Procedure for operating a hearing instrument
By selectively applying signal processing algorithms in frequency bands that contribute to the output sound, the method optimizes hearing instrument performance, reducing artifacts and enhancing sound quality and spatial hearing.
Patent Information
- Application Number
- DE102023200581
- Authority / Receiving Office
- DE · DE
- Patent Type
- Patents
- Current Assignee / Owner
- Filing Date
- 2023-01-25
- Publication Date
- 2024-12-12
- Estimated Expiration
- 2043-01-25
AI Technical Summary
Existing hearing instruments often apply signal processing algorithms indiscriminately, leading to impaired spatial hearing and artifacts due to unnecessary amplification or suppression of sound components that do not contribute to the output sound, particularly in complex listening situations.
The method involves breaking down input signals into frequency bands and applying sub-algorithms like noise suppression or directional microphones only in frequency bands where they contribute significantly to the output sound, determined by a relevant subset exceeding a predetermined threshold, thereby optimizing signal processing based on audiological needs.
This approach enhances the hearing experience by minimizing artifacts and improving sound quality by ensuring that signal processing algorithms are only activated where necessary, thus maintaining a realistic hearing impression and spatial awareness.
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Abstract
Description
[0001] The invention relates to a method for operating a hearing instrument which has at least one acousto-electrical first input transducer and one electro-acoustic output transducer, wherein a first input signal is generated from an ambient sound by the first input transducer, wherein the first input signal and / or a first intermediate signal derived from the first input signal is broken down into a plurality of frequency bands, and wherein an output signal is generated from the first input signal or the first intermediate signal by means of frequency-selective signal processing.
[0002] A hearing instrument is generally defined as an electronic device that supports the hearing of a person wearing the hearing instrument (hereinafter referred to as the "wearer" or "user"). In particular, the invention relates to hearing instruments designed to fully or partially compensate for the hearing loss of a hearing-impaired user. Such a hearing instrument is also referred to as a "hearing aid." There are also hearing instruments designed to protect or improve the hearing of users with normal hearing, for example, to enable improved speech comprehension in complex listening situations, or in the form of communication devices (e.g., headsets or similar, possibly with earplug-like headphones).
[0003] Hearing instruments in general, and hearing aids in particular, are usually designed to be worn on the head or in one ear of the user, particularly as behind-the-ear devices (also known as BTE devices) or in-the-ear devices (also known as ITE devices).
[0004] With regard to their internal structure, hearing instruments usually have at least one (acousto-electrical) input transducer, a signal processing device (signal processor) and an output transducer. When the hearing instrument is in operation, the or each input transducer receives ambient sound and converts this ambient sound into a corresponding electrical input signal. In the signal processing device, the or each input signal is processed (i.e. modified with regard to its sound information) in order to particularly support the user's hearing ability, thus particularly preferably to compensate for any hearing loss the user may have. The signal processing device outputs an appropriately processed audio signal as an output signal to the output transducer, which converts the output signal into an output sound signal. The output sound signal can consist of airborne sound that is emitted into the user's ear canal (if necessary).via a sound tube, as with a BTE device, or by appropriately positioning the hearing instrument in the ear canal. The output sound signal can also be delivered into the user's skull bone.
[0005] Many sub-algorithms within the scope of the said signal processing, for example noise suppression or directional microphones (the latter in conjunction with a second input signal of the hearing instrument) are applied to the first input signal depending on an activation criterion: If the activation criterion is met, which in turn includes the verification of certain characteristics of signal components of the first input signal, the respective sub-algorithm is applied accordingly.
[0006] In this context, the attempt is often made to use signal processing as conservatively as possible, within the context of the user's audiological needs. This is particularly important given that signal processing often leads to a compromised realistic auditory impression, for example, in spatial hearing, or due to artifacts that can arise from signal processing.
[0007] DE 10 2016 200 637 B3 describes a method for reducing the latency of a filter bank for filtering an audio signal, wherein a plurality of signal blocks are formed in the time domain from the audio signal, wherein a filter function is specified for at least a plurality of the signal blocks, at least one subinterval of the signal block is specified as a prediction period, signal components of the signal block in at least one subinterval are estimated for the prediction period, and a predicted signal block is generated from the signal components estimated for the prediction period and the signal components of the signal block outside the prediction period. The predicted signal block is filtered with the specified filter function and transformed into the frequency domain, thereby forming a transformed signal block, so that signal components of the transformed signal block can be output for further processing.
