Abstract: The invention provides a phase calibration module, calibrating phase mismatch between microphone signals output by a plurality of microphones of an array microphone. In one embodiment, the phase calibration module comprises a subband filter, a delay calculation module, and a delay compensation filter. The subband filter extracts a high frequency component and a low frequency component from each of the microphone signals to obtain a plurality of high-frequency component signals and a plurality of low-frequency component signals. The delay calculation module calculates delays between the low-frequency component signals. The delay compensation filter then compensates the low-frequency component signals for phase mismatches therebetween according to the calculated delays to obtain a plurality of calibrated low-frequency component signals.

Abstract: There is provided a method for providing hearing assistance to a user (101, 301), comprising: capturing audio signals by a microphone arrangement (26) comprising at least two spaced apart microphones (M1, M2); estimating the total energy contained in the voice spectrum of the audio signals captured at least one of the microphones; estimating the value of the direction of arrival of the captured audio signals by comparing the audio signals captured by at least two of the spaced apart microphones; judging whether a voice is present close to microphone arrangement by taking into account the estimated total energy contained in the voice spectrum of the captured audio signals and the estimated value of the direction of arrival of the captured audio signals; outputting a signal representative of said judgement; processing said captured audio signals according to said signal representative of said judgement; and stimulating the user's hearing, by stimulating means worn at or in at least one of the user's ears (39),

Abstract: The directional microphone device according to the present invention solves a problem of increase in thermal noise (problem of decrease in sensitivity) that occurs at the time of directivity synthesis. The directional microphone device includes: a plurality of microphones which have directional and non-directional characteristics; a control unit which generates an output signal using signals outputted from each of the plurality of microphones; and an output unit which outputs the output signal generated by the control unit. The control unit generates the output signal such that a nearly non-directional directivity and a high sensitivity are obtained in small amplitude range of the output signal, and a directivity and a low sensitivity are obtained in large amplitude range of the output signal.

Abstract: At least one exemplary embodiment is directed to a new spatial audio enhancing system including a novel Adaptive Sound Upmixing System (ASUS). In some specific embodiments the ASUS provided converts a two-channel recording into an audio signal including four channels that can be played over four different loudspeakers. In other specific embodiments the ASUS provided converts a two-channel recording into an audio signal including five channels that can be played over five different loudspeakersn even other specific embodiments the ASUS provided converts a five-channel recording (such as those for DVD's) into an audio signal including eight channels that can be played over eight different loudspeakers. More generally, in view of this disclosure those skilled in the art will be able to adapt the ASUS to process and provide an arbitrary number of audio channels both at the input and the output.

Abstract: The invention relates to a method for operating a hearing aid. A local source operating mode is established by a signal processing section of the hearing aid for tracking and selecting a local acoustic source of an ambient sound. Electrical acoustic signals from which the local acoustic source is determined by the signal processing section are generated by the hearing aid from the detected ambient sound. The local acoustic source is selectively taken into account by the signal processing section in an output sound of the hearing aid such that the local acoustic source is at least acoustically prominent and is therefore better perceived compared to another acoustic source for a hearing aid wearer.

Abstract: The invention relates to adaptive directional systems, and more particularly to a method and apparatus for producing adaptive directional signals. The invention may be applied to the provision of audio frequency adaptive directional microphone systems for devices such as hearing aids and mobile telephones. The method involves constructing the adaptive directional signal (46) from a weighted sum of a first signal (42A) having an omni-directional polar pattern and a second signal (42B) having a bi-directional polar pattern, wherein the weights are calculated to give the combined signal a constant gain in a predetermined direction and to minimize the power of the combined signal. The method has particular application in producing signals in digital hearing aids, the predetermined direction being in the forward direction with respect to the wearer.

Abstract: This document provides a hearing assistance device for playing processed sound inside a wearer's ear canal, the hearing assistance device comprising a first housing, signal processing electronics disposed at least partially within the first housing, a first microphone connected to the first housing, the first microphone adapted for reception of sound, a second microphone configured to receive sound from inside the wearer's ear canal when the hearing assistance device is worn and in use and microphone mixing electronics in communication with the signal processing electronics and in communication with the first microphone and the second microphone, the microphone mixing electronics adapted to combine low frequency information from the first microphone and high frequency information from the second microphone to produce a composite audio signal.

