Method for processing ambient sound which is captured by an audio device that can be worn on or in the ear, and corresponding device

Abstract

The invention relates to a method for processing ambient sound which is captured by an audio device (10) that can be worn on or in the ear, in which method a reference signal is generated from the captured ambient sound (S1). A measurement signal is generated from the sound captured in the ear canal of a user of the audio device that can be worn on or in the ear (S1). The reference signal is processed into a first signal and the measurement signal is processed into a second signal (S2). A factor is determined on the basis of the first and second signals (S3) such that a measure of the signal strength of the second signal is aligned with a measure of the signal strength of the first signal. A third signal based on the reference signal is weighted with the determined factor (S4). The third signal, which is weighted with said determined factor, is output (S5).

Description

[0001] Description

[0002] Method for processing ambient sound captured by an on-ear or in-ear audio device, and corresponding device

[0003] The present invention relates to a method for processing ambient sound captured by an on- or in-ear audio device. The on- or in-ear audio device can be, in particular, headphones or a hearing aid, but can also be other audio devices, such as hearing protection, glasses with audio functionality, or a headset. The present invention further relates to a corresponding device.

[0004] Headphones with an ambient mode (also known as transparency, hear-through, or pass-through mode) compensate for the passive attenuation of ambient sound caused by the headphones. Such an ambient mode can, for example, aim to make ambient sound sound as if no headphones were being worn when headphones with an active ambient mode are worn. Such an ambient mode can also be used with hearing aids designed to compensate for hearing loss or improve speech intelligibility in noisy environments. Both headphones with an ambient mode and hearing aids with an ambient mode are referred to below as portable audio devices with an ambient mode. The fact that these devices can be worn on or in the ear goes without saying, and for the sake of simplicity, this will not be explicitly mentioned.Furthermore, the portable audio device can also include, for example, hearing protection, glasses with audio function or a headset.

[0005] In a portable audio device with an ambient mode, the ambient sound is recorded by an external microphone, converted into a digital signal by an analog-to-digital converter, processed on a digital signal processor, then converted into an analog signal by a digital-to-analog converter and played back through one or more speakers of the portable audio device. In addition to at least one speaker and at least one external microphone, such a portable audio device has an internal microphone which can be used to reduce structure-borne and ambient noise in the ear canal using a feedback control. For the most natural hearing result possible, the sound that reaches the eardrum is crucial. This sound depends in particular on the passive sound, the ambient sound passively dampened by the portable audio device, and the active sound reproduced by the speaker.Both the passive attenuation characteristics and the transfer function from the speaker to the eardrum depend on the specific fit of the portable audio device for the individual user. For example, a classic ambient mode for headphones can result in unpleasant exaggerations or unnatural attenuation of frequencies compared to the open ear due to the individual fit. To avoid such disruptive effects with a hearing aid, fine-tuning of the volume, timbre, and gain at various frequencies is necessary, which is often performed by a hearing aid acoustician, which is time-consuming and expensive.

[0006] It is an object of the invention to provide an improved method and a corresponding device which enables the most natural hearing result possible even in the case of partial or complete ear closure by an audio device worn on or in the ear.

[0007] This object is achieved by a method having the features of claim 1 and a corresponding device according to claim 10. Preferred embodiments of the invention are the subject of the dependent claims.

[0008] In a method according to the invention for processing ambient sound captured by an on- or in-ear audio device, the following steps are performed: a reference signal is generated from the captured ambient sound; a measurement signal is generated from the captured sound in the ear canal of a user of the on- or in-ear audio device; the reference signal is processed into a first signal and the measurement signal into a second signal; a factor is determined based on the first and second signals, such that a measure of the signal strength of the second signal is adjusted to a measure of the signal strength of the first signal; a third signal based on the reference signal is weighted with the determined factor; and the third signal weighted with the determined factor is output. According to a preferred embodiment of the invention

[0009] - the first and second signals are divided into frequency bands to generate first and second band signals;

[0010] - a factor is determined for each band based on the respective first and second band signals, so that a measure of the signal strength of the respective second band signal is adjusted to a measure of the signal strength of the respective first band signal;

[0011] - the determined factors or a filter based on the determined factors are applied to the third signal in such a way that the third signal is weighted by the factors in corresponding frequency bands;

[0012] - the weighted third signal is output.