[0008] DE 600 37 034 T2 discloses a device for selectively processing an audio signal to thereby generate an output signal. Said device comprises a plurality of audio bandpass filters, which serve to separate the audio signal into respective frequency bands with a resolution corresponding at least to the critical bandwidth. The device further comprises a plurality of circuits, each multiplicative automatic gain control circuit connected to one of the audio bandpass filters to process the bandpass signal from the associated audio bandpass filter to apply an AGC gain and generate an AGC output. At least one such circuit comprises a first low-pass filter, a logarithmic element, an exponential element, and a first summing element for combining the AGC outputs to generate the output signal.The said circuit comprises a second summing element having a first input connected to an output of the first low-pass filter, which also has a second input equal to log[e-max], where e-max is the maximum permissible value of the audio envelope to which the AGC gain is applied.
[0009] It is therefore an object of the invention to improve the control of the application of signal processing to an input signal of a hearing instrument.
[0010] The stated object is achieved according to the invention by a method for operating a hearing instrument which has at least one acousto-electrical first input transducer and one electro-acoustic output transducer, wherein a first input signal is generated from an ambient sound by the first input transducer, wherein the first input signal and / or a first intermediate signal derived from the first input signal is broken down into a plurality of frequency bands, and wherein an output signal is generated from the first input signal or the first intermediate signal by means of frequency-selective signal processing.
[0011] According to the method, a relevant subset of frequency bands is determined from said plurality in such a way that, in each frequency band of the relevant subset, an output sound generated from the output signal by the output transducer provides a contribution that lies above a predetermined and / or desired threshold; that, further, based on signal components of the first input signal or the first intermediate signal, an activation criterion for activating a sub-algorithm for said signal processing is checked only in the frequency bands of the relevant subset; and that said sub-algorithm is applied to the first input signal or the first intermediate signal depending on the activation criterion. Advantageous and, in part, inventive embodiments are the subject of the dependent claims and the following description.
[0012] As described above, the hearing instrument can be designed to support the hearing ability of a user and, in particular, can be designed as a hearing aid “in the narrower sense” (i.e., to treat a hearing impairment).
[0013] An acoustoelectrical input transducer includes, in particular, any device configured to generate a corresponding electrical signal from a sound signal. In particular, preprocessing can also occur during the generation of the first or second input signal by the respective input transducer, e.g., in the form of linear preamplification and / or A / D conversion. During operation of the hearing instrument, the or each input transducer records ambient sound and converts this ambient sound into a corresponding electrical input signal, the current and / or voltage fluctuations of which preferably carry information about the air pressure oscillations in the air caused by the ambient sound.
[0014] An electro-acoustic output transducer is any device intended and configured to convert an electrical signal into a corresponding sound signal, whereby voltage and / or current fluctuations in the electrical signal are converted into corresponding amplitude fluctuations of the sound signal, in particular a loudspeaker, a so-called balanced metal case receiver, but also a bone conduction receiver
[0015] The term “first intermediate signal” derived from the first input signal preferably includes the fact that the signal components of the first input signal are directly incorporated into the first intermediate signal, and thus, in particular, the first input signal is not merely used for generating control parameters or the like which are applied to signal components of other signals.
[0016] The first input signal (or said first intermediate signal) is then broken down into a plurality of frequency bands, preferably using a corresponding analysis filter bank, in order to process the signal components of the first input signal (or the first intermediate signal) in a frequency-band-specific manner, preferably depending on the user's audiological requirements. This frequency-selective processing of the signal components of the first input signal or intermediate signal generates an output signal, which is converted into an output sound by the output transducer, with the voltage fluctuations of the output signal preferably being converted into corresponding air pressure oscillations in the output sound.
[0017] A relevant subset is now determined from the plurality of frequency bands generated for frequency-selective signal processing. This relevant subset of frequency bands is characterized by the fact that the contributions present in these frequency bands in the output sound generated by the output transducer from the output signal are above a desired or specified threshold (e.g., a minimum level in dB or similar).In other words, those frequency bands are selected as the relevant subset in which the frequency-selective signal processing actually leads to relevant contributions in the output sound, since depending on the adjustment of the hearing instrument, or also on the respective algorithm in a given listening situation, certain frequency bands, in particular at one of the edges (or at both edges) of the transmitted frequency spectrum, do not experience any significant (i.e. in particular no perceptible) amplification.
