Patent Grant
11109166
The present application relates to the field of hearing devices, e.g. hearing aids or ear phones or headsets.
In an aspect of the present application, a hearing device, e.g. a hearing aid, configured to play sound into an ear canal of a user is provided. The hearing device comprises
A forward signal path of the hearing device is defined from an acoustic input to said at least one input transducer to an acoustic output of said output transducer having a forward signal propagation delay τHI of the hearing device is provided. The hearing device further comprises a compensation unit for at least partially cancelling directly propagated sound from said sound field S that is propagated to the ear canal via a direct acoustic propagation path. The compensation unit is configured to predict said directly propagated sound and to play it in opposite phase of said directly propagated sound.
Thereby an improved hearing device may be provided.
The term ‘opposite phase’ is intended to indicate a 180° difference in phase between the two signals (to thereby allow a mutual cancellation of the two signals). The direct acoustic propagation path from the acoustic input to said at least one input transducer to the acoustic output of said output transducer shows a direct propagation delay τdir. The direct propagation delay τdir is typically smaller (e.g. more than 5-10 times smaller) than the forward signal propagation delay τHI of the hearing device, such as much smaller (e.g. more than 100-1000 times smaller) than τHI. The number Q of input transducers, e.g. microphones, may e.g. be 2 or 3 or 4 or more.
The compensation unit may be configured to predict the discrete samples sq(p) in dependence of a delay τcomp of the compensation unit. The compensation unit may be configured to predict the discrete samples sq(p) in dependence of a delay τdir of the direct acoustic propagation path. The compensation unit may be configured to predict the discrete samples sq(p) in dependence of a delay τcomp of the compensation unit and of a delay τdir of the direct acoustic propagation path. The delay τcomp of the compensation unit may be larger than the delay τdir of the direct acoustic propagation path. The compensation unit may be configured to predict the discrete samples sq(p), which are τpred=τcomp−τdir [seconds] in the future. The delay τdir of the direct acoustic propagation path may be frequency dependent. The delay τcomp of the compensation unit may be frequency dependent.
When the compensation unit is integrated in the hearing device, the delay τcomp of the compensation unit may comprise the delay of the electric signal path from the input of the at least one input transducer to the output of the output transducer.
The delay τcomp of the compensation unit (CU) may be taken to mean the delay of the electric signal path from the input of an input transducer (e.g. M2 in
The hearing device may be configured to include frequency-shaping of a transfer function representing said direct acoustic propagation path. The frequency-shaping may e.g. comprise application of a frequency dependent gain (e.g. amplification or attenuation, and/or phase change) to the prediction of said directly propagated sound. The shaping may be represented by an impulse response or a frequency transfer function (indicated by hec in
The compensation unit may be configured to predict the directly propagated sound based on a linear or non-linear prediction algorithm or a combination of a linear and a non-linear prediction algorithm. The compensation unit may be configured to predict future samples of the electric input signal (or a processed version thereof) based on current and past samples.
The compensation unit may be configured to predict the directly propagated sound based on linear or non-linear minimum mean square error (MMSE) prediction.
The compensation unit may be configured to predict the directly propagated sound based on a linear combination of a current and a number P−1 of past samples of the electric input signal, or a processed version thereof, using corresponding weights ai, i=0, 1, . . . , P−1 or using a non-linear function f(.). The directly propagated sound may e.g. be predicted based on a neural network. The neural network may e.g. be trained in an on-line or off-line procedure using data from a variety of acoustic environments, e.g. environments appropriate for a specific user.
The hearing device may comprise a memory wherein parameters of relevance for the prediction of the directly propagated sound can be permanently and/or temporarily stored and accessed by the processor and/or by the compensation unit. Parameters of relevance to prediction include the number of historic samples (P, K) to be considered by the prediction algorithm. The parameters (P, K) should be chosen long enough (K being at least ≥P) to catch the periodicity of the current sound, but not so long that it includes sound (e.g. speech) that is not relevant for the prediction of the present sound (e.g. speech) elements (e.g. representing another sentence or word (or even another speaker)). Further, parameters (P, K) should be chosen as small as possible with a view to (limiting) computational complexity (to limit power consumption).
The compensation unit may be configured to determine the weights ai, i=0, 1, . . . , P−1 or the non-linear function f(.) in an off-line procedure. The weights or the non-linear function f(.) (or an approximation thereof) may be stored in the memory prior to use of the hearing device, e.g. during fitting or manufacturing of the hearing device.
The compensation unit may be configured to determine the weights ai, i=0, 1, . . . , P−1 or the non-linear function f(.) during use of the hearing device.
The compensation unit is configured to determine said weights ai, i=0, 1, . . . , P−1 or said non-linear function f(.) using an optimization procedure involving a cost function. The compensation unit may be configured to determine the weights ai, i=0, 1, . . . , P−1 or the non-linear function f(.) by minimizing a least square prediction error (MMSE). The prediction error may be determined from (preferably recent) historic data of the electric input signal for which a predicted value ŝ and a known value s of the directly propagated sound at said number of past samples of the electric input signal, or a processed version thereof, are known, and possibly stored in a memory of the hearing device (or accessible to the hearing device). The prediction error may be minimized (only) in a selected frequency range (e.g. where speech is known to be present or important for speech intelligibility, see e.g.
The weights ai, i=0, 1, . . . , P−1 or said non-linear function f(.) are updated during use of the hearing device. The weights or the non-linear function may e.g. be continuously updated, e.g. for every sample or for every Nth sample. The weights or the non-linear function may e.g. be updated based on a control signal from on an acoustic environment detector, e.g. when a change in acoustic environment is detected. The update process may e.g. be initiated based on a user input, e.g. via a user interface, e.g. implemented as an APP of a smartphone, e.g. a voice controlled interface, cf. e.g.
The hearing device may comprise a time to time-frequency conversion unit for providing a time-domain input signal in a frequency sub-band representation. The hearing device, e.g. the compensation unit, may be configured to execute the prediction algorithm in all frequency bands k=1, . . . , K, e.g. using a different number Pk of historic values to predict the future value in at least some of the K frequency bands.
