METHODS AND APPARATUSES FOR SETTING A HEARING AID TO AN OMNIDIRECTIONAL MICROPHONE MODE OR A DIRECTIONAL MICROPHONE MODE
20170230761 · 2017-08-10
Assignee
Inventors
Cpc classification
H04R25/40
ELECTRICITY
H04R25/407
ELECTRICITY
H04R2225/41
ELECTRICITY
International classification
Abstract
A method of automatic switching between omnidirectional (OMNI) and directional (DIR) microphone modes in a binaural hearing aid comprising a first microphone system, a second microphone system, where the first microphone system is adapted to be placed in or at a first ear of a user, the second microphone system is adapted to be placed in or at a second ear of said user, the method includes a measurement step, where the spectral and temporal modulations of first and second input signals are monitored, an evaluation step, where the spectral and temporal modulations of the first and second input signal are evaluated by the calculation of an evaluation index of speech intelligibility for each of said signals, and an operational step, where the microphone mode of the first and the second microphone systems of the binaural hearing aid are selected in dependence of the calculated evaluation indexes.
Claims
1. A method of automatic switching between a first microphone mode and a second microphone mode in a binaural hearing aid, the binaural hearing aid comprising a first microphone system for provision of a first input signal, and a second microphone system for provision of a second input signal, where the first microphone system is configured for use by a first ear of a user, and the second microphone system is configured for use by a second ear of the user, the method comprising: obtaining modulations of the first input signal and the second input signal, wherein the act of obtaining the modulations of the first input signal and the second input signal is performed with at least the first microphone system being in the first microphone mode; evaluating the modulations of the first input signal and the second input signal; and setting one or both of the first microphone system and the second microphone system based on a result from the act of evaluating; wherein the act of setting comprises setting one of the first microphone system and the second microphone system to the second microphone mode, and setting the other one of the first microphone system and the second microphone system to the first microphone mode, the second microphone mode being different from the first microphone mode.
2. The method of claim 1, wherein the modulations comprise spectral modulations.
3. The method of claim 2, wherein the modulations also comprise temporal modulations.
4. The method of claim 1, wherein the modulations comprise temporal modulations.
5. The method of claim 1, wherein the first microphone mode comprises a directional microphone mode.
6. The method of claim 1, wherein the first microphone mode comprises an omni directional microphone mode.
7. The method of claim 1, wherein the act of evaluating the modulations comprises: determining a first evaluation index of speech intelligibility for the first input signal; and determining a second evaluation index of speech intelligibility for the second input signal.
8. The method of claim 7, further comprising processing the first evaluation index and the second evaluation index.
9. The method of claim 8, wherein the act of setting comprises setting one or both of the first microphone system and the second microphone system to the first microphone mode when a result of processing the first evaluation index and the second evaluation index indicates low speech intelligibility.
10. The method of claim 7, wherein the act of setting comprises: setting one of the first microphone system and the second microphone system with a corresponding one of the first evaluation index and the second evaluation index indicating highest speech intelligibility to an omnidirectional microphone mode, and setting the other one of the first microphone system and the second microphone system with a corresponding one of the first evaluation index and the second evaluation index indicating lowest speech intelligibility to a directional microphone mode.
11. The method of claim 7, wherein the first evaluation index of speech intelligibility is selected from the group consisting of: a speech transmission index (STI), a modified speech transmission index (mSTI), a spectral temporal modulation index (STMI), a modified temporal modulation index (mSTMI), an articulation index (AI), and a modified articulation index (mAI).
12. The method of claim 1, wherein the act of setting comprises setting the first microphone system to a directional microphone mode.
13. The method of claim 1, wherein the act of setting comprises setting the first microphone system to an omni directional microphone mode.
14. The method of claim 1, wherein the act of setting comprises setting one or both of the first microphone system and the second microphone system to a directional microphone mode when a result from the act of evaluating indicates low speech intelligibility.
15. The method of claim 1, wherein the act of obtaining the modulations of the first input signal and the second input signal is performed with the second microphone system being in an omni directional microphone mode.
16. A binaural hearing aid comprising: at least one signal processor; a first microphone system for provision of a first input signal; a second microphone system for provision of a second input signal, where the first microphone system is adapted to be placed in or at a first ear of a user, and the second microphone system is adapted to be placed in or at a second ear of the user; wherein the at least one signal processor is configured for: obtaining modulations of the first input signal and the second input signal with at least the first microphone system being in a first microphone mode; evaluating the modulations of the first input signal and the second input signal; and setting one or both of the first microphone system and the second microphone system based on a result from the act of evaluating; wherein the at least one signal processor is configured for setting one of the first microphone system and the second microphone system to a second microphone mode, and setting the other one of the first microphone system and the second microphone system to the first microphone mode, the second microphone mode being different from the first microphone mode.