[0018] The relevant subset can be determined, in particular, statically, depending on knowledge of the signal amplification in the individual frequency bands. Preferably, an adaptation formula for frequency-band-by-frequency adaptation of the hearing instrument is used to determine the relevant subset. In particular, in an "open adaptation" of a hearing instrument with a large ventilation channel ("vent"), which may be provided in the housing of the hearing instrument to prevent occlusion, and which connects the area of the ear canal closed by the hearing instrument with the free outer area, the signal amplification of low frequency bands from 0 Hz to 500 Hz, preferably up to 1000 Hz, particularly preferably up to 1500 Hz, can be suspended or largely suspended (i.e., in principle, preferably at least 10 dB, particularly preferably at least 20 dB lower than the most strongly amplified bands).
[0019] The relevant subset of frequency bands is then used to check an activation criterion for activating a subalgorithm for said signal processing only or at most in the frequency bands of the relevant subset, and in particular, to not perform such a check in those frequency bands that do not belong to the relevant subset. The subalgorithm is then applied to the first input signal or the first intermediate signal derived from it, depending on the activation criterion.
[0020] This can prevent a sub-algorithm of the signal processing, which preferably comprises frequency band-wise amplification, and / or frequency band-wise compression, and / or noise suppression, from being dynamically activated on the basis of sound events which ultimately do not contribute to the output sound generated by the hearing instrument.
[0021] If, for example, there is significant noise outside the relevant frequency bands (i.e., the frequency bands of the relevant subset) (such as a low-frequency hum) that is not transmitted at all (or not to an audible extent) by the hearing instrument, a noise suppression algorithm will not be activated by the described method either, since this algorithm would not be able to correct the hum, or would not be able to do so satisfactorily, due to the lack of signal amplification in the hum frequency range. However, this would potentially be accompanied by other problems (such as artifacts), which are now avoided. Only when such noise lies within the frequency bands of the relevant subset (and is therefore sufficiently transmitted by the hearing instrument, thus becoming significantly correctable), is a partial algorithm for appropriate noise suppression preferably activated.
[0022] Preferably, to determine the relevant subset, a gain value of signal contributions of the first input signal or the first intermediate signal in the respective frequency band can be determined frequency band by frequency band. The relevant subset is then preferably formed by those frequency bands for which the gain value exceeds a predetermined limit value, which is preferably selected depending on the said threshold for the contribution in the output sound.
[0023] To determine the gain value, the first input signal is advantageously compared with the output signal for each frequency band, and / or a signal gain applied along a signal path from the first input transducer to the output transducer is tracked, whereby the gain value is compared with a first limit value dependent on the said threshold. In other words, the signal gain that "accumulates" for each frequency band along the entire signal path from the first input transducer to the output transducer is tracked, and this cumulative signal gain of all sub-algorithms of the signal processing in the frequency band is compared with the first limit value, which represents the relevant contribution to the output sound at the signal gain level.
[0024] For determining the relevant subset, a signal gain setting made by a user is expediently taken into account. This can be done, in particular, by instantly correcting a signal gain accumulating along the entire signal path by the value of a setting made by the user.
[0025] In an advantageous embodiment, to check the activation criterion, a parameter is determined from the signal components of the first input signal or the first intermediate signal in the frequency bands of the relevant subset, which parameter provides information about a noise component in said frequency bands. In other words, this means that an estimate of the noise component in the "relevant" frequency bands of the relevant subset is made using the parameter. Advantageously, said parameter is compared with a noise threshold value, which corresponds to an upper limit for a permissible noise component in said frequency bands. The activation criterion for activating the subalgorithm is considered to be met if the noise threshold value is exceeded.
[0026] In particular, if a high noise component is detected in the relevant frequency bands, a sub-algorithm for noise suppression can be activated based on the characteristic. The characteristic used is preferably a signal-to-noise ratio (SNR) of the signal components in the frequency bands of the relevant subset. This can be achieved by estimating a noise component and, if applicable, a useful signal component, which can be determined, for example, from the medium-term (e.g., over several frames or several tens of frames) statistical behavior of the respective signal components.