The compensation unit may be configured to minimize a prediction error, which is weighted as a function of time and/or frequency. The prediction of the directly propagated sound may be based on a cost function related to a user's perception, e.g. using a perceptual model. The compensation unit may be configured to determine the prediction of the directly propagated sound that leads to the least perceptually objectionable signal. This may be achieved by minimizing a prediction error, which is weighted as a function of time and frequency (e.g. represented by a function g(l,n), where l is a frequency index and n is the time index, cf. e.g. function G(f) in
The hearing device may be configured to execute the prediction algorithm only in selected frequency bands, e.g. frequency bands having the most importance for speech intelligibility, e.g. frequency bands above a low-frequency threshold frequency fth,low and below a high-frequency threshold frequency fth,high. The high frequency threshold frequency fth,high may e.g. be 4 kHz, or 3 kHz, or 2 kHz or smaller, e.g. 1 kHz. The low-frequency threshold frequency fth,low may e.g. be larger than or equal to 100 Hz or 200 Hz, or larger than or equal to 500 Hz (e.g. to take account of the fact that low frequency sound tend to escape through the vent or dome openings (and thus do not disturb the signal at the eardrum significantly)).
The hearing device may comprise an onset detector for identifying transients in the electric input signal and to provide an onset control signal in dependence thereof, wherein the compensation unit is configured to limit or override the currently predicted value of said directly propagated sound whenever the onset control signal indicates that a transient has been detected. In case of a detection of a transient (e.g. defined by a level increase per time unit above a threshold value) at a given point in time, e.g. at time index n, the currently predicted value of the directly propagated sound may be overridden (ignored, e.g. overwritten). Instead a previously predicted value of the directly propagated sound may be used as the current prediction.
The hearing device may comprise at least two input transducers providing corresponding at least two electric input signals and a beamformer filtering unit for providing a spatially filtered signal based on said at least two electric input signals. The hearing device may comprise three or more or four or more input transducers (or be configured to receive corresponding one or two or more electric input signals (or parts thereof, e.g. selected frequency ranges)). The processed signal of the forward path intended for being presented to the user may be based on the spatially filtered signal.
The hearing device may be constituted by or comprise a hearing aid, or any other wearable earpiece, e.g. a headset, an earphone, a headphone, an ear protection device or a combination thereof.
The hearing device may be adapted to provide a frequency dependent gain and/or a level dependent compression and/or a transposition (with or without frequency compression) of one or more frequency ranges to one or more other frequency ranges, e.g. to compensate for a hearing impairment of a user. In an embodiment, the hearing device comprises a signal processor for enhancing the input signals and providing a processed output signal.
The hearing device may comprise an output unit for providing a stimulus perceived by the user as an acoustic signal based on a processed electric signal. In an embodiment, the output unit comprises a number of electrodes of a cochlear implant (for a CI type hearing device) or a vibrator of a bone conducting hearing device. In an embodiment, the output unit comprises an output transducer. In an embodiment, the output transducer comprises a receiver (loudspeaker) for providing the stimulus as an acoustic signal to the user (e.g. in an acoustic (air conduction based) hearing device). In an embodiment, the output transducer comprises a vibrator for providing the stimulus as mechanical vibration of a skull bone to the user (e.g. in a bone-attached or bone-anchored hearing device).
The hearing device may comprise an input unit for providing an electric input signal representing sound. In an embodiment, the input unit comprises an input transducer, e.g. a microphone, for converting an input sound to an electric input signal. In an embodiment, the input unit comprises a wireless receiver for receiving a wireless signal comprising sound and for providing an electric input signal representing said sound.
The hearing device may comprise a directional microphone system adapted to spatially filter sounds from the environment, and thereby enhance perception of a target acoustic source among a multitude of acoustic sources in the local environment of the user wearing the hearing device. In an embodiment, the directional system is adapted to detect (such as adaptively detect) from which direction a particular part of the microphone signal originates. This can be achieved in various different ways as e.g. described in the prior art. In hearing devices, a microphone array beamformer is often used for spatially attenuating background noise sources. Many beamformer variants can be found in the literature. The minimum variance distortionless response (MVDR) beamformer is widely used in microphone array signal processing. Ideally the MVDR beamformer keeps the signals from the target direction (also referred to as the look direction) unchanged, while attenuating sound signals from other directions maximally. The generalized sidelobe canceller (GSC) structure is an equivalent representation of the MVDR beamformer offering computational and numerical advantages over a direct implementation in its original form.
The hearing device may comprise an antenna and transceiver circuitry (e.g. a wireless receiver) for wirelessly receiving a direct electric input signal from another device, e.g. from an entertainment device (e.g. a TV-set), a communication device, a wireless microphone, or another hearing device. In an embodiment, the direct electric input signal represents or comprises an audio signal and/or a control signal and/or an information signal. In an embodiment, the hearing device comprises demodulation circuitry for demodulating the received direct electric input to provide the direct electric input signal representing an audio signal and/or a control signal e.g. for setting an operational parameter (e.g. volume) and/or a processing parameter of the hearing device. In general, a wireless link established by antenna and transceiver circuitry of the hearing device can be of any type. In an embodiment, the wireless link is established between two devices, e.g. between an entertainment device (e.g. a TV) and the hearing device, or between two hearing devices, e.g. via a third, intermediate device (e.g. a processing device, such as a remote control device, a smartphone, etc.). In an embodiment, the wireless link is used under power constraints, e.g. in that the hearing device is or comprises a portable (typically battery driven) device, e.g. a hearing aid. In an embodiment, the wireless link is a link based on near-field communication, e.g. an inductive link based on an inductive coupling between antenna coils of transmitter and receiver parts. In another embodiment, the wireless link is based on far-field, electromagnetic radiation. In an embodiment, the communication via the wireless link is arranged according to a specific modulation scheme. In an embodiment, the wireless link is based on a standardized or proprietary technology. In an embodiment, the wireless link is based on Bluetooth technology (e.g. Bluetooth Low-Energy technology).
The hearing device may be a portable device, e.g. a device comprising a local energy source, e.g. a battery, e.g. a rechargeable battery.
