Abstract: A method for noise reduction in a hearing aid device is described, with a signal, which comprises a useful and an interference signal part, being processed in the hearing aid device and with the interference signal part being reduced to the benefit of the useful signal part and with the reduction of the interference signal part being carried out as a function of the input level of the signal, with the interference signal part being more heavily attenuated with a high input level than with a low input level.

Abstract: A filter bank with a sufficiently high resolution for amplification and noise reduction and with the lowest possible computational complexity is provided for a hearing device and, in particular, for a hearing aid. Two-stage frequency transformation with little latency is therefore proposed for hearing aids. Some of the processing, for example the amplification, is carried out after high stopband attenuation in the first stage. An increased frequency resolution is achieved in a second stage before the back-transformation in the first stage, which is favorable for noise reduction, for example.

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: Own voice recognition (OVR) for hearing aids, detects time instances where the person wearing the device is speaking. Classification of the own voice is performed dependent on a fixed or adaptive detection threshold. Automatic tuning in a real-time system depends on general noise statistics in the input signals. The noise is removed from the received signal and is characterized by signal-to-noise ratio and noise color. An optimal detection threshold for own voice recognition is determined based on the noise characteristics. A noise detection model is created by smoothed Voronoi tessellation. Own voice detection is performed by a processor.

Abstract: A hearing device has a feedback suppression unit. The hearing device further has a low-pass filter characterized by a first cut-off frequency, which couples out a low-frequency signal component from an output signal of the hearing device, and a high-pass filter characterized by a second cut-off frequency, which couples out a high-frequency signal component from the output signal of the hearing device. A frequency shift unit shifts the frequency of the high-frequency signal component to higher frequencies. A gap exists between the first and the second cut-off frequency. As a result of the different limit frequencies, signal distortions caused by frequency shifts are effectively suppressed. Feedback is suppressed continuously and rapidly at higher frequencies.

Abstract: The phase differences of microphones of a hearing aid microphone are to be reduced. To do this, the level of an output signal of a directional microphone is compared with an omnidirectional signal. If the level of the output signal of the differential directional microphone is above the level of the omnidirectional signal, this level difference is minimized by an adaptive, frequency-selective transit time compensation in individual frequency bands and phase matching of the microphones is thus achieved. By means of an alternative method, microphone matching is achieved in that the measurable delay of the two microphone signals is adaptively limited in individual frequency bands to a maximum value corresponding to the sound transit time between the microphones. Phase matching without knowing the position of a sound source can thus be achieved.

Abstract: A filter bank configuration for a hearing device has filters in an analysis filter bank and corresponding filters in a synthesis filter bank that are coupled pair-wise to form a channel in each case. In order to compensate for a hearing loss, sub-band signals are amplified in the individual channels with the aid of multipliers. In the process, an audible distortion of an output signal Y of the filter bank configuration as a result of differences between amplification factors of the multipliers of neighboring channels should be prevented. Here, at least one channel changes a phase of a sub-band signal transmitted by the channel such that a difference between a group delay of the filter bank configuration and a prescribable reference value is reduced for at least one predetermined frequency. The filter bank configuration is particularly suited to hearing aids.

Abstract: Hearing aids with microphone and telephone coil are to be made simpler and more convenient. For this purpose it is provided to use an adaptive filter to compensate acoustic and electromagnetic feedback. In order to allow for the propagation delay differences, a delay element is connected downstream of the telephone coil. The microphone and telephone coil signals can be individually weighted with the factors a and b so that mixed mode is also possible.

Abstract: In a method and an acoustic system that generate a stereo signal for each of multiple. separate sources, a blind source separation of at least two microphone signals is conducted to acquire BSS filters. Each of the microphone signals is filtered with its own filter transfer function that is the quotient of a power density spectral portion of the respective sound source and the overall power density spectrum of the respective microphone signal, such that two stereo signals are obtained for each microphone signal. An approximation of the signals to be separated, for example for each of two hearing devices, is thereby possible.

Abstract: A method and an acoustic signal processing system for noise reduction of a binaural microphone signal are proposed. A source signal and two interfering signals input to a left and a right microphone of a binaural microphone system respectively. A left and a right microphone signal is filtered by a Wiener filter to obtain binaural output signals of the source signal. The Wiener filter is calculated as H W ? ( ? ) = 1 - S y ? ? 1 , y ? ? 1 ? ( ? ) S v ? ? 1 , v ? ? 1 ? ( ? ) + S v ? ? 2 , v ? ? 2 ? ( ? ) , wherein Sy1,y1(?) is the auto power spectral density of the sum of the interfering signals contained in the left and right microphone signal, Sv1,v1(?) is the auto power spectral density of the filtered left microphone signal and Sv2,v2 is the auto power spectral density of the filtered right microphone signal.

