Abstract: A dynamic range compression system is provided, using either a sample-by-sample or a block processing system. Such a system can be used, for example, in a hearing aid. The system, using a frequency-warped processing system, is comprised of a cascade of all-pass filters with the outputs of the all-pass filters providing the input to the frequency analysis used to compute the filter coefficients. The compression filter is then designed in the frequency domain. Using a compression filter having even symmetry guarantees that the group delay is constant and does not depend on the compression gains at any given time. Additionally, due to the use of all-pass filters, the compression filter group delay more closely matches human auditory latency. An inverse frequency transform back into the warped time domain is used to produce the compression filter coefficients that are convolved with the outputs of the all-pass delay line to give the processed output signal.

Abstract: A dynamic range compression system is provided, using either a sample-by-sample or a block processing system. Such a system can be used, for example, in a hearing aid. The system, using a frequency-warped processing system, is comprised of a cascade of all-pass filters with the outputs of the all-pass filters providing the input to the frequency analysis used to compute the filter coefficients. The compression filter is then designed in the frequency domain. Using a compression filter having even symmetry guarantees that the group delay is constant and does not depend on the compression gains at any given time. Additionally, due to the use of all-pass filters, the compression filter group delay more closely matches human auditory latency. An inverse frequency transform back into the warped time domain is used to produce the compression filter coefficients that are convolved with the outputs of the all-pass delay line to give the processed output signal.

Abstract: The present invention, generally speaking, picks up a voice or other sound signal of interest and creates a higher voice-to-background-noise ratio in the output signal so that a user enjoys higher intelligibility of the voice signal. In particular, beamforming techniques are used to provide optimized signals to the user for further increasing the understanding of speech in noisy environments and for reducing user listening fatigue. In one embodiment, signal-to-noise performance is optimized even if some of the binaural cues are sacrificed. In this embodiment, an optimum mix ratio or weighting ratio is determined in accordance with the ratio of noise power in the binaural signals. Enhancement circuitry is easily implemented in either analog or digital form and is compatible with existing sound processing methods, e.g., noise reduction algorithms and compression/expansion processing. The sound enhancement approach is compatible with, and additive to, any microphone directionality or noise canceling technology.

Abstract: A second-order microphone system is constructed of two null-less first-order microphone elements. The null-less first-order microphone elements prevent the degradations that occur in performance when a second-order microphone system is used at the side of a person's head.

Abstract: A highly discreet and accurate voice pickup device includes a standard miniature pressure gradient type microphone element provided and mounted very close to the side of the user's head, preferably near the user's ear. The microphone element is oriented so that its direction of maximum sensitivity is parallel to the side of the user's head, and points toward the user's mouth so that it will pick up the user's speech sounds as the sounds diffract and travel along the side of the user's face and head, to the microphone element. The microphone element can be configured with low pass networks at front and rear ports of the microphone element, which cooperate with air volumes within the microphone to provide additional directional properties that advantageously vary with frequency and further enhance the noise-rejection capability of the microphone element.

Abstract: An adaptive binaural beamforming system is provided which can be used, for example, in a hearing aid. The system uses more than two input signals, and preferably four input signals. The signals can be provided, for example, by two microphone pairs, one pair of microphones located in a user's left ear and the second pair of microphones located in the user's right ear. The system is preferably arranged such that each pair of microphones utilizes an end-fire configuration with the two pairs of microphones being combined in a broadside configuration. Signal processing is divided into two stages. In the first stage, the outputs from the two microphone pairs are processed utilizing an end-fire array processing scheme, this stage providing the benefits of spatial processing. In the second stage, the outputs from the two end-fire arrays are processed utilizing a broadside configuration, this stage providing further spatial processing benefits along with the benefits of binaural processing.

Abstract: The hearing aid system includes a hearing aid case which is designed to be positioned behind the ear of a user, a tube for conducting sound from the case to the ear canal of the user, and a eartip for anchoring the end of the tube within the ear canal of the user. The tube is formed in a preformed shape of a material with sufficient rigidity to support the hearing aid in a proper position on the user's ear. The eartip may be one of several different designs which secures an end of the tube in place in the ear canal in a comfortable manner without the need for an expensive custom made ear mold and without complete occlusion of the ear canal. A combination of the light weight and small size of the case, the rigidity of the preformed tube, and one of the eartips for anchoring the tube in the ear canal allow the system to be held in place securely without the need for either an inflexible and aesthetically unpleasing ear hook or an expensive custom made ear mold.