The hearing device may comprise a forward or signal path between an input unit (e.g. an input transducer, such as a microphone or a microphone system and/or direct electric input (e.g. a wireless receiver)) and an output unit, e.g. an output transducer. In an embodiment, the signal processor is located in the forward path. In an embodiment, the signal processor is adapted to provide a frequency dependent gain according to a user's particular needs. In an embodiment, the hearing device comprises an analysis path comprising functional components for analyzing the input signal (e.g. determining a level, a modulation, a type of signal, an acoustic feedback estimate, etc.). In an embodiment, some or all signal processing of the analysis path and/or the signal path is conducted in the frequency domain. In an embodiment, some or all signal processing of the analysis path and/or the signal path is conducted in the time domain.
In an embodiment, an analogue electric signal representing an acoustic signal is converted to a digital audio signal in an analogue-to-digital (AD) conversion process, where the analogue signal is sampled with a predefined sampling frequency or rate fs, fs being e.g. in the range from 8 kHz to 48 kHz (adapted to the particular needs of the application) to provide digital samples xn, (or x[n]) at discrete points in time tn (or n), each audio sample representing the value of the acoustic signal at tn by a predefined number Nb of bits, Nb being e.g. in the range from 1 to 48 bits, e.g. 24 bits. Each audio sample is hence quantized using Nb bits (resulting in 2Nb different possible values of the audio sample). A digital sample x has a length in time of 1/fs, e.g. 50 μs, for fs=20 kHz. In an embodiment, a number of audio samples are arranged in a time frame. In an embodiment, a time frame comprises 64 or 128 audio data samples. Other frame lengths may be used depending on the practical application.
The hearing device may comprise an analogue-to-digital (AD) converter to digitize an analogue input (e.g. from an input transducer, such as a microphone) with a predefined sampling rate, e.g. 20 kHz. In an embodiment, the hearing devices comprise a digital-to-analogue (DA) converter to convert a digital signal to an analogue output signal, e.g. for being presented to a user via an output transducer.
The hearing device, e.g. the microphone unit, and or the transceiver unit may comprise a TF-conversion unit (e.g. an analysis filter bank) for providing a time-frequency representation of an input signal. In an embodiment, the time-frequency representation comprises an array or map of corresponding complex or real values of the signal in question in a particular time and frequency range. In an embodiment, the TF conversion unit comprises a filter bank for filtering a (time varying) input signal and providing a number of (time varying) output signals each comprising a distinct frequency range of the input signal. In an embodiment, the TF conversion unit comprises a Fourier transformation unit for converting a time variant input signal to a (time variant) signal in the (time-) frequency domain. In an embodiment, the frequency range considered by the hearing device from a minimum frequency fmin to a maximum frequency fmax comprises a part of the typical human audible frequency range from 20 Hz to 20 kHz, e.g. a part of the range from 20 Hz to 12 kHz. Typically, a sample rate fs is larger than or equal to twice the maximum frequency fmax, fs≥2fmax. In an embodiment, a signal of the forward and/or analysis path of the hearing device is split into a number NI of frequency bands (e.g. of uniform width), where NI is e.g. larger than 5, such as larger than 10, such as larger than 50, such as larger than 100, such as larger than 500, at least some of which are processed individually. In an embodiment, the hearing device is/are adapted to process a signal of the forward and/or analysis path in a number NP of different frequency channels (NP≤NI). The frequency channels may be uniform or non-uniform in width (e.g. increasing in width with frequency), overlapping or non-overlapping.
The hearing device may comprise a number of detectors configured to provide status signals relating to a current physical environment of the hearing device (e.g. the current acoustic environment), and/or to a current state of the user wearing the hearing device, and/or to a current state or mode of operation of the hearing device. Alternatively or additionally, one or more detectors may form part of an external device in communication (e.g. wirelessly) with the hearing device. An external device may e.g. comprise another hearing device, a remote control, and audio delivery device, a telephone (e.g. a smartphone), an external sensor, etc.
One or more of the number of detectors may operate on the full band signal (time domain). In an embodiment, one or more of the number of detectors operate(s) on band split signals ((time-) frequency domain), e.g. in a limited number of frequency bands.
The number of detectors may comprise a level detector for estimating a current level of a signal of the forward path. In an embodiment, the predefined criterion comprises whether the current level of a signal of the forward path is above or below a given (L-) threshold value. In an embodiment, the level detector operates on the full band signal (time domain) In an embodiment, the level detector operates on band split signals ((time-) frequency domain).
In a particular embodiment, the hearing device comprises a voice detector (VD) for estimating whether or not (or with what probability) an input signal comprises a voice signal (at a given point in time). A voice signal is in the present context taken to include a speech signal from a human being. It may also include other forms of utterances generated by the human speech system (e.g. singing). In an embodiment, the voice detector unit is adapted to classify a current acoustic environment of the user as a VOICE or NO-VOICE environment. This has the advantage that time segments of the electric microphone signal comprising human utterances (e.g. speech) in the user's environment can be identified, and thus separated from time segments only (or mainly) comprising other sound sources (e.g. artificially generated noise). In an embodiment, the voice detector is adapted to detect as a VOICE also the user's own voice. Alternatively, the voice detector is adapted to exclude a user's own voice from the detection of a VOICE.
The hearing device may comprise an own voice detector for estimating whether or not (or with what probability) a given input sound (e.g. a voice, e.g. speech) originates from the voice of the user of the device or system. The own voice detector may be configured to provide an own voice control signal indicative of whether or not (or with what probability) a given input sound to the hearing device originates from the voice of the user of the device or system. In an embodiment, the hearing device or system, e.g. a microphone system of the hearing device or system, is adapted to be able to differentiate between a user's own voice and another person's voice and possibly NON-voice sounds. The hearing device or system (e.g. the compensation unit) may be configured to control parameters of the prediction algorithm (e.g. delays and/or frequency shaping) in dependence on detected own voice, such that the hearing device or system (e.g. the compensation unit) copes differently with external sounds compared to sound from the user's mouth (own voice'). The hearing device or system may e.g. be configured to provide more or less aggressive cancellation of the direct propagated sound in dependence of the own voice control signal).
The number of detectors may comprise a movement detector, e.g. an acceleration sensor. In an embodiment, the movement detector is configured to detect movement of the user's facial muscles and/or bones, e.g. due to speech or chewing (e.g. jaw movement) and to provide a detector signal indicative thereof.