Abstract: Adaptation of hearing devices is rendered more convenient and more accurate with a method for adapting a hearing device to a hearing device support. The hearing device is selected on the basis of initial data of several hearing devices with respect to the data relating to hearing loss of the hearing device support. The selected hearing device is preset using a target reinforcement curve and, optionally, a setting of the preset hearing device is finely adjusted. At least one of the mentioned steps of selecting and presetting as well as, optionally, the step of fine adjustment is carried out by means of a single perceptive model which individualizes the hearing loss data projected by the hearing loss of the hearing device support.

Abstract: The preferred direction of a directional microphone of a hearing device and, in particular, a hearing aid should be selected automatically in a quick and reliable fashion. To this end, provision is made for a method for operating a hearing device with such a microphone which can be switched into at least a first and a second directional characteristic. Initially, respectively one signal-to-noise ratio is determined for the first and the second directional characteristic. Subsequently, the directional microphone is switched into that one of the two directional characteristics which leads to the higher signal-to-noise ratio. In particular, the total signal powers of different directional microphone signals can be determined to this end and the interference signal powers can be calculated in parallel thereto in a channel-specific fashion. Corresponding SNR values then result from the differences in the logarithmic power values.

Abstract: A hearing device including a signal processing device for processing an input signal and generating an output signal and a modeling device in which a perceptive model is implemented, in order to generate a psycho-acoustic value for controlling the signal processing device, is provided. Data mapping of the hearing loss, in particular audiogram data, are input into the modeling device and the perceptive model determines the psycho-acoustic value for controlling the signal processing device based on the data from the data mapping and the output signal.

Abstract: A microphone system and an associated method are proposed. The microphone system comprises at least two omnidirectional, microphone signal-emitting directional microphones connected electrically to one another to establish directivity, at least one filter unit with at least one adaptation parameter for the adaptive filtering of the at least two microphone signals and a control unit to change the at least one adaptation parameter such that the sum of interference power is reduced. The value range of the at least one adaptation parameter is limited. The control unit determines limits from a comparison of the noise floor of the ambient noise with a microphone noise number. The adaptation range of an adaptive differential directional microphone is a function of stationary component of background noise, so the directivity can be selected such that the non-stationary microphone noise resulting due to directivity is masked by the stationary component of the background noise.

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 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: The phase differences of microphones of a hearing aid microphone are to be reduced. To do this, the level of an output signal of a directional microphone is compared with an omnidirectional signal. If the level of the output signal of the differential directional microphone is above the level of the omnidirectional signal, this level difference is minimized by an adaptive, frequency-selective transit time compensation in individual frequency bands and phase matching of the microphones is thus achieved. By means of an alternative method, microphone matching is achieved in that the measurable delay of the two microphone signals is adaptively limited in individual frequency bands to a maximum value corresponding to the sound transit time between the microphones. Phase matching without knowing the position of a sound source can thus be achieved.

Abstract: The phase differences of microphones of a hearing aid microphone are to be reduced. To do this, the level of an output signal (y1(t)) of a directional microphone is compared with an omnidirectional signal (y1?(t)). If the level of the output signal of the differential directional microphone (y1(t)) is above the level of the omnidirectional signal (y1?(t)), this level difference is minimized by an adaptive, frequency-selective transit time compensation (A) in individual frequency bands and phase matching of the microphones (M1,M2) is thus achieved. By means of an alternative method, microphone matching is achieved in that the measurable delay of the two microphone signals (x1,x2) is adaptively limited in individual frequency bands to a maximum value corresponding to the sound transit time between the microphones (M1,M2). Phase matching without knowing the position of a sound source can thus be achieved.

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: There is described a method for operation of a hearing device system with two microphones arranged spatially separated from one another and with sound-generating output units assigned to these microphones, in which, by comparison of the microphone signals or of signals derived therefrom, feedback is detected, and on detection of the feedback measures are initiated for reducing the feedback, and with the comparison of the microphone signals or of signals derived therefrom comprising a frequency-selective power comparison. There is also described a hearing device system suitable for this method.