Method for suppressing an acoustic reverberation in an audio signal

Abstract

The invention relates to a method for suppressing acoustic reverberation in an audio signal (50), wherein an audio signal (50) is provided, wherein a first level measurement (p1) of the audio signal (50) is performed, wherein a second level measurement (p2) of the audio signal (50) is performed during the first level measurement (p1), wherein the first level measurement (p1) is performed using a first settling parameter and a first decay parameter such that the first level measurement (p1) has a first settling time (6) and a first decay time (16), wherein the second level measurement (p1) is performed using a second settling parameter and a second decay parameter such that the second level measurement (p2) has a second settling time (12) identical to the first settling time (6) and a second decay time (18) which is greater than the first decay time (6).and wherein a difference (Δ) is calculated between the first level measurement (p1) and the second level measurement (p2). It is provided that a reverberation noise level is estimated based on the difference (Δ) and the second level measurement (p2), and a gain parameter for the audio signal (50) is determined based on the first level measurement (p1) and the reverberation noise level (prn).

Description

[0001] The invention relates to a method for suppressing acoustic reverberation in an audio signal, wherein a first level measurement of the audio signal is carried out, wherein during the first level measurement a second level measurement of the audio signal is carried out, wherein the first level measurement has a first settling time and a first decay time, wherein the second level measurement has a second settling time identical to the first settling time and a second decay time which is greater than the first decay time, and wherein a difference is formed between the first level measurement and the second level measurement,

[0002] Acoustic reverberation usually occurs in closed or at least partially closed spaces as a result of multiple reflections of a generating sound event from the walls of a room and from other objects present in the room. The decay time of the reverberation varies depending on the geometry of the room and its walls, as well as the type, number, and geometry of the objects present in the room. This decay time is also influenced by the nature of the surfaces present in the room. In contrast to an echo, which is perceived in isolation as a kind of "repetition" of the generating sound event, reverberation essentially represents a continuous "reverberation" of the sound event.

[0003] While a minimum amount of reverberation is desirable for a pleasant sound experience, especially in music, to counteract an overly "dry," staccato-like sound, acoustic reverberation is often detrimental to the intelligibility of speech, as the characteristic sound events required to distinguish individual consonants, in particular, are very short-lived, and a corresponding overlay with reverberation can sometimes significantly distort the spectral information. Depending on the decay time, this can even become a problem for distinguishing formants for detecting vowels.

[0004] In hearing instruments that reproduce audio signals for a wearer, it is particularly important that the wearer's conversation partners hear speech that is as intelligible as possible, since a lack of acoustic understanding of a speech contribution and the associated loss of information noticeable to the wearer can be perceived as particularly noticeable and therefore particularly unpleasant. This applies particularly to hearing aids "in the narrower sense," which are often used to compensate for a hearing loss of the wearer in question. Therefore, hearing instruments, especially the aforementioned hearing aids, often incorporate technologies to improve the intelligibility of speech contributions.

[0005] However, especially when dynamic compression is frequently used, acoustic reverberation can have a particularly significant impact on speech intelligibility: Dynamic compression is intended in particular to amplify quiet sound events that are barely or no longer perceived by the wearer (either due to hearing loss on the part of the wearer or due to the generally low sound level of the sound event) to a level that is sufficiently perceptible, without the same amplification being applied to sufficiently loud sound events that can be perceived by the wearer without major problems, and thus further amplification could lead to an unpleasant volume.

[0006] However, this dynamic compression also "compresses" the acoustic reverberation, causing it to experience a correspondingly greater amplification than the sound event that generated it. As a result, the wearer perceives the decay time as longer in the given environment, and the intelligibility of speech contributions is impaired due to the relationships described above.

[0007] DE 10 2018 210 143 A1 discloses suppressing reverberation in an audio signal by performing two level measurements of the reverberation with different time constants and attenuating the audio signal based on the difference between the level measurements.

[0008] It is the object of the present invention to improve the above-described method for suppressing reverberation in an audio signal.

[0009] The stated object is achieved according to the invention by a method for suppressing acoustic reverberation in an audio signal, wherein an audio signal is provided, wherein a first level measurement of the audio signal is carried out, wherein a second level measurement of the audio signal is carried out during the first level measurement, wherein the first level measurement is carried out by means of a first control parameter and a first decay parameter such that the first level measurement has a first control time and a first decay time, wherein the second level measurement is carried out by means of a second control parameter and a second decay parameter such that the second level measurement has a second control time identical to the first control time and a second decay time which is greater than the first decay time, and wherein a difference is formed between the first level measurement and the second level measurement.