The hearing device may comprise a classification unit configured to classify the current situation based on input signals from (at least some of) the detectors, and possibly other inputs as well. In the present context ‘a current situation’ is taken to be defined by one or more of
a) the physical environment (e.g. including the current electromagnetic environment, e.g. the occurrence of electromagnetic signals (e.g. comprising audio and/or control signals) intended or not intended for reception by the hearing device, or other properties of the current environment than acoustic);
b) the current acoustic situation (input level, feedback, etc.), and
c) the current mode or state of the user (movement, temperature, cognitive load, etc.);
d) the current mode or state of the hearing device (program selected, time elapsed since last user interaction, etc.) and/or of another device in communication with the hearing device.
The hearing device may comprise an acoustic (and/or mechanical) feedback suppression system. In an embodiment, the feedback suppression system comprises a feedback estimation unit for providing a feedback signal representative of an estimate of the acoustic feedback path, and a combination unit, e.g. a subtraction unit, for subtracting the feedback signal from a signal of the forward path (e.g. as picked up by an input transducer of the hearing device).
The hearing device may further comprise other relevant functionality for the application in question, e.g. compression, noise reduction, etc.
The hearing device may comprise or consist of a listening device, e.g. a hearing aid, e.g. a hearing instrument, e.g. a hearing instrument adapted for being located at the ear or fully or partially in the ear canal of a user, e.g. a headset, an earphone, an ear protection device or a combination thereof. In an embodiment, the hearing assistance system comprises a speakerphone (comprising a number of input transducers and a number of output transducers, e.g. for use in an audio conference situation), e.g. comprising a beamformer filtering unit, e.g. providing multiple beamforming capabilities.
In an aspect, use of a hearing device as described above, in the ‘detailed description of embodiments’ and in the claims, is moreover provided. In an embodiment, use is provided in a system comprising audio distribution. In an embodiment, use is provided in a system comprising one or more hearing aids (e.g. hearing instruments), headsets, ear phones, active ear protection systems, etc., e.g. in handsfree telephone systems, teleconferencing systems (e.g. including a speakerphone), public address systems, karaoke systems, classroom amplification systems, etc.
In an aspect, a method of operating a hearing device, e.g. a hearing aid, comprising an ITE-part adapted for being located in or at an ear canal of a user and configured to pick up sound from a sound field around the user and to play processed sound into said ear canal is provided. The hearing device has a propagation delay τHI defined by the processing delay for processing said sound in a forward signal path from an acoustic input to an acoustic or vibrational output of the hearing device. The method comprises
The method further comprises at least partially compensating for directly propagated sound from said sound field that is propagated to the ear canal via a direct acoustic propagation path by
It is intended that some or all of the structural features of the device described above, in the ‘detailed description of embodiments’ or in the claims can be combined with embodiments of the method, when appropriately substituted by a corresponding process and vice versa. Embodiments of the method have the same advantages as the corresponding devices.
In an aspect, a tangible computer-readable medium storing a computer program comprising program code means for causing a data processing system to perform at least some (such as a majority or all) of the steps of the method described above, in the ‘detailed description of embodiments’ and in the claims, when said computer program is executed on the data processing system is furthermore provided by the present application.
By way of example, and not limitation, such computer-readable media can comprise RAM, ROM, EEPROM, CD-ROM or other optical disk storage, magnetic disk storage or other magnetic storage devices, or any other medium that can be used to carry or store desired program code in the form of instructions or data structures and that can be accessed by a computer. Disk and disc, as used herein, includes compact disc (CD), laser disc, optical disc, digital versatile disc (DVD), floppy disk and Blu-ray disc where disks usually reproduce data magnetically, while discs reproduce data optically with lasers. Combinations of the above should also be included within the scope of computer-readable media. In addition to being stored on a tangible medium, the computer program can also be transmitted via a transmission medium such as a wired or wireless link or a network, e.g. the Internet, and loaded into a data processing system for being executed at a location different from that of the tangible medium.
A computer program (product) comprising instructions which, when the program is executed by a computer, cause the computer to carry out (steps of) the method described above, in the ‘detailed description of embodiments’ and in the claims is furthermore provided by the present application.
In an aspect, a data processing system comprising a processor and program code means for causing the processor to perform at least some (such as a majority or all) of the steps of the method described above, in the ‘detailed description of embodiments’ and in the claims is furthermore provided by the present application.
In a further aspect, a hearing system comprising a hearing device as described above, in the ‘detailed description of embodiments’, and in the claims, AND an auxiliary device is moreover provided.
In an embodiment, the hearing system is adapted to establish a communication link between the hearing device and the auxiliary device to provide that information (e.g. control and status signals, possibly audio signals) can be exchanged or forwarded from one to the other.
In an embodiment, the hearing system comprises an auxiliary device, e.g. a remote control, a smartphone, or other portable or wearable electronic device, such as a smartwatch or the like.
In an embodiment, the auxiliary device is or comprises a remote control for controlling functionality and operation of the hearing device(s). In an embodiment, the function of a remote control is implemented in a smartphone, the smartphone possibly running an APP allowing to control the functionality of the audio processing device via the smartphone (the hearing device(s) comprising an appropriate wireless interface to the smartphone, e.g. based on Bluetooth or some other standardized or proprietary scheme).
In an embodiment, the auxiliary device is or comprises an audio gateway device adapted for receiving a multitude of audio signals (e.g. from an entertainment device, e.g. a TV or a music player, a telephone apparatus, e.g. a mobile telephone or a computer, e.g. a PC) and adapted for selecting and/or combining an appropriate one of the received audio signals (or combination of signals) for transmission to the hearing device.
In an embodiment, the auxiliary device is or comprises another hearing device. In an embodiment, the hearing system comprises two hearing devices adapted to implement a binaural hearing system, e.g. a binaural hearing aid system.
In a further aspect, a non-transitory application, termed an APP, is furthermore provided by the present disclosure. The APP comprises executable instructions configured to be executed on an auxiliary device to implement a user interface for a hearing device or a hearing system described above in the ‘detailed description of embodiments’, and in the claims. In an embodiment, the APP is configured to run on cellular phone, e.g. a smartphone, or on another portable device allowing communication with said hearing device or said hearing system. The hearing device or system may be configured to allow an active noise suppression mode of operation to be selected and/or configured from the user interface/APP. Parameters of the prediction algorithm may be configured via the user interface/APP.