Abstract: The invention relates to a method for a hearing aid wearer to actively operate a hearing aid. A signal processing section of the hearing aid has a demixing module for separating audio signals and a postprocessor module which sets up a hold mode of operation for the hearing aid. An audio signal, which is preferred by the hearing aid wearer, from an ambient sound, is tracked and selected by virtue of the hearing aid wearer transmitting to the hearing aid a command which sets up the hold mode of operation in the signal processing section of the hearing aid for a certain period. The signal processing section tracks the preferred audio signal and selectively takes account of it in an output sound from the hearing aid such that it is audibly highlighted for the hearing aid wearer in comparison with another audio signal and is thereby perceived better.

Abstract: Method for processing an input signal in a hearing aid, with the input signal being broken down into a discrete signal for each source relative to an acoustic signal, with the discrete signals being assigned to a spatial position of the source and with the discrete signals being output, or output attenuated, relative to the spatial position.

Abstract: A method for improving compatibility of a hearing aid with an antenna, in which at least a metal frame is provided near by a grounding surface of the antenna to change the direction of radiation of the antenna, thereby to enhance the directivity of the antenna on the side away from a hearing aid, and to reduce the quantity of radiation proceeding toward the hearing aid and to improve the near-field quantity (for about 3 dB) of an electric field of an HAC tested plane. This method can further increase the heights of the metal frames to reduce the near-field quantity of the electric field of HAC (hearing aid compatibility) tested plane.

Abstract: A portable electronic device and an auxiliary device each having an interface allowing the establishment of a communication link between them, at least to be able to transmit data representative of a status information from the auxiliary device to the portable electronic device, and-a method of of establishing a measure of an absolute elapsed time in a listening device. The problem is solved in that the portable electronic device has a timing unit for determining a time interval and a memory for storing data, and wherein the auxiliary device has a master timing unit for providing a signal representative of the present time, and wherein the system is adapted to transfer a signal representative of the present time from the auxiliary device to the portable electronic device and to store it in the memory.

Abstract: A composite sound dampening structure includes a first base layer of sound dampening material extending around and against an inside surface of a container and a second wedge layer of sound dampening material attached to an inside surface of the first base layer. The composite sound dampening structure provides improved acoustic dampening in relative small sound chambers. An audio test system generates a composite audio signal of multiple different audio signals that are combined together using linear superposition. The composite audio signal allows a device to be simultaneously tested with multiple different audio frequencies.

Abstract: The invention relates to a method of operating an audio processing device. The invention further relates to an audio processing device, to a software program and to a medium having instructions stored thereon. The object of the present invention is to provide improvements in the processing of sounds in listening devices. The problem is solved by a method comprising a) receiving an electric input signal representing an audio signal; b) providing an event-control parameter indicative of changes related to the electric input signal and for controlling the processing of the electric input signal; c) storing a representation of the electric input signal or a part thereof; d) providing a processed electric output signal with a configurable delay based on the stored representation of the electric input signal or a part thereof and controlled by the event-control parameter. The invention may e.g. be used in hearing instruments, headphones or headsets or active ear plugs.

Abstract: A sound zoom method, medium, and apparatus generating a signal in which a target sound is removed from sound signals input to a microphone-array by adjusting a null width that restricts a directivity sensitivity of the microphone array, and extracting a signal corresponding to the target sound from the sound signals by using the generated signal. Thus, a sound located at a predetermined position away from the microphone array can be selectively obtained so that a target sound is efficiently obtained.

Abstract: A hearing aid device has a directional microphone system with a first microphone outputting a first microphone signal and a second microphone outputting a second microphone signal. A delay unit generates a directivity by delaying the second microphone signal or a fourth microphone signal derived therefrom by an internal time delay and associating it with the first microphone signal or a third microphone signal derived therefrom for generating a directional microphone signal. A cross-correlation analysis unit receives the first or the third microphone signal and the second or the fourth microphone signal and determines a value of a cross correlation of the two microphone signals. A control unit adjusting the time delay depending on the value of the cross correlation of the two microphone signals. A classifier determines audio conditions in which the hearing aid device is currently situated, and the time delay is adjusted depending on the audio conditions.