[0013] According to a further preferred embodiment of the invention

[0014] - the third signal is divided into frequency bands to generate third band signals;

[0015] - the third band signals are multiplied by the respective determined factors;

[0016] - the band signals multiplied by the respective determined factors are summed;

[0017] - the sum signal is output.

[0018] According to a further preferred embodiment, the reference signal is processed into a first signal and the measurement signal is processed into a second signal such that the ratio of the signal strength of the first and second signals corresponds to the ratio of the sound pressure level at the eardrum of the user with an open ear canal and one closed by the audio device worn on or in the ear.

[0019] Advantageously, the first signal is additionally processed by an amplification factor, dynamic processing, and / or noise suppression.

[0020] According to a further preferred embodiment, the factor is calculated recursively as a quotient of a measure of the signal strength of the first signal weighted by the factor and the second signal.

[0021] According to yet another preferred embodiment, the third signal weighted by the factor is delayed and the factor is calculated as a quotient of a measure of the signal strength of the delayed third signal multiplied by the factor and the second signal.

[0022] Advantageously, the output signal is filtered by a forward filter and then reproduced via a loudspeaker of the on-ear or in-ear audio device.

[0023] According to a further embodiment of the invention, the reference signal is filtered by a forward filter before it is processed into the first signal and the output signal is reproduced via a loudspeaker of the on-ear or in-ear audio device.

[0024] In particular, in an ambient mode of the on-ear or in-ear audio device, at least partial compensation for a passive attenuation of the ambient sound caused by the on-ear or in-ear audio device is advantageously carried out, which is individually adapted to the user of the on-ear or in-ear audio device.

[0025] Accordingly, a device according to the invention for processing ambient sound which is detected by an on- or in-ear audio device comprises a processor which is configured to: generate a reference signal from the detected ambient sound; generate a measurement signal from the detected sound in the ear canal of a user of the on- or in-ear audio device; process the reference signal into a first signal and the measurement signal into a second signal; determine a factor based on the first and second signals such that a measure of the signal strength of the second signal is adjusted to a measure of the signal strength of the first signal; weight a third signal based on the reference signal with the determined factor; and output the third signal weighted with the determined factor.

[0026] According to one embodiment of the invention, the device according to the invention comprises a reference sensor for detecting the ambient sound and generating a reference signal based on the detected ambient sound; a measuring sensor for detecting the sound in the ear canal of a user of the on-ear or in-ear audio device and generating a measurement signal based on the detected sound; and a loudspeaker for reproducing an audio signal based on the ambient sound processed by the processor.

[0027] The processor is advantageously configured to apply at least one filter bank to divide the first and second signals into frequency bands.

[0028] Furthermore, the processor may be configured to apply the at least one filter bank in addition to dividing the third signal into frequency bands.

[0029] The device according to the invention can in particular be integrated into the audio device worn on or in the ear.

[0030] In particular, the audio device worn on or in the ear can be designed as a headphone or hearing aid.

[0031] The invention also relates to a computer program with instructions that cause a computer to carry out the steps of the method according to the invention.

[0032] Further features of the present invention will become apparent from the following description and claims in conjunction with the figures.

[0033] Fig. 1 shows schematically an in-ear headphone in the ear canal with essential electronic components;

[0034] Fig. 2 shows a schematic flow diagram of the method according to the invention;

[0035] Fig. 3 shows a first block diagram of an inventive processing of an ambient sound detected by a sensor;

[0036] Fig. 4 shows a second block diagram of an inventive processing of ambient sound detected by a sensor; Fig. 5 shows a third block diagram of an inventive processing of ambient sound detected by a sensor;

[0037] Fig. 6 shows a block diagram of a calculation of a factor according to the invention;

[0038] Fig. 7 shows schematically the magnitude response of individual bandpass filters of a filter bank;

[0039] Fig. 8 shows a block diagram of an apparatus according to the invention for determining the factors in frequency bands;

[0040] Fig. 9 shows a block diagram of an apparatus according to the invention for applying the factors in a filter bank structure;

[0041] Fig. 10 shows a block diagram of another device according to the invention for applying the weighting rule in a filter bank structure with processing of the reference signal;

[0042] Fig. 11 shows a block diagram of a hybrid portable audio device with inventive processing of ambient sound detected by a sensor.

