Abstract: Method and apparatus for signal processing an input signal in a hearing assistance device to avoid entrainment, the hearing assistance device including a receiver and a microphone, the system comprising using a gradient adaptive lattice filter including one or more reflection coefficients to measure an acoustic feedback path from the receiver to the microphone of the hearing assistance device.

Abstract: A method and apparatus for adding audible noise with time varying volume to audio devices are disclosed which makes the time varying volume envelope of the added audible noise proportional to the time varying volume envelope of sound for frequencies where an individual has a restricted range of perception. The method and apparatus are used to improve the audibility, speech intelligibility, and word recognition characteristics in audio devices.

Abstract: A hearing aid device includes a plurality of microphones which converts picked sound to audio signals, an inter-signal phase difference calculation unit which calculates a phase difference between a first audio signal and a second audio signal, a hand placement determination unit, a directivity sound pickup unit which generates an output signal, a nonlinear amplification unit which controls the signal level of the output signal acquired from the directivity sound pickup unit, and a hearing aid process control unit which, when the hand placement determination unit determines that a state where a phase difference is equal to or smaller than a first threshold value continues for a first time, controls the directivity sound pickup unit such that at least one of the sensitivity-frequency characteristic and the frequency characteristics of the audio signals becomes nondirective, and controls the nonlinear amplification unit to amplify the signal levels of the acquired audio signals.

Abstract: A hearing apparatus and in particular a hearing device is to be provided, in which interference noises are to be reduced however as natural an acoustic image as possible is consequently retained. To this end, a hearing apparatus with a first signal processing channel, into which a noise reduction device is integrated and a second signal processing channel, which has the same input as the first signal processing channel and into which no noise reduction device, but instead a level limitation device is integrated, is proposed. An adding device adds the output signals of both signal processing channels to form an overall output signal. A passive input level-dependent noise reduction can thus be realized, with which quiet signals can be heard again despite the noise reduction.

Abstract: A hearing aid (200) comprises at least one microphone (210), a signal processing means (220) and an output transducer (230). The signal processing means is adapted to receive an input signal from the microphone. The signal processing means is adapted to apply a hearing aid gain to the input signal to produce an output signal to be output by the output transducer, and the signal processing means further comprises means for adjusting the hearing aid gain up or down until the hearing aid gain differs from the direct transmission gain by more than a predetermined value.

Abstract: An apparatus for processing an input sound signal, the apparatus including: gain circuitry configured to control a gain based on a plurality of respective sub-signals of the input sound signal; and an amplification apparatus configured to adjust the amplification of all the plurality of amplitudes based on the gain.

Abstract: A hearing aid (200) comprises at least one microphone (210), a signal processing means (220) and an output transducer (230). The signal processing means is adapted to receive an input signal from the microphone. The signal processing means is adapted to apply a hearing aid gain to the input signal to produce an output signal to be output by the output transducer, and the signal processing means comprises means for adjusting the hearing aid gain by a direct transmission gain calculated for the hearing aid. The invention further provides a method and a system for reducing noise, as well as a computer program product.

Abstract: Disclosed herein, among other things, are methods and apparatus for a level-dependent compression system for hearing assistance devices, such as hearing aids. The present subject matter includes a hearing assistance device having a buffer for receiving time domain input signals and a frequency analysis module to convert time domain input signals into a plurality of subband signals. A power detector is adapted to receive the subband signals and to provide a subband version of the input signals. A nonlinear gain stage applies gain to the plurality of subband versions of the input signals, and a frequency synthesis module processes subband signals from the nonlinear gain stage and to create a processed output signal. The device also includes a filter for filtering the signals, and a level-dependent compression module. The level-dependent compression module is adapted to provide bandwidth control to the plurality of subband signals produced by the frequency analysis stage.

Abstract: A signal processing apparatus is configured to change volume level or frequency characteristics of an input signal with a limited bandwidth in a first frequency range. The apparatus includes: an information extracting unit configured to extract second frequency characteristic information from a collection signal with a limited bandwidth in a second frequency range different from the first frequency range; a frequency characteristic information extending unit configured to estimate first frequency characteristic information from the second frequency characteristic information extracted by the information extracting unit, the first frequency characteristic information including the first frequency range; and a signal correcting unit configured to change volume level or frequency characteristics of the input signal according to the first frequency characteristic information obtained by the frequency characteristic information extending unit.