[0010] Here, it is provided that a reverberation noise level is estimated based on the difference and the second level measurement, and an amplification parameter for the audio signal is determined based on the first level measurement and the reverberation noise level. Advantageous and, in part, inventive embodiments are the subject of the dependent claims and the following description.

[0011] Suppression of acoustic reverberation in an audio signal particularly includes suppression of those signal contributions in the audio signal which arise from acoustic reverberation in the real acoustic situation represented by the audio signal. The audio signal is provided in particular by means of one or more electroacoustic transducers which convert said real acoustic situation into one or more, in particular, electrical signals. In order to provide the audio signal, pre-processing can be carried out on the basis of the thus generated, in particular electrical, signal(s), which can include, for example, digitization, amplification, dynamic compression or noise suppression. Acoustic reverberation particularly includes reflections of sound from a generating sound event on walls and / or objects, e.g.an at least partially closed space, wherein multiple reflections of the propagating sound generated by the sound event result in a continuous or almost continuous decay of the sound event at a fixed location.

[0012] In this context, a level measurement particularly encompasses the formation of a mathematical function by the level measurement, or the representation of the level measurement as such a function, by which an amplitude of the audio signal and / or an envelope of the amplitude and / or a square of the amplitude is mapped to a corresponding level value, preferably in a strictly monotonic manner and particularly preferably without an inflection point. In this context, particular attention should be paid to functions for which the relationship between their input variable and the mapped level value is not only logarithmic in nature; rather, the term "level measurement" should also encompass more general functions with suitable monotonic behavior.

[0013] The decay time of a level measurement is understood in particular to be the time that elapses after a signal contribution in the audio signal and a corresponding level deflection of the level measurement until the level measurement has dropped to zero or to a predetermined fraction of the level deflection in the absence of further signal contributions in the audio signal. The settling time of a level measurement is understood in particular to be the time that elapses after a spontaneously occurring, stationary signal contribution in the audio signal until the level measurement has reached a predetermined proportion of the asymptotic limit value for the signal level that corresponds to the stationary signal contribution. In this case, a shorter settling time means in particular a faster reaction of the level measurement to a spontaneously occurring signal contribution in the audio signal.

[0014] The settling time and decay time are set for each of the two level measurements using the respective settling and decay parameters. For example, if the level measurements are each implemented using an asymmetric, smoothing function of the amplitude (such as a recursive averaging function or similar), the settling parameter can be given by the weighting factor of the next amplitude contribution for the rising edge, and the decay parameter can be given by the weighting factor of the next amplitude contribution for the falling edge.

[0015] A sound event encompasses, in particular, any sound-generating event in the real acoustic situation that is represented by the audio signal and / or converted to provide the audio signal using appropriate transducers, whereby a clear temporal end can be assigned to the sound-generating event. In this sense, suppressing the acoustic reverberation of the sound event in the audio signal means, in particular, suppressing those signal contributions that correspond to the acoustic reverberation of the sound event in the real acoustic situation.

[0016] The decay behavior of the two level measurements, characterized by their respective decay times, depends on the specific behavior of the room in which the provided audio signal is recorded. Immediately after the sound event, the first wavefront enters the audio signal directly and without further reflection, followed by the first reflections from various boundaries and / or objects in the room, whose propagation delays depend particularly on the size of the room. These first reflections, which still represent a kind of attenuated and delayed version of the original sound event, are now incorporated into the audio signal on the one hand, but also generate further, cascading reflections on the other.With increasing order of reflection and superposition of the individual wavefronts, the distinguishability of individual reflections is lost; after the early reflections, which can usually still be isolated, an (almost) continuous, diffuse reverberation plume forms, which decays exponentially.