[0043] To better understand the principles of the present invention, embodiments of the invention are explained in more detail below with reference to the figures. It is understood that the invention is not limited to these embodiments and that the described features may also be combined or modified without departing from the scope of the invention as defined in the claims.

[0044] Fig. 1 shows an example of a portable audio device 10 in the form of in-ear headphones, in which the method according to the invention can be carried out. However, the method can equally well be used in other types of headphones, hearing aids, hearing protectors, headsets or glasses with audio functionality. The in-ear headphones 10 are held by an ear insert 17 in the ear canal 15 of a user and acoustically seal it completely or partially. Depending on the fit of the ear insert 17 for an individual user, the ear canal 15 can be opened more or less by a ventilation 18. The portable audio device 10 can also have targeted ventilation in the housing or in the ear insert 17.

[0045] The headphones 10 are equipped with at least one reference sensor in the form of an external microphone 11, which detects the ambient sound and generates a reference signal based thereon. Furthermore, at least one measurement sensor in the form of an inward-facing microphone 12 is provided on the side of the headphones directed in the ear canal toward the eardrum 16, with which the sound in the ear canal is detected and a measurement signal is generated based thereon. The headphones 10 are equipped with at least one processor 14, which processes the ambient sound recorded by the external microphone 11 according to the invention. The at least one processor 14 can in particular be one or more digital signal processors. Based on the ambient sound processed by the processor 14, an audio signal is then reproduced via the loudspeaker 13 inside the headphones.The processed ambient sound reproduced via the loudspeaker can then, for example, result in users being able to hear their surroundings clearly when wearing a portable audio device 10 despite the passive attenuation of ambient sound caused by the earpiece, or in the user's hearing being compensated for, or in the user's hearing being protected from short-term loud noises.

[0046] The acoustic impedance driven by the loudspeaker 13 can depend not only on the ventilation 18 but also on other individual factors such as the ear geometry or the impedance of the user's eardrum. As a result, the transfer function from the loudspeaker 13 to the inner microphone 12 and to the eardrum 16 can vary from user to user. Likewise, the passive damping characteristics of the portable audio device 10 can also differ from person to person. For a larger ventilation 18, for example, more ambient sound can passively reach the eardrum 16. These individual acoustic properties can lead to an undesirable sound pressure level at the eardrum 16. The relative transfer function from an inner microphone 12 to the eardrum 16 and the sound pressure level applied to the outer microphone 11, on the other hand, depend less strongly on the individual acoustic properties.

[0047] Based on the reference signal, a target level can be determined that should be present at the eardrum, regardless of the individual acoustic properties. Based on the measurement signal, however, an actual level that is currently present at the eardrum and is subject to the individual acoustic properties can be estimated. One goal of the method according to the invention is to adjust the actual level to the target level.

[0048] Fig. 2 schematically shows a flowchart of the method according to the invention with the basic steps for processing ambient sound detected by a sensor of an on-ear or in-ear audio device. This can be used in particular for customizing a transparency mode for headphones or a hearing aid, but is not limited to this.

[0049] In the method, sensor signals are recorded in a first step S1. In particular, ambient sound is recorded by an external microphone and a reference signal is generated, and the sound in the ear canal is recorded by an internal microphone and a measurement signal is generated. In a second step S2, the reference and measurement signals are each pre-processed into a first and a second signal. In a third step S3, a factor is calculated based on the first and second signals. In a fourth step S4, a third signal based on the reference signal is weighted with this factor. In a fifth step S5, the third signal weighted with the determined factor is output. The weighting with the determined factor can in particular be carried out by multiplying by the factor.

[0050] Steps S1 to S5 can then be followed by further processing steps not shown in the flow chart, such as a digital-to-analog conversion and subsequent playback via one or more loudspeakers.

[0051] The invention is further explained below using several block diagrams that represent implementations of various embodiments of the method. The block diagrams each illustrate the signal flow and, through the blocks shown, elements for processing the signals. The functions of the various elements shown in the figures can be provided by corresponding software modules or dedicated hardware units of a processor. For reasons of clarity, the function of individual blocks is described only once, unless they differ between the various block diagrams.

