Abstract: A hearing aid is provided which has at least one input transducer for providing an input signal, at least one signal processing channel receiving at least a portion of said input signal, a hearing aid processor for processing said portion of said input signal to produce at least one output signal, an output transducer responsive to said output signal, and a data logger receiving said portion of said input signal for logging of input signal data. The data logger comprises a characterization unit for characterising and logging parameters of the input signal data, and a memory unit for storing said parameters.

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: In a hearing device with an audio signal processing stage (3, 7, 5) and an Rf signal processing stage (9, 11) the Rf signal processing stage is realized in one module (13a), whereas the audio signal processing stage is realized in a second module (13b). The two modules are releasably interlinked by means of a releasable positive locking link (15).

Abstract: Systems and methods for measuring noise exposure associated with use of a wireless headset are presented. In one example, a transition from a wireless headset standby mode operation to a wireless headset active mode operation is identified. A stored noise dose measurement at the wireless headset is recalled, and a current noise dose measurement is calculated at the wireless headset for a duration of the active mode operation. A transition from the wireless headset active mode operation to the wireless headset standby mode operation is identified, and an updated noise dose measurement is recorded at the wireless headset.

Abstract: A hearing aid (62, 72, 82, 92, 102, 112) adapted for alleviating tinnitus of a user (75), comprises an audio input means (83), a signal processing unit (88, 98, 108, 118) and an output transducer (80). The hearing aid further comprises a band stop filter (87) arranged to match a tinnitus of the user and switching means (86, 116) and switch control means (84, 114) for controlling the switching of said band stop filter (87) into and out of the signal path between the audio input means (83) and the output transducer (80), in response to a predefined trigger event. The invention further provides a method of adjusting a hearing aid.

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: 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: A hearing aid includes a microphone for converting sound into an audio input signal, a hearing loss processor configured for processing the audio input signal or a signal derived from the audio input signal in accordance with a hearing loss of a user of the hearing aid, a synthesizer configured for generation of a synthesized signal based at least on a sound model and the audio input signal, the synthesizer comprising a noise generator configured for excitation of the sound model for generation of the synthesized signal including synthesized vowels, and a receiver for generating an output sound signal based at least on the synthesized signal.

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 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 frequency reconfigurable digital filter and an equalizer using the same are disclosed. The digital filter includes a sampling kernel storage section configured to store a sampling kernel for performing upsampling of a model filter response scaled by a sampling constant and generating response of Multi images which are repeatedly formed with a constant period; a complementary conversion section configured to generate response of Multi complementary images repeatedly formed with constant period in frequency domain where the Multi images are not generated, the Multi complimentary images having the same characteristic as the Multi images; and an image response operation section configured to operate response of an image corresponding to a selected band among the Multi complementary images and the Multi images. The filter reconfigures frequency to realize various band pass characteristics only through changing very small number of parameter, and a user may change easily band of the filter.

Abstract: A listening device comprises an input transducer providing an electric input signal comprising audio and a detector coupled to the input transducer, for determining whether the electric input signal is a broadband signal or not and providing a detection signal in response. The listening device furthermore comprises a controllable filter for filtering the electric input signal being coupled to the detector and the input transducer and for outputting a filtered electric input signal such that a component of the electric input signal in the tinnitus frequency range is dampened, if the detection signal indicates that the electric input signal is a broadband signal, and outputting an unfiltered electric input signal such that a component of the electric input signal in the tinnitus frequency range is not dampened, if the detection signal indicates that the electric input signal is not a broadband signal.

Abstract: A hearing device includes an input transducer positionable at an ear of a user for converting an acoustic input to the hearing device into an input signal, and a filter providing a source signal based on a source band of the input signal and for providing a target signal based on a target band of the input signal. The source band contains lower frequencies than the target band. A modulation envelope processor processes the source signal to generate a modulation envelope signal. A signal combiner combines the modulation envelope signal with the target signal from the target band, generating a target output signal. The hearing device may also include a signal processor for processing at least the target output signal and providing a processed output signal, and an output transducer for converting the processed output signal into an acoustic output to be provided to the ear of the user.

Abstract: An earpiece (100) and a method (640) for acoustic management of multiple microphones is provided. The method can include capturing an ambient acoustic signal from an Ambient Sound Microphone (ASM) to produce an electronic ambient signal, capturing in an ear canal an internal sound from an Ear Canal Microphone (ECM) to produce an electronic internal signal, measuring a background noise signal, and mixing the electronic ambient signal with the electronic internal signal in a ratio dependent on the background noise signal to produce a mixed signal. The mixing can adjust an internal gain of the electronic internal signal and an external gain of the electronic ambient signal based on the background noise characteristics. The mixing can account for an acoustic attenuation level and an audio content level of the earpiece.