In the present context, a ‘hearing device’ refers to a device, such as a hearing aid, e.g. a hearing instrument, or an active ear-protection device, or other audio processing device, which is adapted to improve, augment and/or protect the hearing capability of a user by receiving acoustic signals from the user's surroundings, generating corresponding audio signals, possibly modifying the audio signals and providing the possibly modified audio signals as audible signals to at least one of the user's ears. A ‘hearing device’ further refers to a device such as an earphone or a headset adapted to receive audio signals electronically, possibly modifying the audio signals and providing the possibly modified audio signals as audible signals to at least one of the user's ears. Such audible signals may e.g. be provided in the form of acoustic signals radiated into the user's outer ears, acoustic signals transferred as mechanical vibrations to the user's inner ears through the bone structure of the user's head and/or through parts of the middle ear as well as electric signals transferred directly or indirectly to the cochlear nerve of the user.
The hearing device may be configured to be worn in any known way, e.g. as a unit arranged behind the ear with a tube leading radiated acoustic signals into the ear canal or with an output transducer, e.g. a loudspeaker, arranged close to or in the ear canal, as a unit entirely or partly arranged in the pinna and/or in the ear canal, as a unit, e.g. a vibrator, attached to a fixture implanted into the skull bone, as an attachable, or entirely or partly implanted, unit, etc. The hearing device may comprise a single unit or several units communicating electronically with each other. The loudspeaker may be arranged in a housing together with other components of the hearing device, or may be an external unit in itself (possibly in combination with a flexible guiding element, e.g. a dome-like element).
More generally, a hearing device comprises an input transducer for receiving an acoustic signal from a user's surroundings and providing a corresponding input audio signal and/or a receiver for electronically (i.e. wired or wirelessly) receiving an input audio signal, a (typically configurable) signal processing circuit (e.g. a signal processor, e.g. comprising a configurable (programmable) processor, e.g. a digital signal processor) for processing the input audio signal and an output unit for providing an audible signal to the user in dependence on the processed audio signal. The signal processor may be adapted to process the input signal in the time domain or in a number of frequency bands. In some hearing devices, an amplifier and/or compressor may constitute the signal processing circuit. The signal processing circuit typically comprises one or more (integrated or separate) memory elements for executing programs and/or for storing parameters used (or potentially used) in the processing and/or for storing information relevant for the function of the hearing device and/or for storing information (e.g. processed information, e.g. provided by the signal processing circuit), e.g. for use in connection with an interface to a user and/or an interface to a programming device. In some hearing devices, the output unit may comprise an output transducer, such as e.g. a loudspeaker for providing an air-borne acoustic signal or a vibrator for providing a structure-borne or liquid-borne acoustic signal. In some hearing devices, the output unit may comprise one or more output electrodes for providing electric signals (e.g. a multi-electrode array for electrically stimulating the cochlear nerve). In an embodiment, the hearing device comprises a speakerphone (comprising a number of input transducers and a number of output transducers, e.g. for use in an audio conference situation).
In some hearing devices, the vibrator may be adapted to provide a structure-borne acoustic signal transcutaneously or percutaneously to the skull bone. In some hearing devices, the vibrator may be implanted in the middle ear and/or in the inner ear. In some hearing devices, the vibrator may be adapted to provide a structure-borne acoustic signal to a middle-ear bone and/or to the cochlea. In some hearing devices, the vibrator may be adapted to provide a liquid-borne acoustic signal to the cochlear liquid, e.g. through the oval window. In some hearing devices, the output electrodes may be implanted in the cochlea or on the inside of the skull bone and may be adapted to provide the electric signals to the hair cells of the cochlea, to one or more hearing nerves, to the auditory brainstem, to the auditory midbrain, to the auditory cortex and/or to other parts of the cerebral cortex.
A hearing device, e.g. a hearing aid, may be adapted to a particular user's needs, e.g. a hearing impairment. A configurable signal processing circuit of the hearing device may be adapted to apply a frequency and level dependent compressive amplification of an input signal. A customized frequency and level dependent gain (amplification or compression) may be determined in a fitting process by a fitting system based on a user's hearing data, e.g. an audiogram, using a fitting rationale (e.g. adapted to speech). The frequency and level dependent gain may e.g. be embodied in processing parameters, e.g. uploaded to the hearing device via an interface to a programming device (fitting system), and used by a processing algorithm executed by the configurable signal processing circuit of the hearing device.
A ‘hearing system’ refers to a system comprising one or two hearing devices, and a ‘binaural hearing system’ refers to a system comprising two hearing devices and being adapted to cooperatively provide audible signals to both of the user's ears. Hearing systems or binaural hearing systems may further comprise one or more ‘auxiliary devices’, which communicate with the hearing device(s) and affect and/or benefit from the function of the hearing device(s). Auxiliary devices may be e.g. remote controls, audio gateway devices, mobile phones (e.g. smartphones), or music players. Hearing devices, hearing systems or binaural hearing systems may e.g. be used for compensating for a hearing-impaired person's loss of hearing capability, augmenting or protecting a normal-hearing person's hearing capability and/or conveying electronic audio signals to a person. Hearing devices or hearing systems may e.g. form part of or interact with public-address systems, active ear protection systems, handsfree telephone systems, car audio systems, entertainment (e.g. karaoke) systems, teleconferencing systems, classroom amplification systems, etc.
Embodiments of the disclosure may e.g. be useful in applications such as applications.
The aspects of the disclosure may be best understood from the following detailed description taken in conjunction with the accompanying figures. The figures are schematic and simplified for clarity, and they just show details to improve the understanding of the claims, while other details are left out. Throughout, the same reference numerals are used for identical or corresponding parts. The individual features of each aspect may each be combined with any or all features of the other aspects. These and other aspects, features and/or technical effect will be apparent from and elucidated with reference to the illustrations described hereinafter in which:
The figures are schematic and simplified for clarity, and they just show details which are essential to the understanding of the disclosure, while other details are left out. Throughout, the same reference signs are used for identical or corresponding parts.