17. The binaural hearing aid of claim 16, wherein the modulations comprise spectral modulations.
18. The binaural hearing aid of claim 17, wherein the modulations also comprise temporal modulations.
19. The binaural hearing aid of claim 16, wherein the modulations comprise temporal modulations.
20. The binaural hearing aid of claim 16, wherein the first microphone mode comprises a directional microphone mode.
21. The binaural hearing aid of claim 16, wherein the first microphone mode comprises an omni directional microphone mode.
22. The binaural hearing aid of claim 16, wherein the at least one signal processor is configured to evaluate the modulations by: determining a first evaluation index of speech intelligibility for the first input signal; and determining a second evaluation index of speech intelligibility for the second input signal.
23. The binaural hearing aid of claim 22, wherein the at least one signal processor is also configured to process the first evaluation index and the second evaluation index.
24. The binaural hearing aid of claim 23, wherein the at least one signal processor is configured for setting one or both of the first microphone system and the second microphone system to the first microphone mode when a result of processing the first evaluation index and the second evaluation index indicates low speech intelligibility.
25. The binaural hearing aid of claim 22, wherein the at least one signal processor is configured for: setting one of the first microphone system and the second microphone system with a corresponding one of the first evaluation index and the second evaluation index indicating highest speech intelligibility to an omnidirectional microphone mode, and setting the other one of the first microphone system and the second microphone system with a corresponding one of the first evaluation index and the second evaluation index indicating lowest speech intelligibility to a directional microphone mode.
26. The binaural hearing aid of claim 22, wherein the first evaluation index of speech intelligibility is selected from the group consisting of: a speech transmission index (STI), a modified speech transmission index (mSTI), a spectral temporal modulation index (STMI), a modified temporal modulation index (mSTMI), an articulation index (AI), and a modified articulation index (mAI).
27. The binaural hearing aid of claim 16, wherein the at least one signal processor is configured for setting the first microphone system to a directional microphone mode.
28. The binaural hearing aid of claim 16, wherein the at least one signal processor is configured for setting the first microphone system to an omni directional microphone mode.
29. The binaural hearing aid of claim 16, wherein the at least one signal processor is configured for setting one or both of the first microphone system and the second microphone system to a directional microphone mode when a result from the act of evaluating indicates low speech intelligibility.
30. The binaural hearing aid of claim 16, wherein the at least one signal processor is configured for obtaining the modulations of the first input signal and the second input signal with the second microphone system being in an omni directional microphone mode.
Description
BRIEF DESCRIPTION OF THE DRAWINGS
[0072] The drawings illustrate the design and utility of various features described herein, in which similar elements are referred to by common reference numerals. These drawings are not necessarily drawn to scale. In order to better appreciate how the above-recited and other advantages and objects are obtained, a more particular description will be rendered, which are illustrated in the accompanying drawings. These drawings depict only exemplary features and are not therefore to be considered limiting in the scope of the claims.
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DETAILED DESCRIPTION
[0081] Various features are described hereinafter with reference to the figures. It should be noted that the figures are not drawn to scale and that the elements of similar structures or functions are represented by like reference numerals throughout the figures. It should be noted that the figures are only intended to facilitate the description of the features. They are not intended as an exhaustive description of the claimed invention or as a limitation on the scope of the claimed invention. In addition, an illustrated feature needs not have all the aspects or advantages shown. An aspect or an advantage described in conjunction with a particular feature is not necessarily limited to that feature and can be practiced in any other features even if not so illustrated.
[0082] The embodiments will now be described more fully hereinafter with reference to the accompanying drawings, in which exemplary embodiments are shown. The claimed invention may, however, be embodied in different forms and should not be construed as limited to the embodiments set forth herein.
[0083] In the following description of the preferred embodiments primarily the use of a modified Speech Transmission Index (STI) as a fidelity measure in automatic switching between OMNI and DIR microphone modes is used, while it should be understood that other indexes that incorporate spectral and temporal modulations of the input signals, may be applied as well.