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: A method of operating a hearing device monitors for a head movement of a hearing device wearer. When a head movement is detected, an acoustic signal acquired by a microphone is output in an amplified and phase-modulated manner with a receiver. Furthermore, a hearing device includes a motion sensor and a signal processing unit that is configured to execute the method.

Abstract: A method for processing and controlling sound signals in a hearing aid is provided. The method comprises estimating a first (102) and a second (104) signal level of an electric input signal (101) based on a first (103) and a second (105) signal level estimator adapted for responding according to a first and a second speed respectively, where the second speed is lower than the first speed and where the estimated second signal level is subtracted from the estimated first signal level, thereby forming a third signal level (106). Subsequently a first and a second compressor gain control output are determined in a first (107) and second (109) compressor based on said third and second signal level respectively. Then the first and second compressor gain control outputs are summed and hereby a net gain control signal (111) is created. Finally the electric input signal is amplified in accordance with the net gain control signal and thereby creating an electric output signal (112).

Abstract: A method and an acoustic signal processing system for noise reduction of a binaural microphone signal (x1, x2) with one target point source and M interfering point sources (n1, n2, . . . , nM) as input sources to a left and a right microphone of a binaural microphone system, include: filtering a left and a right microphone signal by a Wiener filter to obtain binaural output signals of a target point source, where the Wiener filter is calculated as: H W = 1 - ? ( x 1 , n + x 2 , n ) ? ( x 1 , n + x 2 , n ) ? ( x 1 + x 2 ) ? ( x 1 + x 2 ) , where HW is the Wiener filter, ?(x1,n+x2,n)(x1,n+x2,n) is the auto power spectral density of the sum of all of the M interfering point sources components (x1,n, x2,n) contained in the left and right microphone signals and ?(x1+x2)(x1+x2) is the auto power spectral density of the sum of the left and right microphone signals.

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 hearing aid device has an input converter for receiving an input signal and converting the signal into an electrical input signal, a signal processing unit for processing and frequency-dependent amplification of the electrical input signal and generating an electrical output signal, an output converter for converting the electrical output signal into an acoustic output signal, and a sound detector facility for identifying sounds in a signal. Facilities increase an amplification above a normal amplification for a frequency range in which an identified sound contains signal components. Further facilities set the normal amplification of an electrical input signal in dependence on a signal frequency. The amplification is restricted to a maximum amplification in a specific frequency range. The amplification exceeds the normal amplification or the maximum amplification for the duration of the identified sound and at most for a duration which lies below a setting time of a feedback whistling.

Abstract: A method for signal processing for a hearing aid aims to better match signal processing for a hearing aid and in particular a hearing device to a situation and includes processing an input signal in accordance with a first processing algorithm to form a first intermediate signal and processing the input signal in accordance with a second processing algorithm to form a second intermediate signal in parallel with the processing of the input signal in accordance with the first processing algorithm. The input signal is classified by a classifier. Finally, an output signal with a constant mixture ratio is formed both from the first and from the second intermediate signals, taking into account the result of the classification. This allows the advantages of a plurality of algorithms to be used at the same time. A corresponding hearing aid is also provided.

Abstract: A hearing aid has a coupling structure, a first hearing aid housing portion and a second hearing aid housing portion including a flexible material so that the second hearing aid housing portion is adaptable to different ear canal size. The second hearing aid housing portion is detachably coupled to the first hearing aid housing portion via the coupling structure.

Abstract: In one embodiment, a method for customising a sound processing device for an individual listener including presentation of one or more sounds to the listener directly from the sound processing device, each sound comprising a collection of two or more harmonically related tones, spectrally positioned about a frequency of interest, and having a temporal envelope consisting of a rise time, sustain time, and decay time, obtaining information from the listener, adjusting the level of the sounds, and using the adjusted levels to set up the sound processing device for the listener.

Abstract: The present disclosure relates to methods and apparatus of communicating instructions to a hearing assistance device, such as a hearing aid. In various embodiments instructions are formed using tones sent to the hearing assistance device. The instructions can be used to control the operation of the hearing assistance device. The signals may include dual tone multifunction signals or other nonstandard signals. Various detection processes are provided which include but are not limited to using a modified complex Goertzel algorithm to detect tones. The remote device can be a standard device or can be modified to provide the proper signals. The following techniques can be applied to hearing assistance devices including, but not limited to completely-in-the-canal devices, in-the-canal devices, behind-the-ear devices, receiver-in-canal devices, and implanted devices, such as cochlear implants.