Further scope of applicability of the present disclosure will become apparent from the detailed description given hereinafter. However, it should be understood that the detailed description and specific examples, while indicating preferred embodiments of the disclosure, are given by way of illustration only. Other embodiments may become apparent to those skilled in the art from the following detailed description.
The detailed description set forth below in connection with the appended drawings is intended as a description of various configurations. The detailed description includes specific details for the purpose of providing a thorough understanding of various concepts. However, it will be apparent to those skilled in the art that these concepts may be practiced without these specific details. Several aspects of the apparatus and methods are described by various blocks, functional units, modules, components, circuits, steps, processes, algorithms, etc. (collectively referred to as “elements”). Depending upon particular application, design constraints or other reasons, these elements may be implemented using electronic hardware, computer program, or any combination thereof.
The electronic hardware may include microprocessors, microcontrollers, digital signal processors (DSPs), field programmable gate arrays (FPGAs), programmable logic devices (PLDs), gated logic, discrete hardware circuits, and other suitable hardware configured to perform the various functionality described throughout this disclosure. Computer program shall be construed broadly to mean instructions, instruction sets, code, code segments, program code, programs, subprograms, software modules, applications, software applications, software packages, routines, subroutines, objects, executables, threads of execution, procedures, functions, etc., whether referred to as software, firmware, middleware, microcode, hardware description language, or otherwise.
The present application relates to the field of hearing devices, e.g. hearing aids or ear phones.
Hearing aids pick up sound from microphone(s) (e.g., located behind the pinna of the user), process the sound (e.g., amplifying and compressing), and present the sound to the eardrum of the listener, typically via a loudspeaker located close to the eardrum of the listener (cf. e.g.
With open-fitting hearing aids (i.e. hearing aids comprising an earpiece (located at or in the ear canal), which does not tightly seal the opening between ear drum and environment sound, cf. e.g.
In other words, a version of the acoustic input s(t) propagated via a direct acoustic propagation path (DAP in
Specifically, the presence of the signal component s″(t−τdir) will cause a perceptually annoying comb-filtering effect, which is unwanted.
While presented here in an open-fitting hearing instrument context (cf.
The present disclosure deals with methods for estimating and cancelling the signal component s″(t−τdir).
In
As comb-filtering mainly is perceived at lower frequencies, the signal component to be cancelled may only be estimated at frequencies below a certain threshold, e.g. below 2 kHz or below 1 kHz.
Typically, the compensation unit (CU) will be physically part of the hearing device (HD), e.g. hearing aid (so in this sense, 2 is misleading). However, whereas the traditional signal path in the hearing device may have a delay of τHI˜8 ms, primarily due to the presence of analysis and synthesis filter banks, the compensation unit (CU) (integrated in the hearing device, cf. e.g. processing part eCU in
In the following (and throughout the present disclosure), for simplicity, it is assumed that the transfer function T(s) of the acoustic channel (DPA) simply introduces a pure delay of τdir. It is, however, straight-forward to generalize the exposition to allow for a more general transfer function T(s), which also performs some sort of frequency-shaping (e.g. frequency dependent gain (amplification or attenuation, and/or phase change)). With typical values of τcomp˜3 ms and τdir˜50 μs, it is clear that in order to estimate the signal s″(t−τdir) correctly, the compensation unit (CU) needs to predict signal samples which are τpred=τcomp−τdir seconds in the future, compared to the signal samples that the compensation unit can output. With the typical values above, τpred could be in the order of ˜2950 μs. e.g. by making predictions on a per frequency band level (=>τdir per frequency band)
Fortunately, many sound signals of interest to humans, e.g. speech, music, etc., have a structure that allows such prediction (including a certain (short-time) periodicity (correlation) of the sound signal). Several methods can be envisioned to perform the prediction.
ŝ(n+τpred)=Σi=0P−1ais(n−i). Eq. (1)
where n is an index representing (present) time, and ŝ(n+τpred) represents an estimate (prediction) of signal s at time τpred later than the present time n (i.e. at time t=n+τpred), where i is a time sample index), P is the number of previous sample values that are included in the prediction, and ai represents a weight in the prediction value of the ith previous sample, and s(n−i) is the value of the ith previous sample of the input signal (i.e. the value of s i time units before the present time n). This is schematically illustrated in
The coefficients ai of eq. (1) above related to linear prediction are generally time-varying and may be estimated based on past samples s(n−K+1), . . . , s(n).
More generally, the future signal sample of interest may be based on a linear or non-linear combination of the past and current signal samples available to the compensation unit (e.g. eCU/CU in
ŝ(n+τpred)=f(s(n−P+1), . . . ,s(n)),
where f(.) is a pre-determined or adaptive mapping of the past/current signal samples to the future samples.
The time-varying coefficients ai or more generally the time-varying function f(.), may be estimated from past signal samples. Specifically, based on past samples, s(n′−K+1), . . . , s(n′), s(n), where n=n′+τpred (see
where sample index n′ is chosen such that past s(n′−K+1), . . . , s(n′) and current s(n) samples are used in the minimization. This minimization problem is quadratic in the unknown coefficients ai, i=0, . . . , P−1, which means that a well-known closed-form expression for the solution exists. New coefficients ai, i=0, . . . , P−1 may be determined/adapted as a function of time, e.g., whenever a new sample s(n) is available (as illustrated in
whenever new samples s(n) are available (or f(.) may be updated at a slower rate).
