Abstract: There is disclosed a system (20) for processing sound signals for use in a hearing prosthesis, the system comprising a transducer (1) for converting a sound signal into an electrical signal (30). A first, processor (4) for processing said electrical signal (30) into a plurality of frequency channel signals, each channel signal having an amplitude envelope to define al least one set of channel outputs (40, 41). A second processor (5) for obtaining information relating to a fundamental frequency of the electrical Signal (30). A third processor (6) for obtaining information relating to the harmonic nature of the electrical signal (30). A modulator (7) for modulating the at least one set of channel outputs (40, 41) received from the first processor (4) in accordance with the in formation relating to the fundamental frequency and the harmonic nature of, the electrical signal so as to generate at least one modified set of channel outputs (70, 71).

Abstract: In a method and hearing aid (200) for processing sound signals for hearing impaired persons by providing multi-band compression processing an input sound signal is filtered by a band split filter (202) into a number of frequency bands to obtain band split signals. A signal level for each of the band split signals is determined and the frequency bands are arranged into a number of groups. Based on the signal levels in each of the groups, a compressor input level for a number of band split compressors each associated to one of the frequency bands is calculated. A compressor gain for each band split compressor is determined based on the corresponding compressor input signal and the band split signals are amplified with the corresponding compressor gain and summed in a summing unit (208) to produce an output sound signal.

Abstract: A hearing aid (50) comprises means (55, 56, 57, 58) for reproducing frequencies above the upper frequency limit of a hearing impaired user. The hearing aid (50) according to the invention comprises means (55, 57) for transposing higher bands of frequencies from outside the upper frequency limit of a hearing impaired user down in frequency based on a detected frequency in order to coincide with a lower band of frequencies within the frequency range perceivable by the hearing impaired user. The transposing means (55, 57) comprise an adaptive notch filter (15) for detecting a dominant frequency in the lower band of frequencies, adaptation means (16) controlled by the adaptive notch filter (15), an oscillator (3) controlled by the adaptation means (16), and a multiplier (4) for performing the actual frequency transposition of the signal. The invention further provides a method for processing a signal in a hearing aid.

Abstract: A hearing aid includes a microphone to convert sounds into electrical signals and a memory to store a plurality of voice prints and a plurality of sound-shaping instructions. Each of the plurality of sound-shaping instructions is associated with one of the plurality of voice prints. The hearing aid further includes a processor coupled to the microphone and the memory. The processor is configured to compare at least one sample from the electrical signals to the plurality of voice prints to identify a voice print. The processor selects sound-shaping instructions associated with the voice print and applies the sound-shaping instructions to selectively shape a portion of the electrical signals corresponding to the voice print to produce a shaped signal. The hearing aid further includes a speaker coupled to the processor and configured to reproduce the shaped signal as an audible output.

Abstract: A binaural hearing device system comprises a reception device (1) for one ear with at least two input acoustical/electrical converters (3a, 3b). Via a communication link (5) a signal (A1) which depends on both input converter's output signals is transmitted to a second device (7) for the other ear which comprises at least an output electrical/mechanical converter.

Abstract: A sound processing process, device and software are disclosed which seek to improve pitch perception, with particular application to auditory prostheses. After input sound signals are processed into channels, an algorithm is applied to selectively increase the modulation depth of the envelope signals. In certain embodiments, the channelized signals are adjusted in timing so as to align the phase of modulated envelope signals in different channels. This results in provision of synchronous (phase aligned) modulation periodicity across channels and hence less pitch ambiguity for listeners to the processed signal, or for application of the signal to hearing prostheses such as cochlear implants. In some embodiments, a broadband envelope signal is used to modulate the level of the narrow band channel signals, so that voicing frequency modulation information in the broadband envelope signal is provided in all narrow band channel signals and the phase of modulated signals in the channels are aligned.

Abstract: The present application relates to a hearing aid with suppression of wind noise wherein wind noise detection is provided involving only a single comparison of the input signal power level at first low frequencies with the input signal power level at frequencies that may include the first low frequencies whereby a computational cost effective and simple wind noise detection is provided. The determination of relative power levels of the input signal reflects the shape of the power spectrum of the signal, and the detection scheme is therefore typically capable of distinguishing music from wind noise so that attenuation of desired music is substantially avoided.

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 electrical 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: According to a preferred embodiment of the invention, a system for audiometric assessment and user-specific acoustic enhancement, comprising an audiometry management module, a tone generator coupled to the audiometry management module and adapted to generate streams of acoustical output, and a hearing profile data file, is disclosed. According to the embodiment, the audiometry management module, in a plurality of sessions, instructs the tone generator to send a series of streams of acoustical output of varying acoustical intensity, and, on receiving a series of user responses responsive to audible perception of sounds generated from the streams of acoustical data, the audiometric management module determines, for each of a plurality of frequencies, a hearing threshold, and stores at least that threshold in the hearing profile data file.