[0052] Fig. 3 shows a first block diagram of an inventive processing of ambient sound detected by a sensor. The reference signal x(n) from the external microphone is filtered by an equalizer 25 to generate the open eardrum signal o(n). The open eardrum signal corresponds to an estimate of the signal that would reach the eardrum without wearing the portable audio device. The measurement signal e(n) is filtered by the equalizer 24 to generate the closed eardrum signal c(n). The closed eardrum signal corresponds to an estimate of the signal that currently reaches the eardrum while wearing the portable audio device.

[0053] The equalizers can be determined, for example, through artificial head measurements. For a defined external excitation signal, the signal at the open eardrum microphone of an artificial head can first be measured, with the portable audio device not inserted. The device can then be inserted, and the signal from the external and internal microphones, as well as the signal from the closed eardrum microphone, can be recorded with the same excitation. Equalizer 25 can then be determined as a relative transfer function from the external microphone signal to the open eardrum signal, and equalizer 24 can be determined as a relative transfer function from the internal microphone signal to the closed eardrum signal.

[0054] By estimating the signal strength 22, 22', the signal strengths o0 and a can be calculated based on the open and closed eardrum signals o(n) and c(n). cwhich correspond to the ratio of the sound pressure levels at the open and closed eardrum. The estimators 22, 22' can be implemented, for example, as exponential smoothers of the magnitude of a signal, with a smoothing factor 0 < A < 1.0: o (n) = A ■ a x (n - 1) + (1 - A) • |x(n) |

[0055] Smoothing a squared signal to calculate a short-term power, followed by taking the square root, is also possible.

[0056] The reference signal x(n) is now multiplied by a factor g(n) to generate the modified signal x(n) = g(n) ■ x(n). The signal x(ri) can then be further processed and reproduced via a loudspeaker 13 of the portable audio device 10. It is assumed here that the sound pressure level at the eardrum caused by the signal x(n) is higher than the sound pressure level of the ambient sound, which reaches the eardrum via other paths. This assumption applies in particular when high frequency components are attenuated by the passive attenuation of the portable audio device 10 and low frequency components of the ambient sound are attenuated, for example, by a feedback controller 52, as shown in Fig. 11. The signal strength of the measurement signal u e (n) is proportional to the signal strength of the modified signal u^(n), which is approximately equal to the signal strength of the reference signal scaled by the factor: o e )~Ox ) ~ g(n)a x(n)

[0057] Since the modified signal calculated with the factor is played back via a loudspeaker 13 and correspondingly recorded by the internal microphone 12, the signal strength of the measurement signal, according to the previous equation, and accordingly also the signal strength of the closed eardrum signal, is already subject to the influence of the factor. To compensate for this influence, the factor can, for example, either be divided by the factor delayed by a delay element 21 before calculating the signal strength of the closed eardrum signal c(n) by an estimator 22', or, as shown in Fig. 3, be multiplied by the factor before calculating the signal strength of the open eardrum signal x(n) by an estimator 22' in order to calculate the signal strength cr0> (n). The ratio of crö(n) to u c(n) then corresponds to the ratio of the sound pressure level at the open and closed eardrums, if x(n) = x(n). The factor g(n) can then be calculated in block 23, for example, as a quotient of these signal strengths:

[0058] By multiplying 26 the factor g(ri) by the reference signal x(ri) to generate x(n) and reproducing x(n) via a loudspeaker 13 of the portable audio device 10, the sound pressure level at the eardrum when wearing the portable audio device (actual level) is then adjusted, with the previously described equalizers 24 and 25, to the sound pressure level at the eardrum that would be present without wearing the portable audio device (target level).

[0059] The equalizers 24 and 25 can also be designed differently than previously described in order to achieve a different goal, such as acoustic transparency.

[0060] For example, block 25 may also include dynamic processing or noise suppression to determine an improved target level that protects the user's hearing from loud noises or implements a hearing improvement. An example of this is shown in Fig. 10.

[0061] Fig. 4 shows a second block diagram of an inventive processing of ambient sound detected by a sensor. Here, the equalizer 24 can implement a transfer function corresponding to the quotient of the transfer functions of the equalizers 25 and 24, as previously explained with reference to Fig. 2. As a result, the ratio of the signal strength of the equalized measurement signal and the reference signal continues to correspond to the ratio of the sound pressure level at the open and closed eardrums, so that the equalizer 25 can be omitted. This saving is particularly advantageous when the method is used in a filter bank structure, since it eliminates the need for either an additional filter bank or duplicates of the equalizer 25.