The number of past samples (defined by parameters P and K, respectively, K≥P) should be chosen with a view to the nature of speech (or other sound signals of relevance to the user) large enough to catch characteristics of the present sound signal, e.g. speech (e.g. a time period T0 of a fundamental frequency F0 of current speech), but not so long that it includes speech (e.g. from another source) that is not relevant for the prediction of present speech elements. In practice P, K should be chosen to cover a time range smaller than 1 s, e.g. smaller than 100 ms, e.g. to cover a time range between 1 ms and 50 ms, e g smaller than 25 ms. Preferably, P, K should be chosen to include a time range spanning one to three time periods of the fundamental frequency of interest. Fundamental frequencies for male persons are e.g. typically in the range from 85 Hz to 180 Hz. If e.g. the fundamental frequency F0 is 100 Hz (time period T0=1/F0=10 ms), and if the sampling frequency fs of an analogue to digital converter of a microphone signal is 20 kHz providing a sampling period Ts (time between samples) of Ts=1/fs=0.050 ms, then a time period of the fundamental frequency would correspond to 10 ms/0.050 ms=200 samples. In such case, P (and K) should preferably be chosen to be at least 200, e.g. in the range between 200 and 1000 samples. In general P, K should be chosen to be larger than the number of samples corresponding to a time period of the maximum fundamental frequency expected to occur in the sound signals considered. On the other hand, P, K should be chosen to be small enough to predominantly cover sound segments for which the periodicity (or correlation) is relatively constant, e.g. within a predetermined variance threshold. The system may be configured to determine a fundamental frequency of the current sound signal and to dynamically determine an appropriate value for the parameter P. In any case P (and K) may be limited by a computing capacity (available power budget) of the device in question, here a hearing device, e.g. a hearing aid.
For example, f(.) may be realized in terms of a neural network, whose parameters/weights are updated/adapted e.g., by applying the backpropagation algorithm in a supervised learning setup, where the input to the network is past samples, e.g., s(n′−P+1), . . . s(n′), and desired output of the network are signal samples s(n′+τpred), τpred samples ahead in time—obviously, n′ is chosen such that only past and current (=the correctly ‘predicted’)) samples are involved in the weight updates.
Backpropagation may not always be updated, e.g. only be updated if the error is above a predetermined threshold. Alternatively, the update may run at an auxiliary device. Whether signals are transmitted to the auxiliary device may depend on the size of the error signal.
It is obviously also possible to estimate the linear/non-linear mapping in an offline optimization procedure, and then maintaining a fixed map f(.) during operation of the hearing instrument.
Several extensions of the basic idea exist:
where * denotes linear convolution, and where h(l) is the impulse response of a frequency-shaping filter, e.g., a bandpass filter that emphasizes spectral regions where prediction accuracy is of highest interest, and where l is a frequency index.
where g(l,n) denotes a time-varying impulse response. The purpose of g(l,n) is to emphasize perceptually important spectro-temporal regions and de-emphasize less important or irrelevant regions. In particular, g(l,n) may be found by observing that the total signal presented to the eardrum of the user consists of the direct sound and the prediction error signal ŝ(n+τpred)−s(n+τpred). The presence of the direct sound is going to mask the prediction error signal at certain frequencies (i.e., making the imperceptible (e.g. below a first threshold frequency, e.g. 100 Hz) and, hence, un-important at those frequencies). This masking effect may be estimated by applying a masking model (or more generally, an auditory model) to the direct signal (which is accessible in the hearing instrument). From the masking model, a masked threshold may be computed, from which a time-varying, spectral weighting function g(l,n) may be computed. In addition, S may be normalized with respect to the input level such that the error does not depend on the input level. An exemplary frequency dependent masking function G(f), f representing frequency, is schematically illustrated in
To minimize leakage of sound (played by the hearing device towards the ear drum of the user) from the ear canal, a good mechanical contact between the housing of the hearing device and the Skin/tissue of the ear canal is aimed at. In an attempt to minimize such leakage, the housing of the ITE-part may be customized to the ear of a particular user.
The hearing device (HD) comprises a number Q of microphones Mq, i=1, . . . , Q, here two (Q=2). The two microphones (M1, M2) are located in the housing with a predefined distance d between them, e.g. 8-10 mm, on a part of the surface of the housing that faces the environment when the hearing device is operationally mounted in or at the ear of the user. The microphones (M1, M2) are e.g. located on the housing to have their microphone axis (an axis through the centre of the two microphones) point in a forward direction relative to the user, e.g. a look direction of the user (as e.g. defined by the nose of the user, e.g. substantially in a horizontal plane), when the hearing device is mounted in or at the ear of the user. The microphones are configured to convert sound (S1, S2) received from a sound field S around the user at their respective locations to respective (analogue) electric signals (s1, s2) representing the sound. The microphones are coupled to respective analogue to digital converters (AD) to provide the respective (analogue) electric signals (s1, s2) as digitized signals (s1, s2). The digitized signals may further be coupled to respective filter banks to provide each of the electric input signals (time domain signals) as frequency sub-band signals (frequency domain signals). The (digitized) electric input signals (s1, s2) are fed to a signal processor (SPU) for processing the audio signals (s1, s2), e.g. including one or more of spatial filtering (beamforming), (e.g. single channel) noise reduction, compression (frequency and level dependent amplification/attenuation according to a user's needs, e.g. hearing impairment), spatial cue preservation/restoration, etc. The signal processor (SPU) may e.g. comprise the mentioned filter banks (e.g. analysis as well as synthesis filter banks). The signal processor (SPU) is configured to provide a processed signal ŝ comprising a representation of the sound field S (e.g. including an estimate of a target signal therein). The processed signal ŝ is fed to an output transducer (here a loudspeaker (SPK), e.g. via a digital to analogue converter (DA), for conversion of a processed (electric) signal sout (or analogue version sout) to a sound signal Sout.
In a mode of operation, the hearing device is configured to couple the output ŝ of the signal processor (SPU) (directly) to the loudspeaker (SPK) (possibly via the DA-converter (DA)).
The hearing device may e.g. comprise a venting channel (Vent) configured to minimize the effect of occlusion (when the user speaks). In addition to allowing an (un-intended) acoustic propagation path from a residual volume (cf. Res. Vol in
In a mode of operation, where active noise suppression (ANS) is activated, the hearing device is configured to couple the output ŝ of the signal processor (SPU) to the loudspeaker (SPK) via a combination unit (here sum unit ‘+’). In the sum unit (‘+’), an (equivalent electric) estimate ŝdir of a directly propagated acoustic signal Sdir (e.g. through a vent (Vent) or other leakage channels, e.g. between the housing and the walls of the ear canal) is subtracted from the output ŝ of the signal processor (SPU) to provide a resulting output signal sout which is fed to the loudspeaker (SPK) (possibly via the DA-converter (DA)). In this ‘ANS-mode’, the acoustic signal Sout provided by loudspeaker (SPK) represents an estimate of sound Ŝ in the environment sound field S (at least partially) compensated for directly propagated sound Sdir reaching the residual volume (Res. Vol). The resulting sound SED at the eardrum is then equal to the (possibly enhanced, e.g. amplified) estimate Ŝ of the environment sound S plus the directly propagated sound Sdir, minus an estimate Ŝdir of the directly propagated sound Sdir. The estimate ŝdir of a directly propagated acoustic signal Sdir is preferably shaped to match the shaping by the acoustic propagation path. Ideally (when Ŝdir=Sdir), SED=Ŝ, i.e. the directly propagated sound has been compensated).