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[0085] Based on this and other preliminary work, the STMI appears to show promise as a means for deciding which microphone mode to select as the listening environment changes. However, since the STMI metric may, as stated before, be computationally too intensive or complicated for use in some ordinary hearing aid we will in the following focus on two applications of a modified STI to the problem of automatic switching between OMNI and DIR microphone modes in a binaural hearing aid involving asymmetric fittings. The modified STI used in the two following implementations of the inventive method may comprise an ordinary STI as known in the art, that is modified to include a speech template, codebook or table of certain components of a speech signal that are common in any given language. The modified STI may also comprise different numbers of coefficients and bin sizes than the standard.
[0086] In both implementations, the binaural hearing aid is set in the OMNI.sub.BI configuration only in quiet listening environments. When background noise is present, at least one of the microphone systems is set in the DIR mode, regardless of the location of the primary speech signal.
[0087] Before, the description of the preferred embodiment a more detailed description of the rationale of the STI metric will be explained: The metric needed to identify the key auditory scenes would naturally consist of temporal and spectral feature detectors and a clean speech template. Since, the microphone mode of a hearing aid alters two basic components that can affect speech reception for the hearing impaired, namely ambient (background) noise and reverberation (for more information see for example Ricketts T A, Dittbemer A B: Directional amplification for improved signal-to-noise ratio: Strategies, measurements, and limitations. In Valente M, ed. Hearing Aids: Standards, Options and Limitations, second ed. New York: Thieme Medical Publishers, 2002: 274-346), there is a need for an evaluation index that can classify an environment based on the relationship of speech to reverberation and noise. Such an index is for example the speech transmission index (STI) (e. g. Steeneken, H., & Houtgast, T. 1980. A physical method for measuring speech-transmission quality. Journal of the Acoustical Society of America, 67, 318-326. IEC 60268-16. (2003). Sound system equipment—Part 16: Objective rating of speech intelligibility by speech transmission index, 3rd ed).
[0088] The STI is not sensitive to cross-channel jitter and other nonlinearities (for more information see for example: Hohmann, V., & Kollmeier, B. (1995). The effect of multichannel dynamic compression on speech intelligibility. Journal of the Acoustical Society of America, 97, 1191-1195, which can be introduced by the loudness compensation strategy of the device, and obscure the acoustic environment and its classification. Hence, the STI provides the best means to make decisions what microphone mode is best for a given acoustic environment.
[0089] Speech is a complex signal. Its cues come both from its temporal envelope and spectral fine structure (i.e., low-frequency modulations and high-frequency content). The computation of the STI may be based upon the modulation transfer function (MTF) at temporal (low) and spectral (high) frequency regions, which is derived from objective estimates of the signal-to-noise ratio (SNR).
[0090] The fundamental component of the STI is the modulation index, m, which is a function of both the modulation frequency, mf, and third-octave center frequency, cf. For example we may choose 14 modulation frequencies 0.63, 0.8, 1.0, 1.25, 1.6, 2.0, 2.5, 3.15, 4.0, 5.0, 6.3, 8, 10 and 12.5, with 7 center frequencies at 125, 250, 500, 1000, 2000, 4000 and 8000 Hz. These values may vary dependent upon the fidelity of the device; the width of the filters may also be dependent on device fidelity, the nature of the hearing impairment and the general acoustic attributes of speech.
[0091] The modulation index may then simply be calculated as the ratio of the intensity of the signal to the intensity of the signal and noise; that is:
m.sub.cf,mf=I.sub.signal(cf,mf)I[I.sub.signal(cf,mf)+I.sub.noise(cf,mf)] (1)
[0092] There is a correction to this ratio to account for the upward spread of masking, which again may be corrected by an intensity-dependent auditory masking coefficient (amf): see for example
m′.sub.cf,mf=m.sub.cf,mf.Math.I.sub.cfI[I.sub.cf+(amf.Math.I.sub.cf-1)+(I.sub.noise∀I.sub.noise>I.sub.ART)] (2)
[0093] The contribution of masking and noise in equation (2) above may be modified from the standard to account for changes in masking susceptibility in the peripherally impaired auditory system (Glasberg, B., & Moore, B. (1989). Psychoacoustic abilities of subjects with unilateral and bilateral cochlear hearing impairments and their relationship to the ability to understand speech. Scandinavian Audiology, Supplement, 32, 1-25).
