles, and does not require recorded sounds taking large amounts of storage space.

Abstract: In a stereophonic expansion circuit, the (L+R) sum signal is spectrally modified by increasing the bass and treble frequencies relative to the midrange so as to compensate for a midrange frequency boost in the (L?R) difference signal. The stereophonic expansion effect and manipulation of the signal parameters are produced by active matrixing amplifiers.

Abstract: In accordance with an embodiment of the present invention, a dynamics processor includes a non-linear automatic gain control (AGC) responsive to an input audio signal comprised of a plurality of frequency components, each frequency component having associated therewith an amplitude, said non-linear AGC adaptive to develop a modified gain audio signal. A multiband cross-over device is responsive to the modified gain audio signal and is adaptive to generate ‘n’ number of signals, each of said ‘n’ signals having an amplitude and further having a unique frequency band associated therewith.

Abstract: A computer audio system includes an audio codec and a tone controller. The audio codec is operably coupled to receive audio information, which includes tone control settings, PCM digital audio inputs and PCM digital audio outputs. In addition, the audio codec may receive audio information as analog input signals via a line-in, a CD input, or an auxiliary input. Based on the audio information, the audio codec provides a first stereo output, a second stereo output and a monotone audio output. The tone controller is operably coupled to the audio codec and includes a low pass filter, a high pass filter, a band pass filter, and a summing module. The low pass filter is operably coupled to filter the monotone audio output and isolates bass components of the audio signal being processed. By further coupling a volume control module to the low pass filter, the bass component of the audio signal being processed may be varied.

Abstract: An audio processing apparatus has a converter for converting n-channel (positive integral number: n?1) audio signals input from at least one sound source into two-channel signals, a pair of correcting filters to which a pair of two-channel signals converted by converter are supplied and which correct a difference of the sense of hearing due to a difference of headphone characteristics, and an output section for supplying a pair of output signals from the pair of correcting filters to right and left speaker units of a headphone.

Abstract: In correcting the sound field, loudspeakers 6FL to 6WF are sounded by the noise. The attenuation factors of the inter-band attenuators ATF11 to ATFki for adjusting gains of the band-pass filters BPF11 to BPFki to the frequency in respective channels are corrected based on detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. The attenuation factors of channel-to-channel attenuators ATG1 to ATG5 are corrected based on the detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. The delay times of delay circuits DLY1 to DLY5 are corrected based on the detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. The attenuation factor of a channel-to-channel attenuator ATGk is corrected based on the detection result of the reproduced sound of the loudspeaker 6WF as the subwoofer. Therefore, the levels of the reproduced sounds reproduced by the loudspeakers 6FL to 6WF are adjusted to be made flat over the audio frequency band.

Abstract: An apparatus and method for recording and reproducing a sound (i.e. audio) signal corresponding to a video signal at a higher than normal speed. The method delimits a sound signal reproduced at a recording medium at a speed higher than a normal speed into successive processing unit periods. For each processing unit period, sound absence portion(s) of the reproduced sound signal are deleted (or partially deleted) within a range corresponding to a normal speed reproduction. Sound presence portions preceding and following the deleted absence portions are joined or compressed to produce a recognizable sound signal.

Abstract: A device for and method of calibrating a microphone, includes a loudspeaker (3) for converting a loudspeaker input signal (5) into sound; a microphone (4) for converting received sound into a microphone output signal (16), and a calibration arrangement for calibrating an output power of the microphone relative to a desired power level. The calibration arrangement includes an impulse response estimating device (7) for estimating an acoustic impulse response of the microphone by correlating the microphone output signal (6) and the loudspeaker input signal (5) when the microphone (4) receives the sound from the loudspeaker (3), whereby the output power of the microphone (4) is estimated.

Abstract: A smaller, less expensive, parametric audio amplifier system configured to be conformable to cable routing requirements that are no more stringent than that of conventional loudspeaker systems. The parametric audio amplifier system includes an amplifier assembly for amplifying an ultrasonic carrier signal modulated with a processed audio signal, an acoustic transducer assembly for projecting a sonic beam corresponding to the amplified ultrasonic signal through the air to regenerate the audio signal, and a low voltage connection for carrying the amplified ultrasonic signal from the amplifier assembly to the acoustic transducer assembly. The amplifier assembly includes an amplifier for receiving the modulated ultrasonic signal and generating an amplified ultrasonic signal having a low voltage level, and a voltage source for generating a voltage level.

Abstract: A surround-sound circuit has a delay time memory (31) for storing a digital audio input signal IN, and outputs the stored input signal as a surround-sound signal a predetermined time later. The input signal is compressed by a compressor (32) in the form of ADPCM encoder before it is stored in the memory (31), and the compressed signal retrieved from the memory is expanded by an expander (33) in the form of ADPCM decoder before it is added to the input signal. The invention may reduce appreciably the amount of data to be stored in the memory, and hence the storage capacity of the RAM, without directly loping the audio input signal IN and hence avoiding appreciable deterioration of the signal IN.

Abstract: A high pass filter 32 having a cut-off frequency of 200 Hz and a cut-off response of +12 dB/OCT and a low pass filter 33 having a cut-off frequency of 400 Hz and a cut-off response steeper than ?24 dB/OCT are used to pick out components in a double overtone region of 200˜400 Hz of a bass musical instrument such as a base or a bass drum from an input audio signal. The component which is picked out is input to distortion applying means 34 having an input-output response which is a non-linear response having no point symmetry with respect to 0 point, thus applying a distortion to cause even-numbered overtone components to be produced. An output from the distortion applying means 34 and the input audio signal are summed in a summer 18 to be delivered. Natural tones which have boosted bass tones and which are free from impurity are obtained.

Abstract: A system and method are disclosed which utilize robotic analysis of acoustic data captured during operation of a device to evaluate the operation of such device. An acoustic capture sensor is implemented within a device, such as a paper handling device, automobile, or any other device that generates detectable sound as a by-product of its operation. The acoustic capture sensor captures sound produced by the operation of the device and converts the captured sound from analog to digital. The captured sound is communicated to a processor-based device that is operable to process the sound to evaluate the operation of the device. The processor-based device may be communicatively coupled to a data storage device that has empirical sound data stored therein. In this manner, the processor-based device may compare sound captured by acoustic capture sensor(s) within a device with the stored empirical sound data to analyze the device's operation.

Abstract: A frequency modulation (FM) multiplex demodulation device is provided, in which the feeling of stereo is maintained excellent and the offensive noise to the ears can be reduced.