Abstract: There is described a hearing aid comprising input transducer means for converting input acoustic signals into electrical input signals, signal processor means and output transducer means. The signal processor means is operable to divide said electrical input signals into a plurality of frequency bands and to perform a contrast enhancement operation in each said frequency band, to increase the difference between the amplitudes of those frequency components of the input signals having a relatively high amplitude and those frequency components thereof having a relatively low amplitude to produce output electrical signals in each said respective frequency bands. The output transducer means is operable to produce an output acoustic signal corresponding to a combination of said output electrical signals.

Abstract: A directional hearing aid (100) comprising at least two microphones (201, 202) has means (200) for matching differences in amplitude and phase between the two microphones (201, 202). The microphone matching means (200) compare differences between measured transfer functions of the microphone signal paths at a number of frequencies during use of the hearing aid (100), compares the differences to differences at similar frequencies in a model of the transfer functions of the microphone signal paths, derives a set of parameters based on the comparison, and adjusts the parameters in order to minimize the difference in level differences between the model and the microphones (201, 202). The model is then used to match the microphones (201, 202) mutually by applying appropriate control parameters (103, 104, 105, 106) to an adaptive matching filter (108) carrying one of the microphone signals.

Abstract: A low-power implant system. The system includes an implant for implantation into a person, such as a cochlear implant or a middle ear implant. The implant is capable of communicating with a device via transmission of ultra wideband pulses. The device may be adapted to be worn external to the person, or may be a second implant. So as to conserve battery power, the transmitted ultra wideband pulses may have a low duty cycle of approximately 1/1000 or less. Power savings may also be realized by using time-gating amplifiers in the implant and/or device receiver.

Abstract: A method of adjusting frequency-dependent amplification in an audio amplification apparatus. The audio amplification apparatus includes a forward transfer path (2) connectable to an output transducer, the forward transfer path including a frequency transposing element. The method includes the steps of: presenting stimuli to the output transducer at a plurality of frequencies; adjusting the stimulus level (C) at each frequency to meet a predefined loudness perception level or detection threshold of the listener; deriving an equal loudness contour of output transducer levels from the adjusted stimuli levels; and deriving the frequency-dependent amplification of levels of input signals (I) at each frequency.

Abstract: A method for operating a hearing aid in a hearing aid system where the hearing aid is continuously learnable for the particular user. A sound environment classification system is provided for tracking and defining sound environment classes relevant to the user. In an ongoing learning process, the classes are redefined based on new environments to which the hearing aid is subjected by the user.

Abstract: A device (100) for processing audio data (101), wherein the device (100) comprises a mid-frequency filter unit (105) adapted to selectively filter a mid-frequency range component of the audio data (101) in such a manner that amplitudes of different frequency sub components of the mid-frequency range component of the audio data (101) are scaled so that the scaled amplitudes reflect relations between the original amplitudes of the different frequency sub-components.

Abstract: A hearing apparatus has an input converter for converting an input signal into an electrical signal, an amplifier for amplifying the electrical signal and an output converter for converting the amplified electrical signal into an output signal, and a battery compartment, into which a battery for supplying electrical energy at least to the amplifier can be introduced. The hearing apparatus further has a sensor unit, which is embodied to detect a position of the battery compartment in a contactless fashion and to output signals as a function thereof, by which the hearing apparatus can be switched on or off.

Abstract: A method and apparatus to a hearing ability and a hearing aid using the same are described. The hearing aid outputs a sound to test a hearing ability of a user, to detect an electrical signal generated in a body of the user as a result of the output sound, amplifies the electrical signal detected by an electrode unit, and to determine an amplification ratio of a surrounding sound detected by the hearing aid based on characteristics of peaks of waveforms of the amplified signal.

Abstract: It should be possible to quickly and effectively adjust a hearing aid even in the high-frequency range above 8 kHz. For this purpose it is provided that an open-loop gain measurement is carried out in the upper frequency range and a maximum amplification or a frequency-dependent maximum amplification curve is fixed. This maximum amplification in the high-frequency range should not be exceeded. The hearing aid wearer can then optionally select one of a plurality of amplification curves located there below. In the low-frequency range a conventional amplification adjustment is carried out for example by a prescriptive audiogram-based formula. A hybrid adjustment procedure that is easy to carry out is thereby provided for the entire frequency range.