Abstract: A hearing-aid device comprising a housing and circuitry module, wherein the circuitry module comprises a battery, a microphone, and a receiver, and the circuit module is inserted in the hearing aid housing. The hearing aid device further comprises a retractable battery sleeve, wherein a battery slot is exposed when the battery sleeve is retracted, and a battery can be inserted into the battery slot from either side of the battery slot with either polarity. The hearing aid housing may comprise an ear-tip, wherein a microphone is mounted in the hearing aid housing near the ear-tip and the microphone receives sounds through at least one sound inlet located on a side surface of the hearing aid housing. The ear-tip additionally comprises three flanges decreasing in size toward the flange closest to the ear canal, and the hearing aid can be inserted into the ear canal in any orientation.

Abstract: An exemplary system and method are directed at receiving an audio signal and process the audio signal into a remapped audio signal based on a plot profile. The plot profile may include at least one of an identified range of audio frequencies. The processing may comprise retrieving an identified range of audio frequencies from the plot profile; determining a range of impaired audio frequencies in the audio signal based on the identified range of audio frequencies; shifting the frequency of at least a portion of the impaired audio frequencies to outside of the identified range; and continuing to retrieve identified ranges of audio frequencies from the plot profile. The shifting of the impaired audio frequencies of the audio signal may be performed until no further identified ranges of audio frequencies are available for consideration.

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: 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: Tinnitus masking systems for treating tinnitus are described where a device is coupled to a surface of a bone or to a tooth or several teeth. Such a device may comprise an oral appliance having an electronic and/or transducer assembly for generating sounds via a vibrating transducer element. Generally, the transducer may generate one or more frequencies of sound via the actuatable transducer to transmit a modified audio signal via vibratory conductance to an inner ear of the patient to mask tinnitus during a peak of the audio signal and to allow the user to perceive the tinnitus during a trough of the audio signal. The audio signal is also modified to account for any hearing loss of the patient as well as a bone conductance profile measured from the patient.

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: In accordance with one aspect of the present invention, a method for detecting a change in the performance of an audio signal processing path is disclosed. The method comprises: selecting a characteristic of a received audio signal indicative of its energy content; determining first and second predetermined values of the selected energy characteristic at respective first and second audio signal frequency bands; calculating a ratio of the first and second predetermined values for a reference time period and a test time period; and comparing the ratio at the reference time period with the ratio of the test time period to determine a performance change in the audio path.

Abstract: A system and method for selectively enhancing an audio signal to make sounds, particularly speech sounds, more distinguishable. The system and method are designed to divide an input auditory signal into a plurality of spectral channels having associated unenhanced signals and perform enhancement processing on a first subset of the spectral channels and not perform enhancement processing on a second subset of the spectral channels. The enhancement processing is performed by determining an output gain for at least the first subset of spectral channels based on a time-varying history of energy of the unenhanced signals associated with each channel in the first subset of the spectral channels and applying the output gain for each of the first subset of the spectral channels to the unenhanced signals to form enhanced signals associated with each of the first subset of the spectral channels.

Abstract: An earphone monitoring system is disclosed that provides for user information and control regarding ambient and monitored sound levels.

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: A hearing aid is enabled for dynamic compression in a way to further improve the perception of acoustic signals in the provision of hearing assistance to a hearing-impaired person. Here, an input signal is divided into a plurality of frequency bands. Input-level-controlled dynamic compression is performed in at least one first frequency band and output-level-controlled dynamic compression is performed in at least one second frequency band. This optimizes both loudness perception and speech intelligibility.

Abstract: The present invention relates to a digital hearing aid, which models the structures of external ear canals, sizes and shape characteristics of which differ between respective persons, obtains resonance gains generated due to the structural characteristics of the external ear canals, and performs digitization and signal processing to allow the resonance gains to be used as the gain factors of the digital hearing aid, and thus applies the gain factors to digital signal processing units. Further, the present invention proposes a gain obtainment unit capable of taking both resonance gains, generated due to the structural characteristics, and gains, obtained through a hearing test, into account, thus reducing the time required for gain fitting and possible errors, and optimizing the performance of the digital hearing aid for each individual.

Abstract: A filter bank system that is optimized with respect to group delay and power consumption is provided. The filter bank system has multiple levels and has an input-side and an output-side filter bank based on a defined filter type. The input-side filter bank has input channels as a variable first parameter, and an oversampling factor as a variable second parameter. For optimizing the multilevel filter bank, a group delay and an operation rate are now respectively determined for each of a plurality of value pairs of the first and second parameters. The value pair for which the associated group delay and the associated operation rate satisfy a defined criterion, in particular for which they are as low as possible, is selected from the value pairs. The input-side filter bank is subsequently configured with the number of channels and the oversampling factor corresponding to the selected value pair.