Abstract: A highly discreet and accurate voice pickup device includes a standard miniature pressure gradient type microphone element provided and mounted very close to the side of the user's head, preferably near the user's ear. The microphone element is oriented so that its direction of maximum sensitivity is parallel to the side of the user's head, and points toward the user's mouth so that it will pick up the user's speech sounds as the sounds diffract and travel along the side of the user's face and head, to the microphone element. The microphone element can be configured with low pass networks at front and rear ports of the microphone element, which cooperate with air volumes within the microphone to provide additional directional properties that advantageously vary with frequency and further enhance the noise-rejection capability of the microphone element.

Abstract: The present invention is directed to a communication system wherein bone conducted ultrasonic signals are used as carriers to efficiently produce high fidelity, wide audio bandwidth sound. Exemplary embodiments rely on the use of modulation techniques which can achieve high fidelity audible sound, such as carrier plus single sideband (SSB) modulation. Known non-linearities within the ear itself can be exploited to demodulate the modulated ultrasonic carrier without producing audible sounds at the input to the user's ear. The non-linearities of the ear itself, in conjunction with the human brain's perception of audible frequencies generated in response to modulated ultrasonic stimulation, are relied upon to detect audio information.

Abstract: The present invention is directed to a communication system wherein ultrasonic signals can be used as carriers to efficiently produce high fidelity, wide audio bandwidth sound. Exemplary embodiments rely on the airborne transport of an inaudible ultrasonic carrier directly into the ear canal of a user, such that the non-linearities within the ear itself can be exploited to demodulate the ultrasonic carrier without producing audible sounds at the input to the user's ear. The non-linearities of the ear itself, in conjunction with the human brain's perception of audible frequencies generated in response to ultrasonic stimulation, are relied upon to detect audio information. The ultrasound-to-audio-sound conversion in the confined volume of the inner ear appears to be constant pressure, as opposed to constant velocity, such that all frequencies of the audio bandwidth (including low frequency bass signals) are produced with comparable sound intensity.

Abstract: An apparatus and method are provided for monitoring magnetic hearing systems by receiving a magnetic hearing system in an acoustic hearing aid testing device and then detecting the magnetic field output by the magnetic hearing system with a magnetic-to-acoustic testing device when the magnetic hearing system is being tested by the acoustic hearing aid testing device. The magnetic-to-acoustic testing device then develops an acoustic output signal representative of the detected magnetic field which may then be used in traditional acoustic monitoring techniques, such as performing a listening check of the magnetic hearing aid system or performing standard ANSI types of measurements, by the audiologist or tester.

Abstract: A dynamic range compression system is provided, using either a sample-by-sample or a block processing system. Such a system can be used, for example, in a hearing aid. The system, using a frequency-warped processing system, is comprised of a cascade of all-pass filters with the outputs of the all-pass filters providing the input to the frequency analysis used to compute the filter coefficients. The compression filter is then designed in the frequency domain. Using a compression filter having even symmetry guarantees that the group delay is constant and does not depend on the compression gains at any given time. Additionally, due to the use of all-pass filters, the compression filter group delay more closely matches human auditory latency. An inverse frequency transform back into the warped time domain is used to produce the compression filter coefficients that are convolved with the outputs of the all-pass delay line to give the processed output signal.

Abstract: A frequency-warped processing system using either sample-by-sample or block processing is provided. Such a system can be used, for example, in a hearing aid to increase the dynamic-range contrast in the speech spectrum, thus improving ease of listening and possibly speech intelligibility. The processing system is comprised of a cascade of all-pass filters that provide the frequency warping. The power spectrum is computed from the warped sequence and then compression gains are computed from the warped power spectrum for the auditory analysis bands. Spectral enhancement gains are also computed in the warped sequence allowing a net compression-plus-enhancement gain function to be produced. The gain versus frequency function is a set of pure real numbers, so the inverse frequency domain transform gives a set of time-domain filter coefficients. The speech segment is convolved with the enhancement filter in the warped time-domain to give the processed output signal.

Abstract: A communications earpiece comprises an ear canal tube sized for positioning in an ear canal of a user so that the ear canal is at least partially open for directly receiving ambient sounds. A microphone port in the ear canal tube is located in the ear canal for detecting sounds in the ear canal. A speaker port in the ear canal tube is located in the ear canal in close proximity to the microphone port for broadcasting sound signals into the ear canal. A sound processor amplifies the ambient sounds received by the microphone to produce processed analog signals. The sound processor also comprises noise cancellation means for producing an inverse noise signal of noise detected in the ear canal by the microphone. The inverse noise signal is sent to the speaker port so as to be broadcast into the ear canal thereby substantially canceling the ambient noise in the ear canal.