Abstract: Disclosed herein, among other things, is a system for frequency translation by high-frequency spectral envelope warping in hearing assistance devices. The present subject matter relates to improved speech intelligibility in a hearing assistance device using frequency translation by high-frequency spectral envelope warping. The system described herein implements an algorithm for performing frequency translation in an audio signal processing device for the purpose of improving perceived sound quality and speech intelligibility in an audio signal when presented using a system having reduced bandwidth relative to the original signal, or when presented to a hearing-impaired listener sensitive to only a reduced range of acoustic frequencies.

Abstract: In order to switch between different hearing programs to adjust to a momentary acoustic scene, a method for adjusting a hearing device, in which one of several possible hearing programs can be selected in order to adjust to a momentary acoustic scene, the method comprising the steps of detecting a desired hearing program change, changing parameters (b1, . . . , bm) of a transfer function provided between a microphone (M1) and a receiver of the hearing device in order to adapt it to the detected hearing program change, adjusting the parameters (b1, . . . , bm) to be changed from a momentary value to a desired value in such a manner that a smooth transition is perceived by the hearing device user while changing from a momentary hearing program to the desired hearing program, whereas each of the smooth transition is individually adjustable.

Abstract: In one embodiment, an apparatus for processing sound includes a means (401) for analyzing a sound signal into a number of frequency bands and a means (403) for applying variable gain to each frequency band independently. Gain in applied under control of a number of gain comparator means (409) each of which generates a number of statistical distribution estimates in respect of each signal and compares those estimates to predetermined hearing response parameters stored in memory (411). The numerous gain compensated frequency bands are then combined (415) in order to generate a single sound signal. The apparatus may be implemented in dedicated hardware embodiment or by software running on a microprocessor.

Abstract: A user equipment (UE) without service in a communication system measures the radio-frequency (RF) power in a bandwidth received by the UE on the possible downlink carriers in a frequency band that is supported by the UE. The bandwidth is typically the bandwidth of communication channels in the communication system. If the UE supports more than one frequency band, the UE may scan more than one of those frequency bands. The UE processes the measurement results with a noise-reducing enhancement technique, such as a median filter, and then examines the processed results for a particular spectral shape, e.g., a shape that corresponds to a cell signal.

Abstract: An improved method for adaptively cancelling acoustic feedback in hearing aids and other audio amplification devices. Feedback cancellation is limited to a frequency band that encompasses all unstable frequencies. By limiting the bandwidth of the feedback cancellation signal, the distortion due to the adaptive filter is minimized and limited only to the unstable feedback regions. A relatively simple signal processing algorithm is used to produce highly effective results with minimal signal distortion.

Abstract: An improved method for adaptively cancelling acoustic feedback in hearing aids and other audio amplification devices. Feedback cancellation is limited to a frequency band that encompasses all unstable frequencies. By limiting the bandwidth of the feedback cancellation signal, the distortion due to the adaptive filter is minimized and limited only to the unstable feedback regions. A relatively simple signal processing algorithm is used to produce highly effective results with minimal signal distortion.

Abstract: A processing method and apparatus for reducing noise in an auditory prosthesis, for example a hearing aid or cochlear implant, are disclosed. The noise floor of the input sound signal is estimated, and the base output level of the output signal is moved in response, in a preferred form to substantially the estimated noise floor level, without modifying the maximum output level. This has the effect of maximising the dynamic range of the user in response to the target sound signal.

Abstract: In a hearing aid device system with a hearing aid device which can be worn on or in the left ear of a user and a hearing aid device which can be worn on or in the right ear of the user for the binaural supply of the user, the aim is to reduce feedback tendency. To this end, it is proposed to transmit audio signals resulting from the microphone signals of the hearing aid devices in a crosswise fashion between the hearing aid devices and thus to emit an acoustic signal recorded by the microphone of a hearing aid device, after the signal processing and amplification, by means of the receiver of the respective other hearing aid device. In this way, the distance between each receiver and microphone, between which a feedback path exists, is essentially increased for the relevant audio signals.

Abstract: The hearing device is operable in a fitting mode and in a listening mode and comprises a transducer for receiving, in the fitting mode, audio test signals, and for converting the audio test signals into signals to be perceived by the user in the fitting mode. It comprises a parameter memory means for storing parameter settings, which parameter settings are obtained from user input received through a user interface in response to the signals perceived by the user in the fitting mode. And it comprises a signal processor using the parameter settings for correcting audio signals at least in the listening mode. The user interface is comprised in the hearing device and the hearing device comprises an audio signal source, in which audio signal source the audio test signals are stored or generated.

