Gn Resound A/s

Other

Gn Resound North America Corporation

Audiologic Hearing Systems LP

Rn Resound A/s

Gn Hearing A/s

James Mitchell Kates

John L Melanson

Eric Lindemann

A hearing aid includes a microphone and an A/D converter for provision of a digital input signal in response to a sound signal received at the microphone in a sound environment, a processor that is configured to process the digital input signal in accordance with a signal processing algorithm to generate a processed output signal, a sound environment detector for determination of the sound environment based at least in part on the digital input signal, and for providing an output for selection of the signal processing algorithm, the sound environment detector including (1) a feature extractor for determination of histogram values of the digital input signal in a plurality of frequency bands, (2) an environment classifier configured for classifying the sound environment into a number of environmental classes based at least in part on the determined histogram values from at least two of the plurality of frequency bands, and (3) a parameter map for the provision of the output for the selection of the signal processing algorithm, and a D/A converter and an output transducer for conversion of the processed output signal to an acoustic output signal.

Hearing aid with histogram based sound environment classification

A hearing aid includes a microphone for conversion of acoustic sound into an input audio signal, a signal processor for processing the input audio signal for generation of an output audio signal; and a transducer for conversion of the output audio signal into a signal to be received by a human, wherein the signal processor includes a compressor with a compressor input/output rule that is variable in response to a signal level of the input audio signal.

Hearing aid with improved compression

A hearing device includes a first filter configured for providing a first frequency part of an input signal of the hearing device, the first frequency part comprising a low pass filtered part, a second filter configured for providing a second frequency part of the input signal, the second frequency part comprising a high pass filtered part, a first synthesizing unit configured for generating a first synthetic signal from the first frequency part using a first model based on a first periodic function, and a combiner configured for combining the second frequency part with the first synthetic signal for provision of a combined signal.

Stability and speech audibility improvements in hearing devices

A hearing aid includes an input transducer for provision of an input signal, a high pass filter configured for providing a high pass filtered part of the input signal, a low pass filter configured for providing a low pass filtered part of the input signal, a synthesizing unit configured for generating a synthetic signal from the high pass filtered part using a model based on a periodic function, wherein a phase of the synthetic signal is at least in part randomized, a combiner configured for combining the low pass filtered part with the synthetic signal for provision of a combined signal, a hearing loss processor configured for processing the combined signal for provision of a processed signal, and a receiver coupled to the hearing loss processor, wherein the receiver is configured for converting an audio output signal into an output sound signal.

Stability improvements in hearing aids

A signal processing system, such as a hearing aid system, adapted to enhance binaural input signals is provided. The signal processing system is essentially a system with a first signal channel having a first filter and a second signal channel having a second filter for processing first and second channel inputs and producing first and second channel outputs, respectively. Filter coefficients of at least one of the first and second filters are adjusted to minimize the difference between the first channel input and the second channel input in producing the first and second channel outputs. The resultant signal match processing of the signal processing system gives broader regions of signal suppression than using the Wiener filters alone for frequency regions where the interaural correlation is low, and may be more effective in reducing the effects of interference on the desired speech signal. Modifications to the algorithms can be made to accommodate sound sources located to the sides as well as the front of the listener. Processing artifacts can be reduced by using longer averaging time constants for estimating the signal power and cross-spectra as the signal-to-noise ratio decreases. A stability constant can also be incorporated in the transfer functions of the first and second filters to increase the stability of the signal processing system.

Binaural signal enhancement system

A hearing aid includes a microphone for conversion of sound into an input audiosignal, a signal processor for processing the input audiosignal, the signal processor including a compressor, and a receiver for conversion of the processed signal into sound, wherein the compressor is configured to adapt attack and release time constants in response to input signal fluctuations.

Hearing aid with adaptive compressor time constants

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.

Hearing aid with suppression of wind noise

A dynamic range compression system is provided, using either a sample-by-sample or a block processing system. Such a system can be used, for example, in a hearing aid. The system, using a frequency-warped processing system, is comprised of a cascade of all-pass filters with the outputs of the all-pass filters providing the input to the frequency analysis used to compute the filter coefficients. The compression filter is then designed in the frequency domain. Using a compression filter having even symmetry guarantees that the group delay is constant and does not depend on the compression gains at any given time. Additionally, due to the use of all-pass filters, the compression filter group delay more closely matches human auditory latency. An inverse frequency transform back into the warped time domain is used to produce the compression filter coefficients that are convolved with the outputs of the all-pass delay line to give the processed output signal.