[0094] From the corrected modulation index at each cf and mf, m′.sub.cf,mf, the effective signal-to-noise ratio (SNR.sub.cf,mf) may be computed according to the equation:
SNR.sub.cf,mf=10.Math.log.sub.10[m′.sub.cf,mf/(1−m′.sub.cf,mf)] (3)
[0095] Based on the articulation index formulation of French and Steinberg (reported in: French, N., & Steinberg, J. (1947). Factors governing the intelligibility of speech sounds,” Journal of the Acoustical Society of America, 19, 90-119), the range of SNR values useful for speech transmission is substantially in the range of −15 to +15 dB. Thus, a normalized transmission index (TI.sub.cf,mf) may then be calculated according to the equation:
TI.sub.cf,mf=(SNR.sub.cf,mf+15 dB)/30 dB (4)
[0096] The modulation transfer index may then be calculated as the average of TIs across the modulation frequencies according to the equation:
[0097] The STI is taken from the sum of TIs averaged across modulation frequencies with corrections for octave weighting (α) and redundancy (β; see for example
[0098] See for example
[0099] In order to compute STI based on one of the two input signals, some estimate of a clean signal—“clean speech”—must be made, Instead of attempting to parse the input, one way of providing an estimate of a clean signal is to use a clean-speech template so that the STI of the acoustic environment—the denominator in equation (1)—can be properly estimated.
[0100] Corpuses of utterances by different genders (i.e., male and female), ages (i.e., child and adult), efforts (i.e., soft and loud) and languages are distilled into separate long-term intensity measurements (I.sub.signal) at the same cf and mf values given above. These corpuses may be parsed by language, and may be averaged across gender and age. Because of the disparate difficulty in the classification of female and child speech (see for example Klatt & Klatt, 1990), a disproportionate amount of female and child speech samples may be used to derive each language's clean-speech template. Each clean-speech template may, in a sense, be a set of 98 coefficients (for example arranged as a 14×7 matrix) that is loaded into a soft-switching algorithm—more specifically, the modified STI or Evaluation Index (EI)—when the device is fitted (i.e., when the optimal language is determined).
[0101] In
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[0103] As stated earlier, scientific investigations show that, when background noise is present and the speech is either in front of or behind the listener, it should make little difference which ear receives the OMNI processing and which ear receives the DIR processing. However, when the speech signal is to one side, head shadow effects come into play and the scientific investigations show that a user would prefer that the ear closest to the speech signal should receive the OMNI processing. The STI enables us to determine the preferred ear to receive OMNI processing by comparing the results across ears for the OMNI mode. If the difference between the STI.sub.OMNI for each ear is small, one can assume that the speech signal is coming from in front of or behind the listener. On the other hand, if the difference between STI.sub.OMNI across the ears is large, one can assume that the ear with the greater STI is closest to the speech signal and it should benefit from OMNI processing. Thus, the flow of the algorithm as showed in
[0104] If on the other hand the STI.sub.OMNI difference across ears exceeds 0.1, the ear with greater STI receives OMNI processing and the contralateral ear receives DIR processing. This means that the expression D>0.1 is true, as indicated by the output T of block 18, where after the STI for both input signals, and thereby for both ears is compared in block 20, and the microphone system that generates the input signal with highest STI is set to an OMNI mode, while the other microphone system is set to operate in a DIR mode. This selection of the asymmetrical fit is indicated by block 22 in
[0105] The Implementation of an algorithm according to the inventive method as indicated in
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[0107] As before the default mode for the binaural hearing aid is OMNI.sub.BI, and the default mode for the asymmetric fit is specified as either OMNI.sub.RT/DIR.sub.LT or DIR.sub.RT/OMNI.sub.LT, possibly depending upon patient preferences/needs. In the following description of the embodiment shown in
[0108] The first steps in the algorithm shown in
[0109] If it is true (indicated by the output T of block 26) that STI.sub.LT is substantially equal to the STI.sub.RT then in the processing block 28, it is evaluated whether the expression STI.sub.DIR−STI.sub.OMNI>0 is true. If STI.sub.DIR−STI.sub.OMNI is a positive number, then this is indicative of that the desired speech signal is in front of the user, and the operating state of the binaural hearing aid is chosen to be DIR.sub.BI, i.e. both of the microphone systems is chosen to operate in a DIR mode. This is indicated by the block 30. However, if the expression STI.sub.DIR−STI.sub.OMNI>0 is false, indicated by the output F of block 28, this is indicative of the fact that the desired signal location is behind the user of the binaural hearing aid according to some embodiments, and then a default asymmetric microphone configuration is chosen. If the STI.sub.DIR−STI.sub.OMNI is negative and unequal at the two ears, this would have been reflected in a difference in the STI.sub.OMNI between the two ears and the binaural hearing aid would have already selected an asymmetric fit.