Abstract: A method for operating a hearing device, an associated microphone system comprising at least two omnidirectional microphones, and a hearing device are provided. The microphones emit microphone signals and are electrically interconnected with one another in order to form directional characteristics. A damping of the upper frequency range of the microphone signals is determined from the lower frequency range of the microphone signals. The impression of a wide-band directional microphone is produced as a result.

Abstract: For enabling fast, customer-specific and precise matching of a hearing device's directional characteristic, a hearing device having a signal processing unit for performing a processing algorithm is provided, with at least one design-related parameter of the hearing device having been made available to the signal processing unit and with the signal processing unit performing the processing algorithm based on the design-related parameter of the hearing device. It should be considered as especially advantageous in the case of the inventive hearing device that a processing algorithm can be performed particularly precisely and customer-specifically based on the provided design-related parameters of the hearing device.

Abstract: This invention relates to a system (200) for determining directionality of a sound. The system (200) comprises a first audio device (202) placed on one side of a user's head (100) and having a first microphone unit (110, 112) for converting said sound to a first electric signal, a second audio device (204) placed on the other side of the user's head (100) and having a second microphone unit (114, 116) for converting said sound to a second electric signal, and comprises a transceiver unit (220, 238) for interconnecting the first and second audio device and communicating the second electric signal to the first audio device (202). The first audio device (202) further comprises a first comparator (222) for comparing the first and second electric signals and generating a first directionality signal from the comparison.

Abstract: A sound source localization system includes a plurality of microphones for receiving a signal as an input from a sound source; a time-difference extraction unit for decomposing the signal inputted through the plurality of microphones into time, frequency and amplitude using a sparse coding and then extracting a sparse interaural time difference (SITD) inputted through the plurality of microphones for each frequency; and a sound source localization unit for localizing the sound source using the SITDs. A sound source localization method includes receiving a signal as an input from a sound source; decomposing the signal into time, frequency and amplitude using a sparse coding; extracting an SITD for each frequency; and localizing the sound source using the SITDs.

Abstract: Method and apparatus for microphone matching for wearable directional hearing assistance devices are provided. An embodiment includes a method for matching at least a first microphone to a second microphone, using a user's voice from the user's mouth. The user's voice is processed as received by at least one microphone to determine a frequency profile associated with voice of the user. Intervals are detected where the user is speaking using the frequency profile. Variations in microphone reception between the first microphone and the second microphone are adaptively canceled during the intervals and when the first microphone and second microphone are in relatively constant spatial position with respect to the user's mouth.

Abstract: A Hearing apparatuses is provided. The hearing apparatus includes a programming socket that features at least a first connection and a second connection, with a signal processing system that is connected to the second connection, and with a control system. The control system taps the potential of the first connection. The control system furthermore engages the signal processing system in a first function when the first connection reaches the predetermined potential and in a second function when the potential of the first connection deviates from the predetermined potential. This enables multiple usages for example of the input of a programming interface of a signal processing IC.

Abstract: The invention relates to an assembly comprising a sound emitter and at least two sound detectors fixed to each other, wherein each detector has a sound receiving opening. The sound receiving openings of at least two of the detectors point in opposite directions.

Abstract: An earpiece (100) is provided. The earpiece can include an Ambient Sound Microphone (111) configured to capture ambient sound, an Ear Canal Microphone (123) configured to capture internal sound in the ear canal, a memory (208) configured to record at least a portion of the history of the ambient sound and the internal sound, and a processor (121) configured to save a recent portion of the history responsive to an event.

Abstract: Method and apparatus for automatic reception enhancement of hearing assistance devices. The present subject matter relates to methods and apparatus for automatic reception enhancement in hearing assistance devices. It provides a power estimation scheme that is reliable against both steady and transient input. It provides a TSM estimation scheme that is effective and efficient both in terms of storage size and computational efficiency. The embodiments employing a decision tree provide a weight factor between the omnidirectional and compensated directional signal. The resulting decision logic improves speech intelligibility when talking under noisy conditions. The decision logic also improves listening comfort when exposed to noise. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: The invention relates to a method for operating a hearing system comprising an input unit, an output unit and a transmission unit operationally interconnecting said input output units. Said transmission unit implements a transfer function describing, how audio signals generated by said input unit are processed in order to derive audio signals fed to said output unit, and can be adjusted by one or more transfer function parameters. Said method comprises obtaining, by means of said input unit and with a first directional characteristic, first audio signals from incoming acoustic sound; deriving from said first audio signals a first set of sound-characterizing data; and deriving, in dependence of first directional information, which is data comprising information on said first directional characteristic, and of said first set of sound-characterizing data, a value for each of said one or more transfer function parameters.

