Abstract:
The application relates to a filter bank for an audio processing device, e.g. a hearing aid. The filter bank comprises an analysis filter bank comprising a plurality of M first filters hm(n), where m=0, 1, . . . , M−1 is a frequency band index, n being a time index, the first filters hm(n) having a first filter length of Lhm; a synthesis filter bank comprising a plurality of M second filters gm(n), m=0, 1, . . . , M−1, the second filters gm(n) having second filter lengths of Lgm; the plurality of first and second filters being arranged in pairs, each pair forming a frequency channel. the first filters hm(n) exhibiting a first filter delay τh, the second filters gm(n) exhibiting a second filter delay τg, each of the first filter lengths Lhm and the second filter lengths Lgm is uneven, and wherein the first filters are subject to the constraint that the sum of the first filters hm(n) of the analysis filter bank is a delta function δ(n−τh).
Abstract:
An adaptive level estimator for providing a level estimate of an electric input signal representing sound is provided. The adaptive level estimator comprises a first level estimator configured to provide a first level estimate of the electric input signal in a first number K1 of frequency bands; a second level estimator configured to provide a second level estimate of the electric input signal and/or associated attack/release time constants in a second number K2 of frequency bands, wherein K2 is smaller than K1; and a level control unit receiving said first and second level estimates and configured to provide said resulting level estimate based on said first and said second level estimates and/or said associated attack/release time constants. The invention may e.g. be used in devices or applications that benefit from a dynamic adaptation of an input signal level to a listener's (possibly limited) dynamic range of sound level perception, or to any other specific dynamic range deviating from that of the environment sound.
Abstract:
Signal processing methods for predicting the intelligibility of speech, e.g., in the form of an index that correlate highly with the fraction of words that an average listener (amongst a group of listeners with similar hearing profiles) would be able to understand from some speech material are proposed. Specifically, solutions to the problem of predicting the intelligibility of speech signals, which are distorted, e.g., by noise or reverberation, and which might have been passed through some signal processing device, e.g., a hearing aid are described. In summary, the disclosure present solutions to the following problems: 1. Monaural, non-intrusive intelligibility prediction of noisy/processed speech signals 2. Binaural, non-intrusive intelligibility prediction of noisy/processed speech signals 3. Monaural and binaural intelligibility enhancement of noisy speech signals.
Abstract:
The application relates to a binaural hearing system comprising left and right hearing devices, e.g. hearing aids, each comprising a) a multitude of input units, each providing a time-variant electric input signal xi(t) representing sound received at an ith input unit, t representing time, the electric input signal xi(t) comprising a target signal component si(t) and a noise signal component vi(t), the target signal component originating from a target signal source; b) a configurable signal processing unit for processing the electric input signals and providing a processed signal y(t); c) an output unit for creating output stimuli to the user, d) transceiver circuitry allowing information to be exchanged between the hearing devices, and e) a binaural speech intelligibility (SI) prediction unit for providing a binaural SI-measure of the predicted speech intelligibility of the user when exposed to said output stimuli, based on processed signals yl(t), yr(t) from the signal processing units of the respective left and right hearing devices. This allows the hearing devices to control the processing of the respective electric input signals based on said binaural SI-measure.
Abstract:
The present invention regards a hearing aid device at least one environment sound input, a wireless sound input, an output transducer, electric circuitry, a transmitter unit, and a dedicated beamformer-noise-reduction-system. The hearing aid device is configured to be worn in or at an ear of a user. The at least one environment sound input is configured to receive sound and to generate electrical sound signals representing sound. The wireless sound input is configured to receive wireless sound signals. The output transducer is configured to stimulate hearing of the hearing aid device user. The transmitter unit is configured to transmit signals representing sound and/or voice. The dedicated beamformer-noise-reduction-system is configured to retrieve a user voice signal representing the voice of a user from the electrical sound signals. The wireless sound input is configured to be wirelessly connected to a communication device and to receive wireless sound signals from the communication device. The transmitter unit is configured to be wirelessly connected to the communication device and to transmit the user voice signal to the communication device.