Abstract: In a gain adjusting method and a gain adjusting device for adjusting gain of a processed voice signal that is obtained by signal processing of an input voice signal, a masking power of the processed voice signal is computed, and gain is adjusted for every frequency if the frequency is masked according to the masking power, such that a difference between the processed voice signal and the input voice signal where the frequency is not masked is canceled.

Abstract: One embodiment of the present invention is a method for producing a system for describing an electrical network. An output signal of the electrical network is sampled at a frequency that corresponds to the Nyquist criterion for the input signal to the electrical network. A model with a memory is developed for the output signal, which is sampled at a low sampling rate, with this model approximating the output signal. The model is then transformed by suitable interpolation to an interpolated model with a memory. The interpolation results in the model created in this way providing a good approximation to an output signal which is sampled at a high frequency. The resultant system can be used for predistortion.

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: A hearing aid device for processing an input sound signal via a microphone so as to output a sound signal from an earphone comprises a signal processing element, a couple of acoustic nerve excitation pattern calculation elements, a couple of acoustic filter shape memory elements, a comparison element, a correction processing element, etc. The couple of acoustic nerve excitation pattern calculation elements calculate acoustic nerve excitation patterns of a normal hearing person and a hearing impaired person based on an output signal of the signal processing element and acoustic filter shapes of the normal hearing person and the hearing impaired person which are stored in the couple of acoustic filter shape memory elements. Each of the acoustic nerve excitation patterns is compared via the comparison element.

Abstract: An adaptive place-pitch ranking procedure for use with a cochlear implant or other neural stimulation system provides a systematic method for quantifying the magnitude and direction of errors along the place-pitch continuum. The method may be conducted and completed in a relatively short period of time. In use, the implant user or listener is asked to rank the percepts obtained after a sequential presentation of monopolar stimulation pulses are applied to a selected spatially-defined electrode pair. The spatially-defined electrode pair may be a physical electrode pair or a virtual electrode pair. A virtual electrode pair includes at least one virtual electrode contact. Should the patient's judgment of pitch order be correct for all applied interrogations, then no further testing involving the tested electrode pair (two electrode contacts) is undertaken.

Abstract: An adaptive place-pitch ranking procedure for use with a cochlear implant or other neural stimulation system provides a systematic method for quantifying the magnitude and direction of errors along the place-pitch continuum. The method may be conducted and completed in a relatively short period of time. In use, the implant user or listener is asked to rank the percepts obtained after a sequential presentation of monopolar stimulation pulses are applied to a selected spatially-defined electrode pair. The spatially-defined electrode pair may be a physical electrode pair or a virtual electrode pair. A virtual electrode pair includes at least one virtual electrode contact. Should the patient's judgment of pitch order be correct for all applied interrogations, then no further testing involving the tested electrode pair (two electrode contacts) is undertaken.

Abstract: The wind sensitivity of hearing aids is reduced. The noise level of at least two microphones is measured and to compared with one another. The microphones are controlled according to the comparison result. In one embodiment, the microphone having the lowest noise level is used as an omnidirectional microphone in a wind situation.

Abstract: According to a method embodiment for performing a real ear measurement (REM), a stimulus signal is transformed into a frequency domain stimulus signal with a plurality of frequency ranges. The frequency domain stimulus signal is amplified with a desired gain for each of the plurality of frequency ranges to provide an amplified stimulus signal. The amplified stimulus signal is transformed into the acoustic signal in the ear canal, which is detected to provide a detected acoustic signal, and the detected acoustic signal is transformed into a frequency domain detected signal with the plurality of frequency ranges. A sound level for the plurality of frequency ranges is measured. The desired gain for the frequency ranges is automatically adjusted based on the measured sound levels and the desired sound pressure levels for the plurality of frequency ranges. The method can be performed within a hearing assistance apparatus.

Abstract: A hearing aid (30) includes a microphone (71), a signal processor (20) and an output transducer (22), and the signal processor (20) includes a set of audio processing parameters mapped to a set of stored noise classes (12) and a noise classification block (8) for classifying the background noise for the purpose of optimizing the frequency response in order to minimize the effects of the background noise. The hearing aid may further include a neural net for controlling the frequency response. A method for reducing a noise component in a signal includes the steps of classification of the noise component, comparing the noise component to a set of known noise components, and adapting the processed audio signals according to a corresponding set of frequency response parameters.