Abstract: The invention relates to a method and an apparatus for the configuration of at least one adjustment option on a hearing device, with the hearing device wearer having the task of setting the hearing device by ear with the aid of the adjustment option to be configured such that a predetermined characteristic on the hearing device is adjusted, with the adjustment option being enabled if the characteristics set by the hearing device wearer equate to the predetermined characteristics or has deviations herefrom, which lie within a predetermined tolerance range.

Abstract: A hearing instrument includes a signal processor, and at least two microphones for reception of sound and conversion of the received sound into corresponding electrical sound signals that are input to the signal processor, wherein the signal processor is configured to process the electrical sound signals into a combined signal with a directivity pattern with at least one adaptive null direction ?, and wherein the signal processor is further configured to prevent the at least one null direction ? from entering a prohibited range of directions, wherein the prohibited range is a function of a parameter of the electrical sound signals.

Abstract: A system for adapting a hearing aid to a user includes a unit generating an image of the user's ear together with a model of the hearing aid, wherein the model of the hearing aid in-situ includes a physical model of the hearing aid inserted in the user's ear prior to generating the image. The system also generates a mesh based on the image, calculates acoustic response to a simulated acoustic signal in the mesh, calculates a frequency response curve for a fixed position in the mesh, and identifies an optimal position of a microphone of the hearing aid in-situ from the acoustic response. They system also includes a comparator configured to compare the frequency response curve for the fixed position in the mesh with a maximum frequency response curve for the mesh stored by the system.

Abstract: By using output signals from a plurality of omnidirectional microphones, respectively, the sound processing apparatus outputs a first directivity signal in which the main axis of directivity is formed in the direction of the utterer and outputs a second directivity signal in which the dead zone of directivity is formed in the direction of the utterer. The sound processing apparatus calculates the level of the first directivity signal and the level of the second directivity signal, and determines the distance to the utterer based on the level of the first directivity signal and the level of the second directivity signal. The sound processing apparatus derives a gain to be given to the first directivity signal according to the result of the determination and controls the level of the first directivity signal by using the gain.

Abstract: An apparatus for sound enhancement has at least two microphones (9) that provide a directional microphone array which is arranged to be pointed in the direction of a sound source. The directional microphone array thereby receives sound emitted by the sound source and generates sound signals. A processor (20) processors the sound signals generated by the microphone array to enhance the sound received by the directional microphone array from the sound source relative to other sound received by the directional microphone array. The processor (20) generates a corresponding enhanced signal (ES). Loud speakers (22) reproduce the enhanced signal as audible sound. Furthermore, sound suppression devices (7, 7a) are provided to suppress ambient should from reaching the eardrums of the user. This sound suppression acts in conjunction with the directional microphone array and the processor (20) which enhance the SOI to provide a listening environment in which the SOI is enhanced.

Abstract: A hearing aid that prevents the user from missing new speech when listening again to recorded speech. This hearing aid comprises a speech input section, a speech buffer, first and second direction adjustment sections, a hearing aid processor, and a speech output section. The speech input section inputs a speech signal. The speech buffer holds a speech signal inputted to the speech input section. The first and second direction adjustment sections subject a first speech signal inputted to the speech input section and/or a second speech signal held in the speech buffer to speech processing such that the first speech signal and the second speech signal are each perceived independently. The hearing aid processor subjects signals outputted from the first and second direction adjustment sections to hearing aid processing. The speech output section outputs a speech signal that has undergone hearing aid processing at the hearing aid processor.