[0062] Fig. 5 shows a third block diagram of an inventive processing of ambient sound detected by a sensor. Here, the factor is not calculated based on the signal strength of the reference signal multiplied by the delayed factor, but instead on the signal strength of the modified signal x(n) delayed by a delay element 21. This eliminates the need for a multiplication 26.

[0063] Fig. 6 shows a block diagram of a block 20 according to the invention for calculating the factor g(n), which can be used, for example, in the following filter bank implementations. This structure is characterized by the fact that it uses a measurement signal e that has already been modified, for example, by an equalizer 24. k (n) and a reference signal x k (n) and only a factor g k (n) outputs.

[0064] Fig. 7 shows an example of the magnitude response 30 of a filter bank that divides an input signal into K frequency bands. Each filter, with the impulse response b k (n), the filter bank should ideally be designed to filter out frequency components of the input signal below a lower cutoff frequency f g k and above an upper limit frequency f g , k+1 blocks, as well as frequency components between f g k and f g , k+1 This results in the following objective function for the magnitude response of a filter depending on the frequency f : 0, if f < fg, k < if / 5, fe < f < f g , k+1 0, otherwise

[0065] Furthermore, the sum of the bandpass filters should be 30 Ideally, sf < fg :1 < fg :K+1 otherwise, so that there is no unwanted cancellation or amplification, especially in the transition areas when the bandpass filters are connected in parallel.

[0066] Fig. 8 shows a block diagram of an inventive device for applying a block 20 in a filter bank structure 40. As in Fig. 3, the reference and measurement signals are filtered by respective equalizers 24, 25 and converted into a modified reference and measurement signal. The modified reference signal and the modified measurement signal are each converted into band signals x by a filter bank analysis 41. k (n) and e k (n). These band signals are then fed to an algorithm 20, for example as shown in Fig. 6, to calculate a factor g per band k(n). The bold lines in the block diagram indicate that these are multi-channel signals and factors. The factors are then fed to a unit 45, which performs a filtering of x(n) so that x(n) in corresponding frequency bands with the factors g k (n). The coefficients of the filter can be weighted based on the factors g k (n) are calculated, selected, or otherwise determined. Likewise, filtering can be implemented in the frequency domain with appropriate weighting of frequency coefficients. Fig. 9 shows another form of implementation of unit 45 by weighting band signals.

[0067] Implementing the method in a filter bank structure is advantageous because different factors can be applied in different frequency bands. In a single-channel implementation of the method, for example, it may frequently occur that the measurement level is too low at low frequencies and too high at high frequencies compared to the reference level, but these deviations cannot be compensated simultaneously.

[0068] Fig. 9 shows a further block diagram of an apparatus according to the invention for applying a block 20 in a filter bank structure 40. Similar to Fig. 4, the measurement signal is filtered by an equalizer 24. The reference signal and the modified measurement signal are each converted into band signals x by a filter bank analysis 41. k (n) and e k(n) and fed to an algorithm 20. The unit for filtering the reference signal 45 from Fig. 8 is designed in this case such that the factors per band are calculated with the reference band signals x k (n) and then summed in a filter bank synthesis step 42 and output in the form of x(n).

[0069] Fig. 10 shows a further block diagram of an inventive device for applying a weighting algorithm 20 in a filter bank structure 44. Here, the reference band signals x k(n) supplied to the algorithm 20 are first processed by a block 43, whereby the signal strength of the reference band signals supplied to the algorithm is modified. This block can correspond, for example, to dynamic compression, noise reduction, or simple scaling. This can protect the user's hearing from loud noises, improve hearing through a noise-suppressed ambient signal, and / or adjust the sound perception through user settings.

[0070] The parameters of block 43 can also be configured by the user. For example, users can select a factor with which the reference band signals are scaled in block 43. Since the factors of the algorithm are calculated in block 20 based on the scaled reference band signals, and the reference band signals are in turn scaled with the factors of the algorithm before synthesis 42, the measurement level, and accordingly the sound pressure level at the user's eardrum when wearing the portable audio device, is scaled according to the reference level modified by the user. Accordingly, the reference band signals can be cleaned up by noise reduction or compressed by a dynamic range compressor, so that the measurement level is adjusted by algorithm 20 according to this noise reduction or dynamic range compression. Such a process 43 can also be implemented in the block 20 shown in Fig. 3 and Fig.4, for example in Fig. 3 behind the equalizer 25, or in Fig. 4 after the branching of the reference signal and before the multiplication 26 with the delayed factor.