[0017] Based on the difference between the two level measurements, the contribution of diffuse reverberation in the audio signal can be determined at least approximately and implicitly, particularly with a suitable choice of the two decay times. This applies in particular if the second decay time is preferably adjusted using the second decay parameter (which is preferably estimated for the given environment or room) such that the decay behavior of the second level measurement is essentially determined by the contributions of the original sound event, which gradually become less and less important in the second level measurement. The first decay time is preferably adjusted using the first decay parameter such that the decay behavior of the first level measurement, at least after a short period of early reflections, is essentially determined by the contributions of diffuse reverberation, which continue to "feed" the decaying first level measurement.In particular, the short time of the early reflections can be recognized by the difference between the two level measurements, as can the diffuse reverberation that follows or results from it.

[0018] If the contribution of the diffuse reverberation or the decaying contribution of the sound event in the second level measurement is known, for example by applying a so-called minimum tracker to the difference between the second and the first level measurement (this difference is generally negative), and additionally determining when the minimum reached is left again (or exceeded by a predetermined minimum value), the reverberation noise level can be estimated from this knowledge on the basis of the second level measurement, which can in particular represent the proportion in the second level measurement that is (essentially) based on or corresponds to the contributions of the diffuse reverberation (instead of the actual, decaying sound event).

[0019] The reverberation noise level can then be used to determine a gain parameter so that, in particular, acoustic and especially diffuse reverberation of the sound event in the audio signal is suppressed by attenuating the audio signal depending on the gain parameter, particularly in response to a contribution from the sound event in the audio signal. The gain parameter can be determined as a gain factor, e.g., depending on the reverberation noise level as the noise signal and the first level measurement as the useful signal, which is applied to the audio signal.

[0020] The first level measurement and / or the second level measurement are expediently implemented using a weighted average function. This makes it particularly easy to implement the different decay behaviors of the level measurements with identical control behavior by applying different recursion weighting factors to the falling edge of the audio signal contributions newly added to the first or second level measurement. Preferably, a weighting factor for a subsequent value of the weighted average function is selected "asymmetrically" depending on a rising or falling level, i.e., for example,a value of the audio signal at a point in time is compared with the level value present at that point in time according to the weighted average function, and a weighting factor for the input of the new value of the audio signal into the level measurement is selected depending on whether the value of the audio signal is greater or smaller than the current value of the level measurement.

[0021] Advantageously, the first level measurement or the second level measurement is implemented by an asymmetric recursive low-pass filter, preferably of first order. For an audio signal a (n) in the discrete time domain, the respective level measurement pj (j = p1, p2) can then be represented as pj 2 n = 1 − c n ⋅ a 2 n + c n ⋅ pj 2 n − 1 with c n = c steig für a 2 n > pj 2 n − 1 , c fall otherwise, where pj 2< (n) denotes the level value of the level measurement pj at the discrete time index n, and c (n) a slope parameter which is set to one of the two constants c rise , c fall depending on the above condition for a 2< (n).

[0022] A physical decay time constant of a given environment is expediently determined in which a sound level of a sound signal on which the audio signal is based (i.e. in particular a sound signal from which the audio signal is generated) has dropped to a predetermined proportion of an initial value, wherein the second decay parameter is selected such that the second decay time of the second level measurement is given by said physical decay time constant for the given environment. Preferably, a drop of 60 dB is used as the drop to the predetermined proportion of an initial value, which corresponds to the time constant T60. A drop of 60 dB generally corresponds to a complete decay down to the noise floor.

[0023] According to the invention, the difference between the first level measurement and the second level measurement is compared with a defined first limit value, wherein, if the absolute value of the difference between the two level measurements exceeds the absolute value of the first limit value, the presence of a decaying contribution of the sound event and, in particular, a contribution of diffuse reverberation in the second level measurement is determined. This particularly comprises forming the difference, and, if this is negative, the presence of diffuse reverberation is determined if this difference lies below the now also negative first limit value. As soon as the difference exceeds the first limit value again (or the absolute value of the difference falls below the absolute value of the first limit value), diffuse reverberation no longer exists in this implementation, so that preferably no further attenuation occurs.The attenuation of the audio signal is preferably controlled depending on a comparison of the said difference with the first limit value.