Abstract: The hearing aid of the present invention comprises a gain calculation section for calculating the gain for amplifying or compressing an input sound signal, a sound pressure calculation section for calculating an output sound pressure level from the input signal and the gain, a clock section for calculating exposure time by integrating the time intervals at which the output sound pressure level is generated, and an exposure time determination section for detecting whether or not the exposure time for every output sound pressure level has exceeded an allowable time.

Abstract: A method for determining a maximum gain is disclosed that is applicable in a forward signal path starting at a microphone (1) and ending at a receiver (11) of a hearing device, the maximum gain being a gain value at which just no feedback occurs. The method comprising the steps of estimating a estimated feedback transfer function (F?) characterizing a feedback signal path (15) starting at a receiver (11) and ending at a microphone (1) of the hearing device, while the hearing device is inserted into an ear of a hearing device user, and adapting the estimated feedback transfer function (F?) as a result of a changing feedback signal path (15) by applying an adaptive algorithm. The invention is characterized by determining the maximum gain from the estimated feedback transfer function (F?), in particular by coefficients of the estimated feedback transfer function.

Abstract: In digital hearing devices, it is common to reduce the preamplifier gain when very loud signals are received in order to avoid clipping. It is known to temporarily reduce the preamplifier gain with fast attack and release times when a clipping of the input signal in the digitiser is expected. Perceived signal quality of the audible signal provided to the user of the hearing device is herein improved by simultaneously providing relatively fast and relatively slow temporary reductions of the amplifier gain such that the slow gain reductions reduce the occurrences of fast gain reductions. This allows the hearing device to provide a high quality sound to the user where clipping is reduced due to the fast gain reductions and where signal artefacts, such as pumping and switching noise, produced by the fast gain reductions is reduced due to the slow gain reductions.

Abstract: Embodiments of the invention provide an amplifier including a gain control circuit having a gain measurement circuit as part of a feedback loop. The gain measurement circuit generates an attenuated output signal, based on the gain of the amplifier circuit, which is used to generate a gain control signal. One amplifier includes a first attenuator enabling primary compression and a second attenuator as part of a gain measurement circuit. In some cases the attenuation factors of the first and second attenuators may be proportional or substantially the same. In some embodiments one or more output stages (e.g., differential amplifiers) are provided to generate the gain control signal based on differential combinations of an audio output, the attenuated output signal, and a primary threshold signal. A ratio of the gains of the two or more comparing amplifiers can set a primary compression ratio in some cases.

Abstract: The present invention provides methods and systems for digitally processing audio signals. Some embodiments receive an audio signal and converting it to a digital signal. The gain of the digital signal may be adjusted a first time, using a digital processing device located between a receiver and a driver circuit. The adjusted signal can be filtered with a first low shelf filter. The systems and methods may compress the filtered signal with a first compressor, process the signal with a graphic equalizer, and compress the processed signal with a second compressor. The gain of the compressed signal can be adjusted a second time. These may be done using the digital processing device. The signal may then be output through an amplifier and driver circuit to drive a personal audio listening device. In some embodiments, the systems and methods described herein may be part of the personal audio listening device.

Abstract: A speech enhancement system that improves the intelligibility and the perceived quality of processed speech includes a frequency transformer and a spectral compressor. The frequency transformer converts speech signals from the time domain to the frequency domain. The spectral compressor compresses a pre-selected portion of the high frequency band and maps the compressed high frequency band to a lower band limited frequency range. The speech enhancement system may be built into, may be a unitary part of, or may be configured to interface other systems that process audio or high frequency signals.

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: The present invention provides for methods and systems for digitally processing audio signals by adjusting the gain of a signal a first time, using a digital processing device located between a radio head unit and a speaker. The methods and systems may filter the adjusted signal with a first low shelf filter, compress the filtered signal with a first compressor, process the signal with a graphic equalizer, compress the processed signal with a second compressor and adjust the gain of the compressed signal a second time. The methods and systems may also filter the signal received from the first low shelf filter with a first high shelf filter prior to compressing the filtered signal with the first compressor. The signal may be filtered with a second low shelf filter prior to processing the signal with the graphic equalizer. Some embodiments filter the signal with a second high shelf filter after the signal is filtered with the second low shelf filter.