[0027] In a further advantageous embodiment, a second input signal is generated from the ambient sound by an acoustoelectrical second input transducer of the hearing instrument, wherein the output signal is additionally generated from frequency-selective signal processing of the second input signal, and wherein the sub-algorithm comprises directional microphony of the first input signal or the first intermediate signal and the second input signal and / or a second intermediate signal derived from the second input signal. In particular, this includes using the activation criterion to activate directional noise suppression. Many hearing instruments, in particular hearing aids for treating hearing loss, often have more than one input transducer to enable directional signal processing.In the advantageous embodiment of the invention mentioned, this directional signal processing is activated in the manner described depending on the activation criterion in the relevant frequency bands.
[0028] Preferably, the activation criterion is also checked based on signal components of the second input signal or the second intermediate signal only in the frequency bands of the relevant subset. This includes, in particular, forming a preliminary directional signal from the first and second input signals, the signal components of which can be checked for compliance with the activation criterion in the relevant frequency bands. If the activation criterion is met (i.e., a decision is made to activate the subalgorithm), the directional signal can be directly processed into the output signal in the case of directional microphones as the subalgorithm to be activated.
[0029] The invention further mentions a hearing instrument comprising at least one acousto-electrical first input transducer, an electro-acoustic output transducer, and a signal processing device, wherein the hearing instrument is configured to carry out the method according to one of the preceding claims.
[0030] The hearing instrument according to the invention shares the advantages of the method according to the invention. The advantages stated for the method and its further developments can be applied analogously to the hearing instrument.
[0031] An embodiment of the invention is explained in more detail below with reference to the accompanying drawings, each of which shows schematically: Fig. 1 in a block diagram a hearing instrument with frequency band-wise signal processing, Fig. 2 shows a block diagram of a frequency-selective activation of a sub-algorithm of the signal processing according to Fig. 1.
[0032] Corresponding parts and sizes are provided with the same reference numerals in all figures.
[0033] In Fig. Figure 1 schematically shows a block diagram of a hearing instrument 1, which has an acousto-electrical first input transducer 2 and an acousto-electrical second input transducer 4, as well as an electro-acoustic output transducer 6 and a signal processing device 8. The first input transducer 2 and the second input transducer 4 are designed as a first and second microphone 3, 5, respectively. The output transducer 6 is designed as a loudspeaker 7. The signal processing device 8 comprises at least one signal processor 9 (shown in dashed lines). The hearing instrument 1 is designed as a hearing aid 10, which is configured to treat, i.e., at least partially compensate for or correct, a hearing impairment of a user (not shown in detail).
[0034] For this purpose, a first input signal 14 is generated from ambient sound 12 by the first microphone 3, and a second input signal 16 is generated by the second microphone 5. The first input signal 14 and the second input signal 16 are further processed together in the signal processing unit 8 to form an output signal 18, and in particular are amplified in a frequency band-specific manner. The output signal 18 is converted by the loudspeaker 7 into an output sound 20, which is emitted or guided into an ear canal (not shown) of the user. A ventilation channel 13 (so-called "vent"; shown in dashed lines) is also incorporated in a housing 11 of the hearing aid 10. This is intended to ensure better pressure equalization given that the ear canal is largely closed off by the housing 11.
[0035] The signal processing of the first and second input signals 14, 16 to produce the output signal 18 in the signal processing device 8 is, on the one hand, as already mentioned, carried out depending on the user's audiological requirements, so that, for example, important frequency bands in which the user's hearing loss is particularly pronounced are generally amplified more than those frequency ranges in which the hearing loss is only slight. On the other hand, specific sub-algorithms of the signal processing, such as noise suppression or directional microphony, i.e., the formation of a directional signal from the first and second input signals 14, 16, are applied in a manner to be described below depending on specific acoustic characteristics in the ambient sound 12.This means in particular that a particular sub-algorithm is only applied if the features considered necessary for the application are present in the ambient sound 12 to a sufficient extent.
[0036] This relationship between the characteristics of the ambient sound 12 and the application of a partial signal processing algorithm in the signal processing device 8 will now be explained by means of Fig. 2. In Fig. 2 schematically shows a block diagram of a method by means of which an application of a partial algorithm in the signal processing of the hearing aid 10 according to Fig. 1. It should be noted that the hearing aid 10 is a so-called ITE ("In The Ear") device. However, it could just as easily be designed as a BTE ("Behind The Ear") device, a CIC ("Completely In the Canal") device, or an RIC ("Receiver In the Canal") device.