[0024] According to the invention, if the difference between the two level measurements exceeds the first limit value, the difference between the difference and the limit value is determined as the decaying contribution of the sound event in the second level measurement. This particularly includes, in the case of a negative difference between the level measurements, the value by which the difference falls below the first limit value being determined as the contribution of the diffuse reverberation, which in particular is quantitatively included in the reverberation noise level. In other words, the first limit value provides, on the one hand, a binary criterion for whether diffuse reverberation or a decaying contribution of the sound event is present at all, and, on the other hand, a quantitative measure for said decaying contribution if present. In particular, a minimum tracker can be used for said difference, and the determined minimum, for example,be compared with the first limit value. The decaying contribution of the sound event includes in particular those components in the second level measurement which are essentially or solely based on the actual sound event and which only gradually decrease or disappear due to the time delay resulting from the smoothing. It is particularly advantageous here if the second decay time of the second level measurement is selected as the decay time constant T60; during a phase of late, particularly diffuse reverberation, the first level measurement, which initially has a faster first decay time (and thus a faster decay behavior) for early reflections, also decays at this decay rate in the room due to the sound power of the diffuse reverberation.

[0025] Within the scope of the invention, a time-dependent correction function is generated based on the decaying contribution of the sound event, and in particular also based on the magnitude of the first threshold value. The reverberation noise level is generated by subtracting the correction function from the second level measurement. The time-dependent correction function can, in particular, be determined by a base value that depends on the first threshold value and on the contributions of the decaying sound event determined as described above.

[0026] The gain parameter is determined by spectral subtraction, which uses a quotient of the reverberation noise level and the first level measurement. The gain parameter can then be given, for example, as G n = 1 − prn n / p 1 n with prn (n) = p2 (n) - d (n) as the reverberation noise level, p1 and p2 as the first and second level measurements, and d (n) as the correction function according to d n = min p 1 n − p 2 n , th 1 with the (negative) first limit th1.

[0027] In an advantageous embodiment, the audio signal is broken down into a plurality of frequency bands, wherein the first level measurement and the second level measurement are each carried out frequency band by frequency, wherein the respective gain parameter is determined for a plurality of frequency bands, in particular by forming the difference between the two level measurements frequency band by frequency band, based on which the respective decaying contribution of the sound event in the frequency band and from this the respective reverberation noise level are determined, and wherein the respective gain parameter in the frequency band is applied to the signal component of the audio signal to suppress the acoustic reverberation. In this case, the gain parameter G(n) in equation (ii) is to be replaced by a corresponding plurality of gain parameters G(n,k) in the time-frequency domain, where k is the band index.

[0028] The invention further relates to a method for suppressing acoustic reverberation in an audio signal of a hearing instrument, in particular a hearing aid, wherein the audio signal is provided from a sound signal of the environment by means of an input transducer of the hearing instrument, and wherein acoustic reverberation in the audio signal is suppressed by the method described above, as well as a hearing instrument, which can in particular be provided as a hearing aid, with an input transducer for generating an audio signal and a signal processing unit which is set up to carry out the method described above.

[0029] The method in the hearing instrument and the hearing instrument itself share the advantages of the reverberation suppression method described above. The advantages described for the method for suppressing acoustic reverberation in an audio signal and its further developments can be applied analogously to the method in the hearing instrument and to the hearing instrument itself.

[0030] An embodiment of the invention is explained in more detail below with reference to the accompanying drawings, each of which shows schematically: Fig. 1 shows a time diagram of a first level measurement and a second level measurement of the same audio signal with identical settling times and different decay times, Fig. 2 shows a time diagram of a difference between the level measurements according to Fig. 1 , and a correction function determined therefrom, Fig. 3 in a time diagram the second level measurement and a correction function according to Fig. 2determined reverberation noise level, Fig. 4 in a time diagram of the reverberation noise levels after Fig. 3 and the first level measurement, and Fig. 5 in a block diagram of a hearing instrument.

[0031] Corresponding parts and sizes are provided with the same reference numerals in all figures.

[0032] In Figure 1The level values P of a first level measurement p1 (solid line) and a second level measurement p2 (dashed line) are shown schematically in a time diagram against a time t, each of which is carried out on an audio signal not shown in detail. At a time T0, an isolated sound event 4 (dotted line) is present in the audio signal, which on the one hand has a clearly defined end and on the other hand, due to the physical environment in which the audio signal was recorded for generation, causes contributions from acoustic reverberation in the audio signal (not shown). The sound event 4 is intended to have only a very short time duration z. The entire sound energy of the sound event 4 is therefore concentrated in this time duration z.This can be the case, for example, with a bang, a blow, a clap, but also with consonants of speech, especially plosives, as well as with similar sounds of very short duration.