Abstract: A speech enhancement system that improves the intelligibility and the perceived quality of processed speech includes a frequency transformer and a spectral compressor. The frequency transformer converts speech signals from the time domain to the frequency domain. The spectral compressor compresses a pre-selected portion of the high frequency band and maps the compressed high frequency band to a lower band limited frequency range.

Abstract: Certain embodiments of the present technology provide improved amplification circuits and hearing aids that can utilize such amplification circuits. In an embodiment, for example, an amplification circuit includes: a first sub circuit configured to create a voltage drop when a supply voltage is above a first voltage; and a second sub circuit configured to create a reverse voltage drop when the supply voltage is below a second voltage, wherein the first and second sub circuits operate to maintain idling current within a range. Certain embodiments of the present technology also provide hearing aids that include removable dampers.

Abstract: The hearing device to be worn by a user of the hearing device comprises an input transducer for transducing incoming acoustical sound into audio signals; a signal processor for processing audio signals; an output transducer for transducing audio signals into outgoing acoustical sound. Said signal processor is designed such that there exists an input SPL range of at least a portion of said incoming acoustical sound, in which an increase in input SPL of said at least one portion of said incoming acoustical sound results in a decrease in output SPL of at least a portion of said outgoing acoustical sound, wherein SPL stands for sound pressure level. Through this, at high input SPL the hearing device's power consumption can be reduced by making use of direct sound, which propagate as sound waves from outside the user's ear canal to the user's ear drum.

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: The present invention relates to a hearing aid comprising a receiver and signal processor circuitry operatively connected hereto. The signal processor circuitry comprises amplifying means (e.g., a Class D amplifier) adapted to generate a switched output voltage for driving the receiver of the hearing aid. Moreover, a power supply unit is included and is adapted to generate a DC output voltage from the switched output voltage. The present invention further relates to an associated method for generating a DC output voltage.

Abstract: A hearing aid (or other assistive listening device) that is suitable for limited use is disclosed. The hearing aid can be configured to be usable for a predetermined period of time. After the expiration of the predetermined period of time, the hearing aid can cease to operate or operate in a degraded manner. Once the hearing aid has become significantly degraded, it can be either disposed of or reconfigured (or reprogrammed) so as to be usable for an additional period of time.

Abstract: The present invention relates to a process for adjusting the sound volume of a digital sound recording characterized in that it comprises: a step consisting of determining, in absolute values, for a recording, the maximum amplitude values for sound frequencies audible for the human ear, a step consisting of calculating the possible gain for a specified sound level setting, between the maximum amplitude value determined above and the maximum amplitude value for all frequencies combined, a step consisting of reproducing the recording with a sound card by automatically adjusting the amplification gain level making it possible to obtain a sound level for the recording of a specified value so that it corresponds to the gain calculated for this recording.

Abstract: A hearing apparatus is provided with a housing including a first housing mounting element and a control element that is mounted rotatably in the first housing mounting element via a control mounting element. Furthermore the housing has at least one second housing mounting element so that the control element can be optionally mounted rotatably in the first housing mounting element or in the second housing mounting element. Thus the control element can be actuated in a desired position.

Abstract: The present invention provides for methods and systems for digitally processing an audio signal. Specifically, the present invention provides for a headliner speaker system that is configured to digitally process an audio signal in a manner such that studio-quality sound that can be reproduced.

Abstract: Certain embodiments of the present technology provide improved amplification circuits and hearing aids that can utilize such amplification circuits. In an embodiment, for example, an amplification circuit includes: a first sub circuit configured to create a voltage drop when a supply voltage is above a first voltage; and a second sub circuit configured to create a reverse voltage drop when the supply voltage is below a second voltage, wherein the first and second sub circuits operate to maintain idling current within a range. Certain embodiments of the present technology also provide hearing aids that include removable dampers.

Abstract: A volume control in a hearing aid which has a signal path from a microphone to a receiver and which provides amplification of the signal delivered to the receiver, the volume control including first and second user input elements to allow the user to change the amplification in a downward and an upward direction, whereby use of the first and second user input elements has different impact on the size of the amplification change effected.

Abstract: A hearing aid comprises a noise suppressor that calculates noise suppression gain, an adjustment amount calculator that calculates an adjustment amount on the basis of a signal strength and a noise component strength, and a nonlinear compressor that calculates a reference gain on the basis of a signal strength and specific reference gain information and adjusts the reference gain on the basis of an adjustment amount, thereby calculating a nonlinear compression gain.