Abstract: A hearing aid device suitable for executing a frequency compression is optimized to achieve a better assignment between channels of a source frequency range and channels of a target frequency range. A frequency range that can be transmitted by the hearing aid device is split into a number of frequency bands, after which a number of adjacent frequency bands are combined into at least one group of frequency bands and then one frequency band is selected from each group of frequency bands. The selected frequency bands are moved respectively into a target frequency band permanently assigned to the respective group. This prevents that a number of frequency bands from the source frequency range are shifted to the same target frequency band.

Abstract: Sound quality and speech comprehensibility are to be improved for hearing device wearers when watching television. Provision is made for this purpose to record acoustic signals at a recording site on a data medium. The acoustic signals are recorded simultaneously with a first microphone as a first recording. The data medium is played back on an individual playback device in an individual environment. Here the signal played back from the data medium is re-recorded as a second recording with a second microphone. The two recordings are connected to each other, subtracted in particular, and the result is used to adjust a hearing device program. It is thus possible to take into account individual acoustic environmental conditions in the hearing device program.

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 binaural supply with a hearing system is to be enabled, whereby the computing outlay and energy consumption are to be kept as minimal as possible. A hearing system comprising a first hearing apparatus including a first signal input facility and a first communication facility and a second hearing apparatus including a second signal input facility, a second communication facility for receiving a signal from the first communication facility and a second signal processing facility for processing signals from the second signal input facility and the second communication facility are provided to form a common output signal. The signal transmitted from the first to the second communication facility corresponds to a real spectral part of the overall frequency spectrum of the first input signal. As only one part of the overall spectrum is transmitted and/or binaurally processed, the computing outlay and energy consumption is reduced.

Abstract: It should be possible to carry out dynamic range compression in hearing devices, and in particular in hearing aids, so it is free from distortion and practically in real time. For this purpose it is proposed that the modulation spectrum be obtained from the audio or input signal. The modulation spectrum is subsequently directly modified corresponding to a predefined compression function. Finally a modified or compressed output signal is recovered from the modified modulation spectrum. Alternatively a complex envelope may be obtained from the input signal, which is filtered using time-variant modulation filtering corresponding to a predefined compression rule. A distortion-free, compressed output signal may also be recovered herefrom.

Abstract: A hearing device wearer is to have improved possibilities of being able to perform a fine adjustment of his/her hearing device according to a basic setting. To this end, each processing value of a multi-channel processing system is standardized, in particular a filter bank, to a respectively associated basic setting value. A fine adjustment of the processing values to the hearing device wearer can now be carried out in relation to the standardized processing values starting from a standardized base line. It is thus possible for the hearing device wearer to implement a standardization of the setting values at any point in time and based hereupon to intuitively perform his/her setting requests.

Abstract: A system that records audio and stores the recording is provided. The system includes first and second monitoring assemblies mounted in an earpiece that occludes and forms an acoustic seal of an ear canal. The first monitoring assembly includes an ambient sound microphone (ASM) to monitor an ambient acoustic field and produce an ASM signal. The second monitoring assembly includes an ear canal microphone (ECM) to monitor an acoustic field within the ear canal and produce an ECM signal. The system also includes a data storage device configured to act as a circular buffer for continually storing at least one of the ECM signal or the ASM signal, a further data storage device and a record-activation system. The record-activation system activates the further data storage device to record a content of the data storage device.

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 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: A device to transmit an audio signal to a user comprises a transducer and a support. The support is configured for placement on the eardrum to drive the eardrum. The transducer is coupled to the support at a first outer location to decrease occlusion and a second inner location to drive the eardrum. The transducer may comprise one or more of an electromagnetic balanced armature transducer, a piezoelectric transducer, a magnetostrictive transducer, a photostrictive transducer, or a coil and magnet. The device may find use with open canal hearing aids.

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: The object of the invention is to make sub-band processing in hearing aids less computationally intensive. For this purpose a filter bank system comprising an analysis filter bank (AFB) for decomposing an input signal into sub-band signals, a processing device for amplifying at least one of the sub-band signals, and a synthesis filter bank (SFB) for combining the processed sub-band signals into an output signal is provided. The sub-band signals are oversampled and downsampled compared to the input signal. The stopband attenuations of the individual filters of the AFB and SFB are at least as high as a predefined signal-to-noise ratio increased by an attenuation value which is a function of the oversampling factor, the downsampling factor and possibly the gain. The magnitude frequency response of the SFB is approximately matched to the magnitude frequency response of the AFB.

Abstract: Estimates of spectral magnitude and phase are obtained by an estimation process using spectral information from analysis filter banks such as the Modified Discrete Cosine Transform. The estimation process may be implemented by convolution-like operations with impulse responses. Portions of the impulse responses may be selected for use in the convolution-like operations to trade off between computational complexity and estimation accuracy. Mathematical derivations of analytical expressions for filter structures and impulse responses are disclosed.