Abstract:
The present invention regards a hearing aid device at least one environment sound input, a wireless sound input, an output transducer, electric circuitry, a transmitter unit, and a dedicated beamformer-noise-reduction-system. The hearing aid device is configured to be worn in or at an ear of a user. The at least one environment sound input is configured to receive sound and to generate electrical sound signals representing sound. The wireless sound input is configured to receive wireless sound signals. The output transducer is configured to stimulate hearing of the hearing aid device user. The transmitter unit is configured to transmit signals representing sound and/or voice. The dedicated beamformer-noise-reduction-system is configured to retrieve a user voice signal representing the voice of a user from the electrical sound signals. The wireless sound input is configured to be wirelessly connected to a communication device and to receive wireless sound signals from the communication device. The transmitter unit is configured to be wirelessly connected to the communication device and to transmit the user voice signal to the communication device.
Abstract:
A hearing device comprises A) a forward path, comprising a1) an input unit for providing a time-domain electric input signal as digital samples, a2) an analysis filter bank configured to provide a time-frequency representation of said electric input signal, a3) a signal processing unit for processing a signal of the forward path and providing a number of processed channel-signals, B) an onset detector configured to receive said time-domain electric input signal before entering said analysis filter bank, and to provide an onset control signal dependent on a current first order derivative of an envelope thereof, C) a level estimation unit for estimating a current level of said frequency sub-band signals, and comprising c1) a level adjustment unit configured to adjust the current levels of said frequency sub-band signals, and to control said level adjustment in dependence of said onset control signal. The invention may be used in audio devices, e.g. hearing aids.
Abstract:
The application relates to a hearing aid comprising a forward path comprising a) a multitude of input units for providing a multitude of electric input signals INi, i=1, . . . , M, representative of sound, b) a multi input beam former filtering unit for providing a beam formed signal YBF from said multitude of electric input signals, c) a gain unit for applying a hearing aid gain GHA to said beam formed signal YBF, and providing a processed signal, and d) an output unit for providing stimuli perceivable by a user as sound based on said processed signal or a signal derived therefrom. The hearing aid further comprises e) a gain control unit for limiting said hearing aid gain GHA to a modified full-on gain value G′FOG. The multi input beam former filtering unit is configured to apply a current frequency dependent directional gain GDIR,i to each of said multitude of electric input signals INi, and the gain control unit is configured to determine the modified full-on gain value G′FOG in dependence of said current directional gains GDIR,i, i=1, . . . , M, and a previously determined full-on gain value GFOG. Thereby an improved hearing aid is provided. The invention may e.g. be used for hearing instruments, headsets, or active ear protection systems.
Abstract:
The application relates to a filter bank for an audio processing device, e.g. a hearing aid. The filter bank comprises an analysis filter bank comprising a plurality of M first filters hm(n), where m=0, 1, . . . , M−1 is a frequency band index, and whose impulse responses are modulated from a first linear phase prototype filter h(n) with a first predetermined modulation sequence ms1, n being a time index, the first prototype filter h(n) having a first filter length of Lh; a synthesis filter bank comprising a plurality of M second filters gm(n), m=0, 1, . . . , M−1, whose impulse responses are modulated from a second linear phase prototype filter g(n) with a second predetermined modulation sequence ms2, the second prototype filter g(n) having a second filter length of Lg; the plurality of first and second filters being arranged in pairs, each pair forming a frequency channel. the first modulation sequence is a complex or real function of time n, frequency band index m, and a first prototype filter delay τh, the second modulation sequence is a complex or real function of time n, frequency band index m, and a second prototype filter delay τg, the first filter length Lh and the second filter length Lg are both uneven, and the first prototype filter delay τh is equal to (Lh−1)/2 and second prototype filter delay τg, is equal to (Lg−1)/2, and the first and second prototype filter delay τh and τg, are constants of the analysis filter bank and the synthesis filter bank, respectively.
Abstract:
An intrusive binaural speech intelligibility predictor system receives a target signal comprising speech in left and right essentially noise-free and noisy and/or processed versions at left and right ears of a listener. The system comprises a) first, second, third and fourth input units for providing time-frequency representations of said left and right noise-free and noisy/processed versions of the target signal, respectively; b) first and second Equalization-Cancellation stages adapted to receive and relatively time shift and amplitude adjust the left and right noise-free and noisy/processed versions, respectively, and to provide resulting noise-free and noisy/processed signals, respectively; and c) a monaural speech intelligibility predictor unit for providing final binaural speech intelligibility predictor value SI-Measure based on said resulting noise-free and noisy/processed signals. The Equalization-Cancellation stages are adapted to optimize the SI-Measure to indicate a maximum intelligibility of said noisy/processed versions of the target signal by said listener. The invention may e.g. be used in development systems for hearing aids.