[0071] Fig. 11 shows a block diagram of a hybrid portable audio device 10 with method 44 according to the invention, here using the filter bank structure from Fig. 10 as an example. The portable audio device is equipped with an outer microphone 11, an inner microphone 12 and a loudspeaker 13.

[0072] An audio signal a(n), e.g., music or a telephone call, can be filtered by an equalizer 50 and then reproduced via the loudspeaker 13. In addition, the signal e(n) of the internal microphone 12 is offset against the output signal of a secondary path estimator 53, filtered by a feedback controller 52, and also reproduced via the loudspeaker 13. Since the signal reproduced by the loudspeaker 13 into the ear canal is recorded by the internal microphone 12, a closed control loop is formed with the controller 52. The controller 52 can be designed so that sound in the ear canal, e.g., passively attenuated ambient sound or structure-borne sound, is attenuated in certain frequency ranges. In this way, for example, the occlusion effect caused by the closure of the ear canal can be compensated.

[0073] The secondary path estimate 53 corresponds to an estimate of the transfer function from the loudspeaker 13 to the internal microphone 12, the so-called secondary path. The secondary path estimate 53 can be used to reduce the influence of the controller 52 on the audio signal a(n) or the output signal of the forward filter 51. The switch 54 can be set such that in switch position 1 only the equalized audio signal is fed to the secondary path estimate, in position 2 no signal is fed to it, and in position 3 the sum of the equalized audio signal and the output signal of the forward filter is fed to it.

[0074] The signals from the outer microphone 11 and inner microphone 12 are fed to the device 44. The processed ambient sound signal is output by the device 44 and filtered through a forward filter 51 before being reproduced via the loudspeaker together with the signal from the controller 52 and the equalizer 50. Advantageously, the forward filter can be designed such that the sound pressure level caused by ambient sound at the eardrum when the portable audio device 10 is worn corresponds to the sound pressure level that would be present at the eardrum if no portable audio device were worn. However, the forward filter can also be designed differently. For example, the sound pressure level can be implemented only in a specific frequency range corresponding to the open ear, or a targeted amplification / attenuation can be set.

[0075] Contrary to the block diagram shown in Fig. 11, it is also possible to feed the signal of the inner microphone 12, combined with the output signal of the secondary path estimator 53, to the processing stage 44 instead of the signal of the inner microphone 12 by performing the addition 55. In this way, for example, the influence of the audio signal on the processing of ambient sound can be reduced.

[0076] It is also possible to first apply the forward filter to the signal from the external microphone before feeding it to process 44. A further forward filter after the process can then be omitted. Ideally, the influence of the forward filter should be taken into account when determining equalizers 24 and 25.

[0077] It is also possible to feed the sum of the audio signal, optionally equalized by the equalizer 50, and the signal of the external microphone 11, optionally already filtered by a forward filter 51, to the method instead of the signal of the external microphone 11. For example, the inventive processing unit 44 can adjust not only the level of the ambient sound but also the level of an audio signal. It is also possible to feed only the equalized audio signal to the processing unit 44. As a result, only the level of the audio signal would be adjusted by the inventive method.

[0078] List of reference symbols

[0079] headphones

[0080] External microphone

[0081] Internal microphone

[0082] Speaker

[0083] Signal processor

[0084] ear canal

[0085] eardrum

[0086] Ear insert

[0087] Ventilation

[0088] Block for calculating a factor from modified measurement signal and reference signal

[0089] Delay element, 22' signal strength estimator

[0090] Block for calculating a factor from signal strengths

[0091] Equalization filter for a second signal

[0092] Equalization filter for a first signal, 26' multiplier

[0093] Magnitude response of a filter of the filter bank

[0094] Device for applying factors in a filter bank structure

[0095] Filter bank analysis

[0096] Filter bank synthesis

[0097] Process for preparing a first signal

[0098] Device for applying factors in a filter bank structure with processing

[0099] Unit for filtering a third signal based on the factors

[0100] Equalization filter for external audio signals

[0101] Forward filter

[0102] Feedback controller

[0103] Secondary path estimation Switch for selecting the input signal of the secondary path estimation, 55', 55" Adder Process step for acquiring the sensor signals Process step for preprocessing the sensor signals Process step for calculating a factor Process step for applying the factor Process step for outputting the resulting signal