Dynamic range compression using digital frequency warping

A multi-channel signal processing system adapted to provide binaural compressing of tonal inputs is provided. Such a system can be used, for example, in a binaural hearing aid system to provide the dynamic-range binaural compression of the tonal inputs. The multi-channel signal processing system is essentially a system with two signal channels connected by a control link between the two signal channels, thereby allowing the binaural hearing aid system to model behaviors, such as crossed olivocochlear bundle (COCB) effects, of the human auditory system that includes a neural link between the left and right ears. The multi-channel signal processing system comprises first and second channel compressing units respectively located in first and second signal channels of the multi-channel signal processing system. The first and second channel compressing units receive first and second channel input signals, respectively, to generate first and second channel compressed outputs. The multi-channel signal processing system further includes peak detecting means detecting signal peaks of the first and second channel input signals for generating first and second channel control signals. Thereafter, gain adjusting means adjusts signal gains of the first and second channel control signals. The first and second channel compressing units then respectively compress the first and second channel input signals to produce the first and second channel compressed outputs in accordance with the adjusted first and second channel control signals, respectively.

Binaural compression system

A frequency-warped processing system using either sample-by-sample or block processing is provided. Such a system can be used, for example, in a hearing aid to increase the dynamic-range contrast in the speech spectrum, thus improving ease of listening and possibly speech intelligibility. The processing system is comprised of a cascade of all-pass filters that provide the frequency warping. The power spectrum is computed from the warped sequence and then compression gains are computed from the warped power spectrum for the auditory analysis bands. Spectral enhancement gains are also computed in the warped sequence allowing a net compression-plus-enhancement gain function to be produced. The gain versus frequency function is a set of pure real numbers, so the inverse frequency domain transform gives a set of time-domain filter coefficients. The speech segment is convolved with the enhancement filter in the warped time-domain to give the processed output signal. Processing artifacts are reduced since the frequency-warped system has no temporal aliasing.

Spectral enhancement using digital frequency warping

A digital hearing aid according to the present invention is capable of measuring its own performance. The measurement and initialization capability may be entirely integral to the hearing aid, or an external processor may be used to download the measurement program and the run time program, and assist in computing the parameters. The hearing aid includes a test signal generator for feeding a test signal into the hearing aid amplifier. The response to the test signal is acquired at a specific point in the hearing aid, depending upon what aspect of performance is to be measured. Various elements of the hearing aid and/or the hearing aid feedback may be bypassed. The hearing aid further includes the capability of initializing hearing aid parameters based upon the performance measurements.

Apparatus and methods for hearing aid performance measurement, fitting, and initialization

A frequency-warped processing system using either sample-by-sample or block processing is provided. Such a system can be used, for example, in a hearing aid to increase the dynamic-range contrast in the speech spectrum, thus improving ease of listening and possibly speech intelligibility. The processing system is comprised of a cascade of all-pass filters that provide the frequency warping. The power spectrum is computed from the warped sequence and then compression gains are computed from the warped power spectrum for the auditory analysis bands. Spectral enhancement gains are also computed in the warped sequence allowing a net compression-plus-enhancement gain function to be produced. The speech segment is convolved with the enhancement filter in the warped time-domain to give the processed output signal. Processing artifacts are reduced since the frequency-warped system has no temporal aliasing.

A feedback cancellation system with reduced sensitivity to low-frequency tonal inputs is provided. Such a system can be used, for example, in a hearing aid to prevent cancellation of the desired tonal inputs to the hearing aid, thus improving the gain at high frequencies of the hearing aid while simultaneously preserving the desired tonal inputs at low frequencies. The feedback cancellation system comprises a first adaptive filter block for adaptively filtering an error signal to remove the low-frequency tonal components from the error signal. The first adaptive filter block is constrained so that only low-frequency tones in the error signal are cancelled, thus enabling the feedback cancellation system to still cancel &ldquo;whistling&rdquo; at high frequencies due to the temporary instability of the hearing aid. A second adaptive filter block adaptively filters a feedback path signal to produce an adaptively filtered feedback path signal. The first and second adaptive filter blocks are identical and filter coefficients of the first adaptive filter block are copied to those of the second adaptive filter block. Using an LMS adaptation algorithm, filter coefficients of an adaptive filer of the feedback cancellation system are controlled by the adaptively filtered error signal and the adaptively filtered feedback path signal respectively inputted from the first and second adaptive filter blocks. The adaptive filter then produces an adaptively filtered modeled feedback signal to be subtracted from an electrical audio signal input for updating the error signal of the hearing aid. The hearing aid processes the updated error signal with a digital signal processor to generate an audio output.

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Niwot, CO, United States of America

James Mitchell Kates