The input unit (e.g. comprising one or more transducer(s), e.g. microphone(s), appropriate AD-converters, analysis filter banks, etc., as the case may be), the signal processor (SPU, e.g. comprising appropriate analysis and synthesis filter banks, as the case may be, and one or more processing algorithms for enhancing the input audio signal(s)), the combination unit (‘+’), and the output unit (e.g. comprising appropriate digital to analogue converter, output transducer, e.g. loudspeaker, etc., as the case may be) form part of or constitute a forward signal path of the hearing device. The forward signal path is configured to pick up sound, process the sound and provided a processed version of the sound to the user, e.g. the user's ear drum. The forward path of the hearing device (HD) has (at a given point in time) a propagation delay τHI from an acoustic input to the acoustic output. The propagation delay τHI of the hearing device (HD) may be predetermined or adaptively determined. The propagation delay τHI of the hearing device (HD) is a sum of the processing delay of each of the elements in the forward path (e.g. input unit, signal processor, combination unit, output unit, cf. e.g.
The hearing device comprises a compensation unit (CU, cf. dashed enclosure denoted CU (τdir), τdir denoting the delay of the compensation unit CU). The compensation unit (CU) comprises a processing part (eCU) coupled to at least one of the input transducers (here M2) and for providing an (equivalent electric) estimate ŝdir of the directly propagated sound Sdir. The compensation unit has (or is coupled to) a memory (MEM) wherein relevant information about the delay of the forward signal path of the hearing device is stored/accessible (e.g. the processing delays of the individual parts or elements as mentioned above). The compensation unit (CU), e.g. the processing part (eCU), may e.g. implement a prediction algorithm as described above in connection with
The hearing device comprises an energy source, e.g. a battery (BAT), e.g. a rechargeable battery, for energizing the components of the device.
In the embodiment of a hearing device in
The substrate (SUB) further comprises a configurable signal processor (DSP, e.g. a digital signal processor), e.g. including a processor for applying a frequency and level dependent gain, e.g. providing beamforming, noise reduction, filter bank functionality, and other digital functionality of a hearing device, e.g. implementing a compensation unit, according to the present disclosure (as e.g. discussed in connection with
The hearing device (HD) further comprises an output unit (e.g. an output transducer) providing stimuli perceivable by the user as sound based on a processed audio signal from the processor or a signal derived therefrom. In the embodiment of a hearing device in
The electric input signals (from input transducers MBTE1, MBTE2, MITE) may be processed in the time domain or in the (time-) frequency domain (or partly in the time domain and partly in the frequency domain as considered advantageous for the application in question).
The embodiments of a hearing device (HD) exemplified in
In an embodiment, at least some of the calculations related to active noise suppression (e.g. sound prediction) are performed in the auxiliary device. In another embodiment, the calculations are performed in the left and/or right hearing devices. In the latter case the system may be configured to exchange the data between the auxiliary device and the hearing device(s). The hearing device (HD1, HD2) are shown in
In an analogue to digital (AD) process, a digital sample x(n) has a length in time of 1/fs, e.g. 50 μs, for fs=20 kHz. A number of (audio) samples Ns are e.g. arranged in a time frame, as schematically illustrated in the lower part of
A time frame of an electric signal may e.g. comprise a number Ns of consecutive samples, e.g. 64, (written as vector xm) of the digitized electric signal representing sound, m being a time index, cf. e.g.
The electric input signal(s) representing sound may be provided as a number of frequency sub-band signals. The frequency sub-bands signals may e.g. be provided by an analysis filter bank, e.g. based on a number of band-pass filters, or on a Fourier transform algorithm as indicated above (e.g. by consecutively extracting respective magnitude spectra from the Fourier transformed data).
As indicated in
It is intended that the structural features of the devices described above, either in the detailed description and/or in the claims, may be combined with steps of the method, when appropriately substituted by a corresponding process.
As used, the singular forms “a,” “an,” and “the” are intended to include the plural forms as well (i.e. to have the meaning “at least one”), unless expressly stated otherwise. It will be further understood that the terms “includes,” “comprises,” “including,” and/or “comprising,” when used in this specification, specify the presence of stated features, integers, steps, operations, elements, and/or components, but do not preclude the presence or addition of one or more other features, integers, steps, operations, elements, components, and/or groups thereof. It will also be understood that when an element is referred to as being “connected” or “coupled” to another element, it can be directly connected or coupled to the other element but an intervening element may also be present, unless expressly stated otherwise. Furthermore, “connected” or “coupled” as used herein may include wirelessly connected or coupled. As used herein, the term “and/or” includes any and all combinations of one or more of the associated listed items. The steps of any disclosed method is not limited to the exact order stated herein, unless expressly stated otherwise.
It should be appreciated that reference throughout this specification to “one embodiment” or “an embodiment” or “an aspect” or features included as “may” means that a particular feature, structure or characteristic described in connection with the embodiment is included in at least one embodiment of the disclosure. Furthermore, the particular features, structures or characteristics may be combined as suitable in one or more embodiments of the disclosure. The previous description is provided to enable any person skilled in the art to practice the various aspects described herein. Various modifications to these aspects will be readily apparent to those skilled in the art, and the generic principles defined herein may be applied to other aspects.
The claims are not intended to be limited to the aspects shown herein, but is to be accorded the full scope consistent with the language of the claims, wherein reference to an element in the singular is not intended to mean “one and only one” unless specifically so stated, but rather “one or more.” Unless specifically stated otherwise, the term “some” refers to one or more.
Accordingly, the scope should be judged in terms of the claims that follow.
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