[0037] The first input signal 14 generated by the first microphone 3 is broken down in the signal processing device 8 by a first analysis filter bank 22 into a plurality 23 of frequency bands 24a-z. Corresponding signal components 26a-z of the first input signal 14 are then subjected to an analysis 27 in the signal processing device 8 and, depending on the analysis 27, to the respectively provided signal processing 28. The signal components 26a-z are amplified, among other things, in a frequency-dependent manner by applying corresponding gain factors Ga-Gz, and are also compressed in a frequency-band-dependent manner (i.e., the gain factors Ga-Gz are adjusted almost instantaneously depending on the dynamics of the signal components 26a-z). The processed signal components 29a-z resulting from the signal processing 28 are combined by a first synthesis filter bank 30 to form the output signal 18.
[0038] In particular, a sub-algorithm 32 is used in the signal processing 28, which can be provided, for example, by the already described, frequency-band-dependent amplification using the amplification factors Ga-Gz, a noise suppression, or even directional microphony with signal components of the second input signal 16, whereby this directional microphony can also be used as directional noise suppression. However, depending on the acoustic situation contained in the first input signal 14, the sub-algorithm 32 should only be used in those situations, in particular listening situations, in which an improvement in the hearing ability or perception for the user of the hearing aid 10 is to be expected through its application to the corresponding signal components 26a-z.
[0039] This means in particular that, for example, noise suppression is not applied permanently, since noise suppression algorithms can, for example, produce undesirable artifacts in the output signal, but only when this appears to be appropriate in view of the acoustic information contained in the first input signal 14 about the ambient situation (i.e., a noisy environment rich in interference is assumed or detected). Likewise, directional processing of the first and second input signals 14, 16 to form a directional signal or an amplification of the directional effect of such a directional signal is only applied when this appears to be appropriate in view of the acoustic analysis of the first (and possibly second) input signal 14 (or 16), since directional microphony is fundamentally capable of disrupting the spatial hearing perception, so that, for example, sound sources may no longer be correctly localized.
[0040] For this reason, an activation criterion 34 is checked for an application of the sub-algorithm 32 of the signal processing 28 in a manner to be described below, which activation criterion is intended to ensure that in the respective existing environmental situation with its acoustic conditions, the application of the sub-algorithm 32 for the user, also and in particular in view of his individual audiological requirements, the advantages of the application outweigh possible disadvantages (such as those mentioned above).
[0041] However, in certain of the frequency bands 24a-z, the signal processing is such that these do not make any significant contributions to the output signal 18. This may be due, for example, to the fact that for a particularly large ventilation duct 13, a high proportion of direct sound in lower frequency bands passes through said ventilation duct 13 into the ear canal (and thus to the eardrum), and thus said direct sound overlaps with amplified signal components in the lower frequency bands, potentially leading to undesirable comb filter effects. Furthermore, above 8 kHz or 10 kHz, there is often no significant amplification of the relevant signal contributions, since the relevant frequency bands are generally no longer relevant for speech intelligibility.
[0042] Thus, the activation criterion 34 could possibly evaluate signal components 26a-z of the first input signal 14 whose equivalents in the output sound 20 do not make a significant contribution (i.e., in particular, one that is not easily perceptible to the user) to an overall sound (not shown) that reaches the eardrum (not shown). In particular, the overall sound includes, in addition to the output sound 20, a portion of direct sound that enters the ear canal through the ventilation channel 13. This could potentially result in a deterioration in the sound quality or the spatial hearing perception as a result of the application of the sub-algorithm 32 to the signal components 26a-z of the first input signal 14, even though the sub-algorithm 32 is, in certain cases, only applied to those signal components 26a-z whose contributions are not audible at all in the output sound 20.
[0043] To prevent this, a relevant subset 25 of frequency bands 24b-x is determined from the frequency bands 24a-z of said plurality 23, which contribute to the output sound 20 to a relevant extent. This can be done, for example, statically using an adaptation formula of the hearing aid 10, which provides information about which target amplification values are preferentially achievable for certain frequency band-wise input levels, and which thus also provides information about those of the frequency bands 24a-z that, as a result of the adaptation, will fundamentally not make a significant contribution to the output sound 20.