Claims

Claims 1. Method for processing ambient sound which is detected by an on- or in-ear audio device (10), wherein a reference signal is generated (S1) from the detected ambient sound; a measurement signal is generated (S1) from the detected sound in the ear canal of a user of the on- or in-ear audio device; the reference signal is processed into a first signal and the measurement signal is processed into a second signal (S2); a factor is determined (S3) based on the first and second signals, such that a measure of the signal strength of the second signal is adjusted to a measure of the signal strength of the first signal; a third signal based on the reference signal is weighted (S4) with the determined factor; and the third signal weighted with the determined factor is output (S5).

2. The method according to claim 1, wherein the first and second signals are divided into frequency bands to generate first and second band signals; a factor is determined for each band based on the respective first and second band signals, so that a measure of the signal strength of the respective second band signal is adjusted to a measure of the signal strength of the respective first band signal; the determined factors or a filter based on the determined factors are applied to the third signal such that the third signal is weighted with the factors in corresponding frequency bands; the weighted third signal is output.

3. The method according to claim 2, wherein the third signal is divided into frequency bands to generate third band signals; the third band signals are multiplied by the respective determined factors; the band signals multiplied by the respective determined factors are summed; the sum signal is output.

4. Method according to one of the preceding claims, wherein the reference signal is processed into a first signal and the measurement signal is processed into a second signal (S2) such that the ratio of the signal strength of the first and second signals corresponds to the ratio of the sound pressure level at the eardrum of the user with the ear canal open and the ear canal closed by the on-ear or in-ear audio device.

5. The method according to claim 4, wherein the first signal is additionally processed by an amplification factor, a dynamic processing, and / or a noise suppression.

6. Method according to one of the preceding claims, wherein the factor is calculated recursively as a quotient of a measure of the signal strength of the first signal weighted with the factor and the second signal (S3).

7. The method according to claim 5, wherein the third signal weighted by the factor is delayed and the factor is calculated as a quotient of a measure of the signal strength of the delayed third signal weighted by the factor and the second signal (S3).

8. Method according to one of the preceding claims, wherein the output signal is filtered by a forward filter and then reproduced via a loudspeaker of the on-ear or in-ear audio device.

9. Method according to one of the preceding claims 1 to 7, wherein the reference signal, before being processed into the first signal, is filtered by a forward filter and the output signal is reproduced via a loudspeaker of the on-ear or in-ear audio device.

10. Method according to one of the preceding claims, wherein in an ambient mode an at least partial compensation of a passive attenuation of the ambient sound caused by the on- or in-ear audio device takes place, which is individually adapted to the user of the on- or in-ear audio device.

11. Device for processing ambient sound which is detected by an on-ear or in-ear audio device (10), comprising a processor (14) configured to generate a reference signal from the detected ambient sound; to generate a measurement signal from the sound detected in the ear canal of a user of the on- or in-ear audio device; to process the reference signal into a first signal and the measurement signal into a second signal; to determine a factor based on the first and second signals so that a measure of the signal strength of the second signal is adjusted to a measure of the signal strength of the first signal; to weight a third signal based on the reference signal with the determined factor; and to output the third signal weighted with the determined factor.

12. The device according to claim 11, comprising a reference sensor (11) for detecting the ambient sound and generating a reference signal based on the detected ambient sound; a measurement sensor (12) for detecting the sound in the ear canal of a user of the on-ear or in-ear audio device and generating a measurement signal based on the detected sound; and a loudspeaker (13) for reproducing an audio signal based on the ambient sound processed by the processor (14).

13. The apparatus of claim 11 or 12, wherein the processor (14) is further configured to apply at least one filter bank (41) to divide the first and second signals into frequency bands.

14. The apparatus of claim 13, wherein the processor (14) is further configured to apply the at least one filter bank (41) in addition to dividing the third signal into frequency bands.

15. Device according to one of the preceding claims 11 to 14, wherein it is integrated into the on-ear or in-ear audio device.

16. Device according to claim 15, wherein the on-ear or in-ear audio device is designed as a headphone (10) or hearing aid.

17. A computer program comprising instructions for causing a computer to carry out the steps of a method according to any one of claims 1 to 10.

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