[0110] Note that the decision to select the DIR.sub.BI configuration is conservative in that four conditions must be met. First, the STI.sub.OMNI score in both ears must be below 0.6 (noise present). Second, there must be a STI.sub.OMNI difference between ears of less than 0.1 (symmetrical signal input). Third, the STI.sub.DIR−STI.sub.OMNI must be positive in both ears (desired signal in front of the user). Fourth, the magnitude of the STI must be equal at the two ears (symmetrical DIR benefit). As noted above, when the condition of block 28 is not met, i.e. the expression STI.sub.DIR−STI.sub.OMNI>0 is false, it is assumed that the desired signal source is located behind the listener. In this case, DIR processing is not likely to be beneficial in either ear and, it could be argued that an OMNI.sub.BI configuration might be optimal. Nevertheless, as currently envisioned, the inventive binaural hearing aid is configured in the fixed asymmetric setting. The rationale here is that, with noise present, the potential for directional benefit exists if the listener should turn to face the signal of interest. In this case, the inventive binaural hearing aid would already be configured for DIR processing in one ear, thus avoiding the processing delay that would be required to reconfigure the system from OMNI.sub.BI to a directional mode.
[0111] The scientific investigations have involved laboratory testing of speech recognition for four hearing aid fitting strategies (OMNI.sub.BI, DIR.sub.BI, OMNI.sub.RT/DIR.sub.LT, and DIR.sub.RT/OMNI.sub.LT) for speech stimuli presented from four source locations surrounding a listener. In addition, STI analyses have been carried out to determine whether STI scores accurately predict the performance differences observed in the behavioral data, across processing modes and source locations.
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[0113] The first housing structure 24 comprises a first microphone system 38 for the provision of a first input signal, an A/D converter 40 for converting the first input signal into a first digital input signal, a digital signal processor (DSP) 42 that is adapted to process the digitalized first input signal, a D/A converter 44 for converting the processed first digital input signal into a first analogue output signal. The first analogue output signal is then transformed into a first acoustical output signal (to be presented to a first ear of a user) in a first receiver 46.
[0114] Similarly the second housing structure 36 comprises a second microphone system 48 for the provision of a second input signal, an A/D converter 50 for converting the second input signal into a second digital input signal, a digital signal processor (DSP) 52 that is adapted to process the digitalized second input signal, a D/A converter 54 for converting the processed second digital input signal into a second analogue output signal. The second analogue output signal is then transformed into a second acoustical output signal (to be presented to a second ear of a user) in a second receiver 56. In a preferred embodiment, the first and second housing structures are individual hearing aids, possibly known in the art.
[0115] The binaural hearing aid 32 furthermore comprises a link 58, between the two housing structures 34 and 36. The link 58 is preferable wireless, but may in another embodiment be wired. The link 58 enables the two housing structures to communicate with each other, i.e. it may be possible to send information between the two housing structures via the link 58. The link 58, thus, enables the two digital signal processors, 42 and 52, to perform binaural signal processing according to the inventive method described above, wherein information derived from both microphone systems, 38, 48, is used in the signal processing in order to determine the operating state (OMNI or DIR) of each of the microphone systems 38, 48, that provides the user with optimal speech intelligibility in compliance with user preferences.
[0116] As illustrated above, the use of spectral and temporal modulations of the input signals of a binaural hearing aid is feasible and may be used to predict beneficial microphone configurations in compliance with user preferences. However, as will be understood by those familiar in the art, the present embodiments may be embodied in other specific forms and utilize any of a variety of different algorithms without departing from the spirit or essential characteristics thereof. For example the selection of an algorithm may typically application and/or user specific, the selection depending upon a variety of factors including the size and type of the hearing loss of the user, the expected processing complexity and computational load. Accordingly, the disclosures and descriptions herein are intended to be illustrative, but not limiting, of the scope of the invention which is set forth in the following claims.
[0117] Although particular features have been shown and described, it will be understood that they are not intended to limit the claimed invention, and it will be made obvious to those skilled in the art that various changes and modifications may be made without departing from the spirit and scope of the claimed invention. The specification and drawings are, accordingly to be regarded in an illustrative rather than restrictive sense. The claimed invention is intended to cover all alternatives, modifications and equivalents.