Abstract: Disclosed is method of generating an audible signal in a hearing aid by estimating a weighting function of received audio signals, the hearing aid is adapted to be worn by a user; the method comprises the steps of: estimating a directional signal by estimating a weighted sum of two or more microphone signals from two or more microphones, where a first microphone of the two or more microphones is a front microphone, and where a second microphone of the two or more microphones is a rear microphone; estimating a direction-dependent time-frequency gain, and synthesizing an output signal; wherein estimating the direction-dependent time-frequency gain comprises: obtaining at least two directional signals each containing a time-frequency representation of a target signal and a noise signal; and where a first of the directional signals is defined as a front aiming signal, and where a second of the directional signals is defined as a rear aiming signal; using the time-frequency representation of the target signal an

Abstract: The selection of a plurality of sound sources for the purposes of preferential presentation should be structured to be more convenient for hearing instrument wearers. For this purpose, provision is made to separate the signals of the plurality of sources picked up and present them one after another with the aid of a corresponding control unit. On the basis of the presentation, the user can, likewise by means of the control units, select one of the plurality of signals for the purposes of further processing. The selection is preferably effected by pressing a button or by looking in the direction of the desired source plus pressing a button by way of acknowledgment.

Abstract: A method for processing at least one first and one second input signal in a hearing aid, with the input signals being filtered to create intermediate signals, the intermediate signals being added to form output signals, the input signals being assigned to a defined signal situation, and with the signals being filtered as a function of the assigned defined signal situation.

Abstract: A “speaker” operating mode is established by a signal processor of a hearing aid for tracking and selecting an acoustic speaker source in an ambient sound. Electric acoustic signals are generated by the hearing aid from the ambient sound that has been picked up, from which signals an electric speaker signal is selected by the signal processor by a database of speech profiles of preferred speakers. The electric speech signal is selectively taken into account in an output sound of the hearing aid in such a way that it will for the hearing-aid wearer acoustically at least be prominent compared with another acoustic source and consequently be better perceived by the hearing-aid wearer.

Abstract: A method for programming a hearing device is described, in which audio data and programming data is transmitted from a programming device to the hearing device, with the audio data and the programming data being converted into data packets and transmitted via a common channel of a digital radio connection from the programming device to the hearing device.

Abstract: To reliably and consistently detect desirable sounds, a system detects the presence of wind noise based on the power levels of audio signals. A first transducer detects sound originating from a first direction and a second transducer detects sound originating from a second direction. The power levels of the sound are compared. When the power level of the sound received from the second transducer is less than the power level of the sound received from the first transducer by a predetermined value, wind noise may be present. A signal processor may generate an output from one or a combination of the audio signals, based on a wind noise detection.

Abstract: An in-ear hearing aid device that provides users with a customizable ear piece to fit the ear canal of any user. More specifically, the present invention can comprises of a hard case to hold all the electronics of the device with a hearing device jacket that envelops the device. An ear mould is connected to the jacket and allows the user to insert the device into the ear canal to enhance the user's hearing.

Abstract: A device for generating a filtered activity pattern based on a first activity pattern at an auditory model of a first ear and a second activity pattern at an auditory model of a second ear includes an identifier for identifying a first trajectory in the first activity pattern and a second trajectory in the second activity pattern which are associated with the same sound event. The device further includes a determiner for determining whether the two trajectories are associated with a sound event of a useful sound source. The device further includes a filter for filtering the first activity pattern or the second activity pattern based on a result of determining whether a trajectory is associated with a sound event of the useful sound source.

Abstract: A hearing aid comprises two microphones, directional processing means for combining the respective audio signals to form a spatial signal, a beamformer for controlling the directional processing means to provide adaptation of the spatial signal, and means for boosting low frequencies of the spatial signal. A feedback estimator generates a feedback compensation signal, which is combined with the boosted spatial signal. By applying feedback compensation only after directional processing and low frequency boosting, the device avoids interference by the feedback estimator with the function of the beamformer. The invention also provides a method of processing signals in a hearing aid.

Abstract: Provided are a hearing aid, signal processing method, and program that enable TV sound to be made easier for a hearing aid user to hear when wishing to watch TV, and a person's voice to be made easier to hear when wishing to converse with that person. A hearing aid (100) is provided with an other-person's speech detection section (150) that detects speech of a speaker other than the wearer using detected sound source direction information, an own-speech detection result, and a TV sound detection result, and a per-sound-source frequency calculation section (160) that calculates the frequency of each sound source using an own-speech detection result, TV sound detection result, other-speaker's speech detection result, and sound source direction information. A scene determination section (170) determines a scene to be a “conversation scene,” “TV viewing scene,” and “‘TV viewing while . . .

Abstract: A conversation detection apparatus uses a head-mounted microphone array to accurately determine whether a speaker in front is a conversing person or not.