Abstract: The invention regards a method for sound processing in an audio device wherein an audio signal is provided and the audio signal is frequency shaped according to the need of a user of the audio device and the frequency shaped signal is served at the user in a form perceivable as sound. According to the invention at least two different frequency shaping schemes are available and a choice is made of the frequency shaping scheme to be used.

Abstract: The hearing aid system (102) has a sound-to-vibration conversion circuitry including a microphone system (108), an electronic amplifier (114) and a vibrator (120). A housing (121) accommodates the vibrator (120). The vibrator (120) is connected to an abutment (124) that goes through the skin. The abutment (124) is connected to a fixture (126) that is anchored in the skull bone (128). The sound-to-vibration conversion circuitry has an AfD converter (212) that converts an analogue microphone signal into a digital signal.

Abstract: The invention relates to a hearing device 1 adapted for placement in, at or near a person's ear, the hearing device 1 comprising a microphone 2, a receiver 4 and a signal conditioning means 3 connected to the microphone 2 and to the receiver 4, the microphone 2 being arranged for receiving acoustical signals from the person's surroundings 7 and converting these acoustical signals into electrical signals and the receiver 4 being arranged for converting electrical signals into acoustical signals and transmitting these into the ear's ear canal 13. The object of the present invention is to provide a small, light-weight hearing device 1. The problem is solved in that the receiver 4 comprises a thermoacoustical transducer 18, which allows for a receiver 4 which may take up less space in the hearing device 1 and may have a smaller weight. This has the advantage of allowing the hearing device 1 to be small and light-weight, thus providing an improved wearing comfort. The invention may e.g.

Abstract: Disclosed is a method of reducing feedback in a hearing aid adapted to be worn by a user, the method comprising the step of: receiving an audio input signal in an input transducer in the hearing aid; wherein the method further comprises the steps of: transforming the input signal into the frequency domain; dividing the audio signal into a plurality of frequency bands; determining a threshold frequency over which a plurality of upper frequency bands lies; multiplying each of the plurality of upper frequency bands by a random phase, thereby obtaining a plurality of phase randomized upper frequency bands; synthesizing the plurality of phase randomized upper frequency bands and the lower frequency bands to an output signal; transforming the output signal into the time-domain; and transmitting the output signal to an output transducer of the hearing aid.

Abstract: Method and system for automatically adjusting a hearing aid. The method includes measuring an acoustic reflectance associated with an ear canal as a function of an incident pressure and an acoustic frequency, processing information associated with the measured acoustic reflectance, determining a reflectance slope based on, at least, information associated with the measured acoustic reflectance, and adjusting, at least, one parameter associated with the hearing aid based on, at least, information associated with the reflectance slope. The reflectance slope is associated with a reflectance component varying with the incident pressure.

Abstract: A method for individually fitting a hearing instrument to a user, by: starting operation of the hearing instrument; pre-defining a desired target loudness function perceived by the user defined as function of frequency and input sound pressure level at the microphone; measuring for a given measurement parameter set of perceived loudness levels and frequencies or frequency bands the respective transducer input audio signal level to be applied to the transducer input to achieve the respective perceived loudness level at the respective frequency or frequency band, said measurement parameter set having at least a low, intermediate and high loudness levels, and said intermediate loudness level being measured for a larger number of frequencies or frequency bands and with a finer frequency resolution than said low and high loudness levels; calculating an individual gain function to be implemented in the audio signal processing unit to achieve the pre-defined target loudness function by taking into account the measur

Abstract: A tone control circuit includes a capacitor-resistor circuit; a resistor-capacitor circuit which is connected to the capacitor-resistor circuit; and a variable resistor which is connected between the resistor-capacitor circuit and capacitor-resistor circuit or a connecting point of both circuits. The variable resistor controls tone easily at high and low frequencies without change of the loudness when voice is output.

Abstract: A hearing prosthesis, including receiver means for receiving a signal representative of a signal over a frequency range; a first filter bank, having a relatively higher resolution, adapted to process said received signal and produce a first set of channel outputs relating to a selected region or regions of said frequency range; and a second filter bank having a relatively lower resolution, adapted to process said received signal and produce a second set of channel outputs relating to at least the rest of said frequency range; combination means to combine the first and second sets of channel outputs, and processing means operative upon the combined outputs so as to produce a set of stimulation signals for said hearing prosthesis.

Abstract: Techniques and tools are described for processing reconstructed audio signals. For example, a reconstructed audio signal is filtered in the time domain using filter coefficients that are calculated, at least in part, in the frequency domain. As another example, producing a set of filter coefficients for filtering a reconstructed audio signal includes clipping one or more peaks of a set of coefficient values. As yet another example, for a sub-band codec, in a frequency region near an intersection between two sub-bands, a reconstructed composite signal is enhanced.