[0033] The first level measurement p1 has a first settling time 6, which elapses after the time T0, at which the sound event 4 begins, until the first level measurement has assumed a predetermined proportion 8 of the asymptotic level 10, wherein the asymptotic level 10 corresponds to the level that the first level measurement would assume for a stationary, continuous sound event with a signal level identical to the sound event 4. The second settling time 12 of the second level measurement p2 is in this case identical to the first settling time 6 of the first level measurement.

[0034] For this reason, the first level measurement p1 and the second level measurement p2 have the same adjustment behavior and thus at time T1, which marks the end of the time period z and thus the end of the sound event 4, they assume the same maximum value 14 for the level, which is just below the asymptotic level 10.

[0035] At time T1, the decay behavior characterized by reverberation begins in the real acoustic situation, which is represented by the audio signal, so that the first level measurement p1 and the second level measurement p2 now also transition to their decay behavior according to the respective decay time.

[0036] The response and decay behavior of the first and second level measurements p1, p2 are shown schematically as linear, which also results in a peak in this representation at the transition from response to decay. However, the response can already increase more slowly before the transition, and the transition can also be smoothed.

[0037] The first level measurement p1 and the second level measurement p2 have a first decay time 16 and a second decay time 18, respectively, with the second decay time 18 being greater than the first decay time 16. The second decay time can be related to the decay time constant T60, after which a sound level has decreased by 60 dB from a maximum value, and is often used as a measure of decay in a room, as follows: The "decay rate" of the second level measurement is given by the difference between the maximum value 14 and the initial value 24, divided by the second decay time 18; in this case, this decay rate corresponds to a rate of 60 dB / T60. The first decay time 16 can, for example, be given by half of the second decay time 18 (or a similar value), and in particular determines the beginning of the decay behavior 15 of the first level measurement p1 (see dotted line for extrapolation of the first decay time 16).

[0038] The first and second level measurements p1, p2 are each implemented as an asymmetric recursive first-order low-pass filter, so that the respective decay time 16, 18 can be set via a corresponding filter decay parameter, which controls the retention of existing level values for the next time point in the level measurement p1, p2 (see equation (i), above). For the second level measurement p2, the second decay time 18 set in this way is slow enough that the decay behavior is determined by the linear decline of the filter. The acoustic contributions of the diffuse reverberation of the reverberation tail may contribute to the value of the second level measurement p2, but do not determine its decay behavior.

[0039] For the first level measurement p1, however, the first decay time 16 is such that after a first peak 15, which corresponds to the maximum value 14 and is determined by the sound event 4, and after a rapid decay 17 corresponding to the first decay time 16 (to which the first and early reflections in the audio signal may still contribute), a transition to a flatter slope 19 occurs at a time T2. In this slope, the gradually decreasing contributions of the diffuse reverberation actually "feed" the first level measurement p1, thus determining its decay behavior in this range.

[0040] Because the decay behavior in the flatter flank 19 of the first level measurement p1 is actually determined by the diffuse reverberation, which decays in the room with the decay time T60, this exponential decay behavior from time T2 in the logarithmic representation of the first level measurement p1 is parallel to the decay behavior of the second level measurement p2, until the first level measurement has dropped to the initial value 24 before the sound event 4 at a time T3, and the diffuse reverberation in the room in which the audio signal was generated has thus decayed. Due to the longer second decay time 18, the second level measurement p2, on the other hand, only drops to the initial value 24 at a later time T4. The initial value 24 can, for example, be given by a noise background of the audio signal.

[0041] In Figure 2 is shown schematically in a time diagram a difference Δ from the first level measurement p1 and the second level measurement p2 after Figure 1 shown (dashed line). Due to the fact that p1 and p2 are both equal to the initial value 24 up to time T0, and due to the identical first and second settling times 6, 12, they exhibit the same settling behavior up to time T1, the difference Δ = p1 - p2 is exactly 0 up to time T1. Due to the different first and second decay times 16, 18 (see Fig. 1 ), the difference Δ decreases to a minimum value min < 0 until time T2, and remains due to the identical decay behavior of the two level measurements p1, p2 between times T2 and T3 (see Fig. 1 , parallel course) at the said minimum value min. Only after the time T3, from which the first level measurement p1 has already taken the initial value 24, to which the second level measurement p2 still falls until the time T4 (see Fig. 1), the amount of the difference Δ decreases again, i.e. the difference Δ increases again to the value zero by time T4.