[0044] In particular, the adaptation formula can also be used to specify frequency band-by-frequency base gain values for respective level values in the relevant frequency band, so that the said relevant subset 25 (of the "relevant frequency bands" for the output sound 20) can be determined with knowledge of such base gain values based on the signal components 26a-z (e.g., from their respective signal levels) in the individual frequency bands 24a-z. Furthermore, a user input (not shown) can also change the gain factors Ga-Gz frequency-selectively or broadband, so that in a specific situation (i.e., for a given set of signal components 26a-z), a change in the signal levels and thus in the respective contributions to the output sound 20 is accompanied by this user input (relative to the state prior to the user input).
[0045] An efficient way to take all of this into account and to determine the relevant subset 25 of the "relevant frequency bands" is to track the entire signal amplification along a signal path from the first microphone 3 (if applicable, including its input characteristic curve and pre-amplification) to the loudspeaker 7 (if applicable, including its output characteristic curve) for each frequency band 24a-z, and thereby determine a gain value Ga`-Gz` in each frequency band, which thus represents the entire signal amplification along the described signal path. The gain value Ga'-Gz comprises the respective gain factor Ga-Gz from the sub-algorithm 32, and if applicable, further gain factors from other sub-algorithms (not shown) of the signal processing 28 (as well as, if applicable, the aforementioned characteristic curves). To form the gain values Ga'-Gz', the (instantaneous) gain factors Ga-Gz (and, if applicable,additional gain factors from other sub-algorithms) in order to avoid a dependence of the relevant frequency bands (i.e. the relevant subset 25) on level peaks.
[0046] The gain values Ga'-Gz', which are influenced to the extent described by the adaptation formula and, if applicable, a user input, and further by the signal components 26a-z present at the relevant moment, are now compared with a first limit value 36. If the first limit value 36 is exceeded by the respective gain value Ga`-Gz`, the associated frequency band 24a-z is assigned to the relevant subset 25; otherwise, it is not. The first limit value 36 is preferably selected such that signal amplification with the corresponding gain value leads to a contribution in the output sound that lies above a desired threshold, preferably dependent on the ambient sound and / or the direct sound arriving at the eardrum.
[0047] In the present case, the frequency bands 24c-24x are determined as the relevant subset 25, i.e., the frequency bands 24a-b and 24y-z do not make any relevant contributions to the output sound 20 (in relation to the ambient sound or the direct sound at the eardrum).
[0048] Across frequency bands 24c-24x of the relevant set 25, a characteristic 38 is now determined from the respective signal components 26c-26x for the activation criterion 34. This characteristic provides information about a noise component (particularly in said frequency bands 24c-24x or in their entirety). Characteristic 38 is given by the broadband SNR 40 in said frequency bands 24c-24x.
[0049] The SNR 40 is then compared to a noise threshold 42, and if the SNR 40 exceeds said noise threshold 42, it is concluded that the noise component in the relevant frequency bands 24c-24x is so high that the advantages of the sub-algorithm 32 in terms of improving the SNR 40 now outweigh its disadvantages in sound quality (e.g., with regard to artifacts), and activation of the sub-algorithm 32 is therefore justified. The sub-algorithm 32 is therefore applied to the signal components 26a-26z (thus, at least potentially, also to the signal components 26a-b, 26y-z of the frequency bands 24a-b, 24y-z, which are not part of the relevant subset 25).
[0050] The processed signal components 29a-29z, which result from the signal processing 28 which, as a result of the described activation, also includes the sub-algorithm 32, are now combined at the synthesis filter bank 30 to form the output signal 18.