Abstract: This invention relates to a hearing instrument, which comprises a first microphone converting ambient sound to an ambient sound signal, a signal processor generating a processed sound signal based on the ambient sound signal, a controllable output stage generating a driving signal based on the processed sound signal and in accordance with a control signal, a speaker unit generating a sound in the ear canal based on said driving signal, a second microphone located in the ear canal of the user and converting the sound in the ear canal to the monitor sound signal, and a linearization stage comparing the processed sound signal and the monitor sound signal and generating the control signal based thereon.

Abstract: A hearing aid, that outputs a microphone input signal and an external input signal, improved includes a directional microphone, an external input terminal, a hearing aid processor that inputs sound signals from the microphone and the external input terminal, and a receiver that outputs sound signal that have undergone hearing aid processing by the hearing aid processor. The hearing aid processor has a mixer that mixes a sound signal from the microphone with a sound signal from the external input terminal and outputs a sound signal to the receiver, a mixing ratio decider that decides a mixing ratio between the sound signal from the microphone and a sound signal from the external input terminal, a front sound detector that is connected to the mixing ratio decider, and a similarity calculator that determines whether or not sound collected by the directional microphone is that of an external device.

Abstract: Hearing glasses with a left temple and a right temple connected to a front portion supporting a pair of lenses, at least one temple with one single microphone and a processor connected to the single microphone. The single microphone is an omnidirectional microphone and located in the temple such that, when the hearing glasses are worn by a human's head where the at least one temple is at a predetermined side of the head, the single microphone can receive sound substantially unblocked by the head from sound sources both at a left and right frontal side of the head, as well as from one back side of the head corresponding to the predetermined side of the head.

Abstract: An image reading apparatus according to an aspect of the invention includes a first reading unit which reads a surface of a document and a second reading unit which reads the other surface of the document. The first and second reading units include off-axial imaging units respectively. The off-axial imaging unit includes plural imaging mirrors in which off-axial reflecting surfaces having curvatures are formed. In the off-axial imaging unit, an incident direction and an outgoing direction of a reference axis light beam are different from each other. The off-axial imaging units are provided such that vertical directions of the first and second reading units become identical while an incident side of the light beam is located below an outgoing side with respect to the imaging unit.

Abstract: Disclosed are a hearing aid device and a hearing aid method capable of controlling a directivity process or an amplification process in a hearing aid with a user's hand placement so as to match a users intuition.

Abstract: In one aspect, a hearing device is provided with a multi-stage activation circuit including a primary activation unit, an evaluation unit and a main activation unit. The primary activation unit monitors a sound-induced input level and forwards incoming signals to the evaluation unit when a threshold value of the input level is exceeded. The evaluation unit is activated when the threshold value is exceeded and determines a feature of incoming signals suitable for identifying and/or classifying a sound causing the incoming signals. The main activation unit is activated, for example, when the feature determined by the evaluation unit indicates a noise that should be perceptible to the wearer of the hearing device. The main activation unit activates further components of the hearing device including the signal processing and amplification circuit and the output unit.

Abstract: A multi-channel digital hearing instrument is provided that includes a microphone, an analog-to-digital (A/D) converter, a sound processor, a digital-to-analog (D/A) converter and a speaker. The microphone receives an acoustical signal and generates an analog audio signal. The A/D converter converts the analog audio signal into a digital audio signal. The sound processor includes channel processing circuitry that filters the digital audio signal into a plurality of frequency band-limited audio signals and that provides an automatic gain control function that permits quieter sounds to be amplified at a higher gain than louder sounds and may be configured to the dynamic hearing range of a particular hearing instrument user. The D/A converter converts the output from the sound processor into an analog audio output signal. The speaker converts the analog audio output signal into an acoustical output signal that is directed into the ear canal of the hearing instrument user.

Abstract: Acoustical signals from the acoustical surrounding (U) which impinge upon a reception unit 30 are evaluated and direction of arrival (DOA) of such signals is determined. From signals indicative of such direction of arrival (DOA) a histogram is formed in unit 32. The behavior of such histogram is classified under different aspects or criteria and dependent on classification results in a classifying unit 34 the hearing device and thereby especially its signal transfer characteristic from input acoustical signals to output mechanical signals is controlled or adjusted.