[0042] The difference Δ is then compared with a first limit value th1, whereby the comparison is used to determine, in a manner to be described later, a contribution of diffuse reverberation in the second level measurement p2. The first limit value is smaller in magnitude than the absolute value of the minimum value min, i.e. | th1 | < | min |, or 0 > th1 > min.

[0043] The first limit value th1 should preferably be selected such that the minimum value min of the difference Δ can be reliably identified by a corresponding comparison. Thus, it is assumed that if Δ < th1 (i.e., if the difference Δ is below the first limit value th1), a contribution 30 of diffuse reverberation is present in the audio signal. This is because, for the period between the respective times T2 and T3, the decay behavior of the first level measurement p1 is directly determined by said diffuse reverberation, while the decay behavior of the second level measurement p2 is still largely determined by the original sound event 4 due to the slow second decay time 18. However, the contribution 30 of the diffuse reverberation is also included in the second level measurement p2.

[0044] Based on the difference Δ = p1 - p2 and the first limit value, a correction function d is now formed (solid line) according to equation (iii). This correction function d is formed by the first limit value th1 up to a time T1' > T1 (with T1' < T2), and changes to the difference Δ for t > T1'. At a time T3' > T3 (with T3' < T4), the correction function d is again given by the first limit value th1.

[0045] The benefit of this correction function d is determined by Figure 3 clearly: In Figure 3 is schematically shown in a time diagram the second level measurement p2 (solid line) as in Figure 1 Furthermore, a reverberation noise level prn (dashed line) is entered, which is determined by the second level measurement p2, from which the correction signal d is Figure 2was subtracted. Since, for the period between times T2 and T3, the first level measurement p1 is determined by the contributions of the diffuse reverberation, and while said contributions are also included in the second level measurement p2 during this period, the latter is primarily determined by the contribution of sound event 4 decaying according to the second decay time 18 (i.e., the "isolated" decaying contribution, without newly added sound contributions), the correction function d between times T2 and T3 (or transitionally already between T1' and T3') is essentially given by the decaying contribution of sound event 4 in the second level measurement p2.In other words, between times T2 and T3, the actual sound power is essentially equal to the first level measurement p1, so the (negative) correction function d is added to the second level measurement p2 to also arrive at the actual sound power in the reverberation noise level prn during this interval. The second level measurement p2 is always higher than the actual sound power and thus also higher than the diffuse reverberation power.

[0046] The reverberation noise level prn = p2 + d is thus essentially determined by the diffuse reverberation during the said period. Before time T1' and after time T3', the reverberation noise level prn is determined by the constant offset resulting from the first limit value th1 in the correction function d.

[0047] In Figure 4is a schematic time diagram showing the reverberation noise level prn (dashed line) and the first level measurement p1 (solid line). In the period between times T2 and T3, in which the first level measurement p1 is determined by the diffuse reverberation, both of these lines essentially lie on top of each other. Even though a schematic, idealized representation is shown here, the real case will be similar, namely that both lines in the range of diffuse reverberation will each deliver almost identical values. If an amplification factor is now determined in a frequency band according to equation (ii) and applied to the signal component of the audio signal 50 in the frequency band, the diffuse reverberation (between times T2 and T3) can be suppressed, while the other contributions of the audio signal 50 in the frequency band are retained, since p1 > prn then applies there.

[0048] The basis of Figure 1The method described for suppressing acoustic reverberation in the audio signal can be implemented, in particular, frequency band by frequency band. For this purpose, the audio signal is divided into individual frequency bands, in each of which the first and second level measurements p1, p2 are performed as shown. The attenuation of the audio signal can then be controlled individually for each frequency band based on the gain parameter G determined in this frequency band as a function of the response and decay behavior.

[0049] In Figure 5A block diagram of a hearing instrument 40 is shown schematically, which has an input transducer 42, a signal processing unit 44, and an output transducer 46. The input transducer 42, which in this case is a microphone, generates the audio signal 50 from a sound signal 48 of the environment, which also includes a specific sound event 4. An acoustic reverberation of the sound signal 4 in the audio signal 50 can now be determined using Figure 1 or based on Figure 2 described manner in the signal processing unit 44. The resulting signal is further processed, in particular subjected to dynamic compression and frequency band-dependent amplification, and an output signal 52 is generated therefrom, which is converted by the output transducer 46 into an output sound signal 54.

