Abstract: In one embodiment, the present invention provides a sound processing device with a binaural input and binaural output, where “binaural input” means at least one microphone mounted in or near each ear of the device user, and “binaural output” means at least one output signal directed to each ear. The device may be comprised of two parts connected by a wired or wireless link.

Abstract: In one embodiment, the present invention provides a sound processing device with a binaural input and binaural output, where “binaural input” means at least one microphone mounted in or near each ear of the device user, and “binaural output” means at least one output signal directed to each ear. The device may be comprised of two parts connected by a wired or wireless link.

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: The invention relates to oscillation suppression and, more particularly, concerns a method and apparatus for suppressing oscillation in a signal identified as or suspected of containing an oscillation due to feedback. The method involves converting the signal into frequency bands in the frequency domain, applying, for a selected period of time, a randomly changing phase to the signal in at least one of said frequency bands, and reconverting the converted signal into an output wave form signal. The selected period is divided into a series of successive time windows, and for each successive time window a newly generated random or pseudo-random phase is applied to the signal. The method can be used in combination with a method for detecting oscillation in said signal in each of the frequency bands, a randomly changing phase applied in each frequency band for which said oscillation has been detected. The invention has particular application in hearing aid devices.

Abstract: The invention relates to oscillation detection and, more particularly, concerns a method and apparatus for identifying oscillation in a signal due to feedback, permitting appropriate action to be taken to suppress the oscillation. The method involves using an FFT device or similar to convert a signal at each of a series of successive time windows into the frequency domain, calculating, for each of a plurality of frequency bands, the change in signal phase from a time window to a subsequent time window, and comparing, for some or all of said frequency bands, the results of the calculation step to one or more defined criteria to provide a measure of whether oscillation due to feedback is present in the signal. For additional discrimination, the change in signal amplitude from a time window to a subsequent time window may also be calculated for each of the frequency bands, and the result compared with one or more further defined criteria. The invention has particular application in hearing aid devices.

Abstract: A sound processing method for auditory prostheses, such as cochlear implants, which is adapted to improve the perception of loudness by users, and to improve speech perception. The overall contribution of stimuli to simulated loudness is compared with an estimate of acoustic loudness for a normally hearing listener based on the input sound signal. A weighting is applied to the filter channels to emphasize those frequencies which are most important to speech perception for normal hearing listeners when selecting channels as a basis for stimulation.

Abstract: The sound processor and method uses a model of basilar membrane motion to select stimuli, based upon the predicted motion which the acoustic signal presented would produce in an acoustically excited normally hearing cochlea. The filter; used, in contrast to single channel per electrode approaches, cover multiple channels and overlap with each other. Consequently the stimuli presented produce a neural excitation pattern which approximates the spatio-temporal travelling wave observed on the basilar membrane in an acoustically excited normally hearing cochlea. Preferably, the predicted electrode stimuli are based upon the instantaneous predicted amplitude of the electrode location.

Abstract: A method and apparatus for detecting an envelope of an audio signal, and a method and apparatus for enhancing the pitch cue of an audio signal perceived by a cochlear implant patient where the audio signal is processed and input to an implant device of the recipient. The methods and apparatuses use techniques such as filtering, rectifying, detecting peak values, sampling, resetting, comparing and multiplying various signals to detect the envelope or enhance the pitch cue of the audio signal.

Abstract: In one aspect of the invention, a cochlear implant constructed and arranged to successively generate stimulation signals each comprising at least one stimulus pulse such that said successive stimulation signals incrementally build a neural excitation pattern that accurately reflects a received sound. In one embodiment, each said successive stimulation signal is generated based on the cumulative effect of all previous stimulus pulses, thereby compensating for finite spatial spreading of individual stimulus pulses as well as for the temporal integration of the neural excitation pattern along the neural pathways.

Abstract: An input sound signal (210) is processed in order to meet a target dynamic range (910, 920). At least one gain, specific to the input sound signal (210), is applied to the input sound signal (210) to produce a processed sound signal (214). A dynamic range of the processed sound signal is measured, and a match of the measured dynamic range with the target dynamic range (910, 920) is determined. The gain is adjusted in accordance with at least one input sound signal-specific parameter, to improve the match of dynamic range of the processed sound signal (214) to the target dynamic range (910, 920). The input sound signal-specific parameter is adaptive in response to at least one monitored signal condition.

Abstract: In one embodiment, an apparatus for processing sound includes a means (401) for analysing 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 ot statistical distribution estimates in respect of each signal and compares those estimates to predetermined hearing presponse 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.