Abstract: A hearing aid processing apparatus (100) includes: audio input units (101, 121) configured to receive an input audio; hearing aid signal processing units (104, 124) configured to generate first and second output signals each having different frequency characteristics, from an input signal of the input audio received by the audio input units (101, 121), based on the characteristics of the band pass filter having the greatest bandwidth among virtual band pass filters composing an auditory filter of a listener: a first audio output unit (106) configured to output, as an audio, the first output signal generated by the hearing aid signal processing unit (104) to the left ear of the listener; and a second audio output unit (126) configured to output, as an audio, the second output signal generated by the hearing aid signal processing unit (124) to the right ear of the listener.

Abstract: Method and apparatus for determination of gain margin of a hearing assistance device under test. In varying examples, the impulse response for multiple levels can be taken and used to arrive at a gain margin. The method and apparatus, in various examples, process critical portions of the resulting data for efficient processing and to increase accuracy of measurements. The method and apparatus performing a plurality of measurements to determine impulse responses and to derive gain margin as a function of frequency therefrom. The present subject matter includes principles which may are adapted for use within a hearing assistance device using a single white noise stimulus, according to one example. The principles set forth herein can be applied to occluding and non-occluding hearing device embodiments. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: The object of the claimed hearing aid device is to improve the speech intelligibility of a speech signal transmitted by the hearing aid device. To this end provision is made to define a maximum gain of the input signal and to determine a target gain at least at a first and second frequency of an input signal. A resulting gain is set in the hearing aid device, which does not exceed the maximum gain. A reduction of the resulting gain compared with the target gain at the first frequency is compensated for according to the invention by an automatic increase in the set resulting gain compared with the target gain at the second frequency, with compensation preferably being achieved by improving the speech intelligibility of a speech signal transmitted with the aid of the hearing aid device.

Abstract: Multi-stage filter bank systems with filter banks of different bandwidth frequently cause interference in hearing apparatuses and in particular in hearing devices. Therefore a hearing apparatus is proposed with a filter bank system having a multi-stage analysis filter bank and/or a multi-stage synthesis filter bank, to break down an input signal of the hearing apparatus into a number of partial band signals by way of a number of filter bank channels and/or to recombine partial band signals of a number of filter bank channels. The filter bank system is equipped with at least one equalization filter, to equalize differences in the complex frequency responses between filter bank channels.

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: 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 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 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: The present invention relates to a method of adapting signal processing characteristics of a portable communication device to a hearing impaired user, comprising the steps of determining a perceptual reference level (PRL) of a first stimuli signal (FSS) in a reference frequency band (RFB) by presenting said first stimuli signal (FSS) to said hearing impaired user, and obtaining perceptual judgements of a loudness of said first stimuli signal (FSS) from said hearing impaired user, and determining said perceptual reference level (PRL) of a second stimuli signal (SSS) in a further frequency band (FFB) by presenting said second stimuli signal (SSS) to said hearing impaired user, and requesting said hearing impaired user to compare a loudness of said second stimuli signal (SSS) with said loudness of said first stimuli signal (FSS).

Abstract: Hearing aid wearers can find themselves in situations in which the necessary replacement and/or recharging of a discharged voltage source is not immediately possible. In such a situation, reducing the low-frequency signal elements in the acoustic output signal allows a type of emergency mode to be maintained for a certain period. The majority of important acoustic information thus remains comprehensible to the hearing aid wearer.

Abstract: A wireless transmitter includes an antenna and a signal generator. A control signal input to the signal generator controls a frequency band of a carrier frequency. The control signal also controls the transmitted signal power. The control signal is generated by a decoder unit that decodes an audio signal.

Abstract: A method of enhancing sound heard by a hearing-impaired listener comprises monitoring the sound in an environment in which the listener is located; and manipulating the frequency of high frequency components of the sound in a high frequency band, with little, if any, distortion to components of the sound in a speech frequency band, to enhance spectral cues to aid the listener in sound externalisation and spatialisation.

Abstract: A sound processing process is disclosed, with particular application to auditory prostheses. After input sound signals are processed into channels, an algorithm is applied to selectively increase the modulation depth of the envelope 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: A wearable communication device includes a bone conduction actuator which is applicable for being in contact with a user's wrist, hand, back of the hand, finger or nail in order to transmit voice signals. The user inserts the user's finger into the user's ear canal, or touches the user's finger to a part near the user's ear, or puts the user's fingertip or nail on the user's ear canal so as to block the user's ear canal when the user uses the wearable communication device.

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.