[0051] In particular, the sub-algorithm 32, which is activated by the described activation 34, can also use the second input signal 16, and thus, for example, can be configured as a directional microphone. In particular, the activation criterion 34 can also utilize signal components of the second input signal 16 (not shown in each case). List of reference symbols 1 hearing instrument 2 first input converter 3 first microphone 4 second input converter 5 second microphone 6 output converters 7 speakers 8 Signal processing device 9 Signal processor 10 hearing aids 11 housings 12 Ambient noise 13 Ventilation duct 14 first input signal 16 second input signal 18 Output signal 20 output sound 22 first analysis filter bank 23 majority (of frequency bands) 24a-z frequency bands 25 relevant subset (of the majority of frequency bands) 26a-z signal components 27 Analysis 28 Signal processing 29a-z processed signal components 30 first synthesis filter bank 32 sub-algorithms 34 Activation criterion 36 first limit 38 Parameter 40 SNR 42 Noise limit Ga-Gz gain factor Ga`-Gz` gain value
Claims
[1] Method for operating a hearing instrument (1) which has at least one acousto-electrical first input transducer (2) and one electro-acoustic output transducer (6), - wherein a first input signal (14) is generated from an ambient sound (12) by the first input transducer (1), - wherein the first input signal (14) and / or a first intermediate signal derived from the first input signal (14) is decomposed into a plurality of frequency bands (24a-z), - wherein an output signal (18) is generated from the first input signal (14) or the first intermediate signal by means of frequency-selective signal processing (28), - wherein a relevant subset (25) of frequency bands (24a-z) from said plurality (23) is determined such that in each frequency band (24c-x) of the relevant subset (25) an output sound (20) generated from the output signal (18) by the output transducer (6) provides a contribution which lies above a predetermined or desired threshold, - wherein an activation criterion (34) for activating a sub-algorithm (32) for said signal processing (28) is checked on the basis of signal components (26c-x) of the first input signal (14) or of the first intermediate signal only in the frequency bands (24c-x) of the relevant subset (25), and - wherein said sub-algorithm (32) is applied to the first input signal (14) or the first intermediate signal as a function of the activation criterion (34). [2] Method according to claim 1, wherein an adaptation formula of a frequency band-wise adaptation of the hearing instrument (1) is used to determine the relevant subset (25). [3] Method according to claim 1 or claim 2, wherein for determining the relevant subset (25) frequency band by frequency, a gain value (Ga`-Gz`) of signal contributions (26a-z) of the first input signal (14) or of the first intermediate signal in the respective frequency band (24a-z) is used. [4] Method according to claim 3, wherein, to determine the gain value (Ga`-Gz`), the first input signal (14) is compared with the output signal (18) frequency band by frequency band, and / or a signal gain applied along a signal path from the first input transducer (2) to the output transducer (6) is tracked, and wherein the gain value (Ga`-Gz`) is compared with a first limit value (36) dependent on said threshold. [5] Method according to one of the preceding claims, wherein a setting of a signal amplification made by a user of the hearing instrument (1) is taken into account for determining the relevant subset (25). [6] Method according to one of the preceding claims, wherein, in order to check the activation criterion (34), a characteristic value (38) is determined from the signal components (26c-x) of the first input signal (14) or of the first intermediate signal in the frequency bands (24c-x) of the relevant subset (25), which provides information about a noise component in said frequency bands (24c-x). [7] Method according to claim 6, wherein a signal-to-noise ratio (40) of the signal components (26c-x) in the frequency bands (24c-x) of the relevant subset (25) is determined as the characteristic value (38). [8] Method according to claim 6 or claim 7, wherein said characteristic value (38) is compared with a noise limit value (42) which corresponds to an upper limit for a permissible noise component, and wherein the activation criterion (34) for the activation of the sub-algorithm (32) is considered to be fulfilled if the noise limit value (42) is exceeded. [9] Method according to one of the preceding claims, wherein a second input signal (16) is generated from the ambient sound (12) by an acousto-electrical second input transducer (4) of the hearing instrument (1), wherein the output signal (18) is additionally generated from a frequency-selective signal processing of the second input signal (16), and wherein the sub-algorithm (32) comprises a directional microphone of the first input signal (14) or of the first intermediate signal and the second input signal (16) and / or of a second intermediate signal derived from the second input signal (16). [10] Method according to claim 9, wherein the activation criterion (34) is also checked on the basis of signal components of the second input signal (16) or the second intermediate signal only in the frequency bands (24c-x) of the relevant subset (25). [11] Method according to one of the preceding claims, wherein the sub-algorithm (32) - frequency band amplification, and / or - frequency band compression, and / or - includes noise reduction. [12] Hearing instrument (1) comprising at least one acousto-electrical first input transducer (2), one electro-acoustic output transducer (6), and a signal processing device (8), wherein the hearing instrument (1) is configured to carry out the method according to one of the preceding claims.
Citation Information
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