[0050] Although the invention has been illustrated and described in detail by the preferred embodiment, the invention is not limited to this embodiment. Other variations may be devised by those skilled in the art without departing from the scope of the invention. List of reference symbols

[0051] 4Sound event 6First settling time 8Preset proportion 10Asymptotic level 12Second settling time 14Maximum value 15Peak 16First decay time 17Fall 18Second decay time 19Flatter slope 24Noise background 40Hearing aid 42Input transducer 44Signal processing unit 46Output transducer 48Sound signal 50Audio signal 52Output signal 54Output sound signal dCorrection function minMinimal value PPevel values p1First level measurement p2Second level measurement prnReverberation noise level tTime th1First limit value T0 - T4Time T1', T3'Time zDuration ΔDifference

Claims

1. Method for suppressing an acoustic reverberation in an audio signal (50), - wherein an audio signal (50) is provided, - wherein the first level measurement (p1) of the audio signal (50) is carried out, - wherein during the first level measurement (p1), a second level measurement (p2) of the audio signal (50) is carried out, - wherein the first level measurement (p1) is carried out by means of a first attack parameter and a first release parameters such that the first level measurement (p1) has a first attack time (6) and a first release time (16), - wherein the second level measurement (p1) is carried out by means of a second attack parameter and a second release parameters such that the second level measurement (p2) has a second attack time (12) identical to the first attack time (6) and a second release time (18), which is greater than the first release time (6), and - wherein a difference (Δ) is formed from the first level measurement (p1) and the second level measurement (p2), characterized in that - on the basis of the difference (Δ) of the first level measurement (p1) and the second level measurement (p2) and a defined first limiting value (th1), a time-dependent correction function (d) is generated as the minimum of said difference (Δ) and the first limiting value (th1), - on the basis of the difference (Δ) and on the basis of the second level measurement (p2), a reverberation interference level is estimated, and the reverberation interference level (prn) is generated from an addition of the correction function (d) to the second level measurement (p2), and - on the basis of the first level measurement (p1) and on the basis of the reverberation interference level (prn), an amplification parameter for the audio signal (50) is determined, and - the audio signal (50) is attenuated as a function of the amplification parameter (G).

2. Method according to Claim 1, wherein upon a contribution of the sound event (4) in the audio signal (50), and acoustic reverberation of the sound event (4) in the audio signal (50) is suppressed by an attenuation of the audio signal (50) as a function of the amplification parameter, in particular by means of application of the amplification parameter to the audio signal (50).

3. Method according to Claim 1 or Claim 2, wherein the first level measurement (p1) and / or the second level measurement (p2) is implemented by a weighted mean value function.

4. Method according to Claim 3, wherein a weighting factor for a following value of the weighted mean value function is selected as a function of a rising or falling level.

5. Method according to any one of the preceding claims, wherein the first level measurement (p1) or the second level measurement (p2) is implemented by an asymmetrical recursive low-pass filter, preferably of first order.

6. Method according to any one of the preceding claims, wherein a physical release time constant of existing surroundings is determined, in which a sound level of a sound signal (48), on which the audio signal (50) is based, has sunk to a predetermined proportion of a starting value, wherein the second release parameter is selected such that the second release time (18) of the second level measurement (p1) is given by said physical release time constant for the existing surroundings.

7. Method according to any one of the preceding claims, wherein the audio signal (50) is decomposed into a plurality of frequency bands, wherein the first level measurement (p1) and the second level measurement (p2) are each carried out by frequency band, wherein for a plurality of frequency bands, the respective amplification parameter is determined, and is applied to the signal component of the audio signal (50) in the respective frequency band to suppress the acoustic reverberation.

8. Method for suppressing an acoustic reverberation in an audio signal (50) of a hearing instrument (40), wherein the audio signal (50) is provided by means of an input transducer (42) of the hearing instrument (40) from a sound signal (48) of the surroundings, and wherein an acoustic reverberation in the audio signal (50) is suppressed by a method according to any one of the preceding claims.

9. Hearing instrument (40) having an input transducer (42) for generating an audio signal (50) and a signal processing unit (44), which is configured to carry out the method according to any one of Claims 1 to 7.

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