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: The capabilities of a helix hearing instrument may expanded with a behind-the-ear accessory module. The module accommodates additional batteries and a variety of accessory functions including an external volume control.

Abstract: A hearing instrument, e.g. a hearing aid, has a mechanical, manually actuated volume setting element and also remote control volume adjustment. The volume setting element is a control dial with a fixed assignment of the respective rotational position to a respective volume setting. An adjustment by remote control changes the volume setting but not the position of the control dial. This results in the problem that the normally fixed assignment of the respective position of the control dial to the respective volume setting no longer holds true. If there is once again an adjustment by control dial after the volume setting was changed by remote control and the volume prescribed by the control dial is then set, unpleasant jumps in volume may therefore occur.

Abstract: Fine customization of a hearing aid to the individual hearing environments and habits of a user are to be simplified and improved. To this end, when the hearing aid is turned on or when the hearing aid is switched to a particular operating mode, neither the value most recently valid for the parameter prior to turning off or switching to a different operating mode nor the parameter value transferred to the hearing aid at the beginning of programming is set. Rather, from the changes to the value of the parameter taking place during operation of the hearing aid a new start value is ascertained and stored, which value is then set automatically after turning on or switching mode.

Abstract: A hearing aid (22) having a microphone (1), a processor (53) and an output transducer (12), is adapted for obtaining an estimate of a sound environment, determining an estimate of the speech intelligibility according to the sound environment estimate, and for adapting the transfer function of the hearing aid processor in order to enhance the speech intelligibility estimate. The method according to the invention achieves an adaptation of the processor transfer function suitable for optimizing the speech intelligibility in a particular sound environment. Means for obtaining the sound environment estimate and for determining the speech intelligibility estimate may be incorporated in the hearing aid processor, or they may be wholly or partially implemented in an external processing means (56), adapted for communicating data to the hearing aid processor via an appropriate link.

Abstract: The object is to improve the action of a directional microphone in real acoustic environments. To do this, it is envisaged that the interference powers in a directional microphone with three microphones are reduced in that a first and a second microphone signal are adaptively filtered with respect to a first direction, with a direction-determining first parameter being adapted in such a way that the summation of interference powers is reduced. The second and a third microphone signal is adaptively filtered with respect to the first direction, with a direction-determining second parameter being adapted in such a way that the summation of interference powers is reduced. The two parameters are different from each other. This makes it possible, even in real environments, to suppress two interference sources from different directions with one second-order directional microphone.

Abstract: A hearing compensation system comprises a plurality of bandpass filters having an input connected to an input transducer and each bandpass filter having an output connected to the input of one of a plurality of multiplicative automatic gain control (MAGC) circuits whose outputs are summed together and connected to the input of an output transducer. The MAGC circuits attenuate acoustic signals having a constant background level without the loss of speech intelligibility. The identification of the background noise portion of the acoustic signal is made by the constancy of the envelope of the input signal in each of the several frequency bands. The background noise that will be suppressed includes multi-talker speech babble, fan noise, feedback whistle, florescent light hum, and white noise.

Abstract: A hearing device and method for monitoring the hearing ability of a person with impaired hearing allow changes of the hearing ability specifically of a person wearing a hearing aid are to be better able to be registered and taken into account. For this purpose, the setting of an operating element, this serves for setting an output sound level of a hearing device. Furthermore, the ambient sound level in the vicinity of the hearing device is measured. After that, the current setting is compared with the current, measured ambient sound level and a corresponding control signal is output dependent on the result of the comparison. This control signal serves, for example, for signaling to the wearer of a hearing aid the loss of hearing or the automatic adjustment of the hearing aid.

Abstract: A speech enhancement system that improves the intelligibility and the perceived quality of processed speech includes a frequency transformer and a spectral compressor. The frequency transformer converts speech signals from the time domain to the frequency domain. The spectral compressor compresses a pre-selected portion of the high frequency band and maps the compressed high frequency band to a lower band limited frequency range.

Abstract: Data is exchanged between a hearing device and a connectable unit. The data, such as an identification code, may be read from the unit. Data may also be output to the connectable unit. The connectable unit may include a random access memory. Furthermore, the connectable unit may include a sequence control element.