NO313400B1 - Noise terminal for noise control - Google Patents

Abstract

Ear protecting device for being positioned in the ear meatus (3) of a human, and comprising an inner microphone (M2), wherein the inner microphone (M2) has a sound inlet (S2) for being directed toward the meatus (3), arranged for converting a picked-up sound signal (52) to an output signal (51); the ear protecting device also comprising a an electronics unit (11) including sound analyzing means arranged for analyzing the output signal (51) and comparing the signal with predetermined signals corresponding to acceptable noise limits and for activating an indicator (72) when detecting a passing of these limits.

Description

The invention relates to the physical design of an adapted hearing protection earplug, combined with an acoustic communication terminal.

There are a number of solutions for hearing protection and acoustic communication adapted to noisy environments, which are based on earplugs and earmuffs with earphones (loudspeakers), hoop microphones, cheek microphones or throat microphones. All of these solutions have one or more of the following undesirable properties:

heavy and bulky

uncomfortable

inferior quality for sound reception and sound reproduction poor sound attenuation

attenuation of both desired and unwanted sounds.

An object of this invention is to provide an ear terminal which does not have any of these shortcomings, which is light in weight, which constitutes an intelligent in-ear hearing protector with wireless communication. The noise reduction is automatically adapted to the noise conditions and communication mode. The present invention therefore simultaneously protects the hearing and provides improved characteristics for communication in different environments with noise. It is intended for continuous use during the working day or in other periods where hearing protection and/or voice communication is necessary.

The invention also relates to a device for utilizing the speech sounds produced in the ear of a person wearing hearing-protective communication earplugs according to the present invention.

Today's devices, which are intended to capture speech from a person in a very noisy environment, represent a technological challenge and can take several forms. Common types include

A microphone in close proximity to the mouth, carried on a microphone arm.

The microphone is given a characteristic that particularly emphasizes the near field from the mouth. This type of microphone is sometimes referred to as "noise-cancelling".

A vibration sensor in contact with the throat, which picks up the vibrations

from the vocal cords.

A vibration sensor in contact with the wall of the meatus, the external ear canal,

which picks up the vibrations in the tissue in the head.

A corresponding sensor in contact with the cheekbone.

These types of devices are either reasonably sensitive to acoustic noise that covers speech, or to certain speech sounds that are poorly transmitted, especially the high-frequency consonant sounds that are necessary for speech understanding.

For people who are exposed to high noise levels, the health and safety regulations require that person wear hearing protection. Hearing protection takes the form of either sealing bells that surround the ear, or as earplugs that block the ear canal. The latter type of hearing protection is often preferred due to its limited size and relatively good comfort.

It is an additional object of this invention to provide an earplug with two desired properties: The cavity closed by the earplugs in the inner part of the meatus,

is relatively free of external noise, which is the purpose

with earplugs to protect your hearing.

The sound field in the cavity, generated by the persons own voice, contains

all the frequency components that are necessary to reconstruct the speech with good speech understanding.

The solution according to the invention takes advantage of these facts. By using a microphone to capture the acoustic sound field in the inner part of the meatus and processing the microphone signals according to the invention, a speech signal with high quality and low noise masking is produced.

It is a further purpose of the present invention to provide a system which enhances the user's own experience of naturalness in the user's own voice when using a hearing protected communication terminal according to the invention is used.

By using regular earplugs or earmuffs, the user will normally experience their own voice as distorted, a feature that reduces the comfort of wearing hearing protection. Conventional hearing protection alters the normal path of sound transmission from the mouth to the eardrums. The auditory feedback from the user's own voice is affected as a result of the unintended change in speech output. A normal reaction is to raise one's own voice level when using a headset or earplugs.

The invention solves these problems by filtering and mixing in the user's own voice which is picked up either by the external or internal microphone in one ear and reproducing the signal at the loudspeaker in the other ear. It is also possible to reproduce the signal at the loudspeaker in the same ear, in which case feedback cancellation must be used. The user's own voice is therefore experienced more naturally both with regard to frequency response and speech level. This move will increase the extent of acceptance for continuous use of hearing protection throughout the working day. The own speech signal is added and reproduced in such a way that the noise-reducing properties of the hearing protection are maintained.

An additional object of this invention is to provide a programmable, personal noise exposure dosimeter that measures the real exposure in the user's ear and calculates the risk of hearing damage.

The usual dosimeters for noise exposure, which are also referred to as dosimeters, usually consist of a microphone and a small electronic device that can be attached to the body or can be carried in the pocket. The microphone can be attached to the shirt collar or on the shoulder. ANSI Sl.25 specifies dosimeters.

Today's dosimeters have several disadvantages:

The dosimeters do not measure the noise that actually affects the hearing organ

(such as when the user wears hearing protection, a helmet, etc.). Even when the ear is not covered, the measurements can be influenced by body shielding.

The dosimeters are sensitive to accidental or intentional error,

that can affect readings, such as the wearer's light tapping or singing into the dosimeter's microphones

or of noise generated by wind.

The dosimeter is inaccurate if noise, caused by impulses or blows hitting the dosimeter.

The invention solves these problems by using a microphone which measures the sound at the eardrums and which uses procedures for analysis which take both stationary and sudden sound into account. When the dosimeter is part of a communication terminal, this includes external noise, incoming communication signals, as well as equipment malfunction.

It is also an object of the invention to provide a device for verifying on the spot that a hearing protector functions satisfactorily.

Today's hearing protection takes the form of either sealing caps that surround the ear, or earplugs that block the ear canal. For both types, it is critically important to avoid leakage of noise through or around the sealing or blocking parts of the hearing protection.

Experience shows that several factors can affect the sealing of the hearing protection and thereby increase the risk of hearing damage. These factors include: Irregular surfaces that the sealing material does not adequately cope with

to obey. Examples are glasses used together with earmuffs, and earplugs used by people with

irregularly shaped ear canals.

Incorrect positioning of the hearing protection. Experience and patience are required

the user to get the hearing protection fitted correctly. In cases where the user wears a helmet or a headgear, the hearing protection during use may be inadvertently pushed out of

position.

Aging of the material in the seal can reduce the material's ability to yield

after in the seal and thereby enable leakage around the seal.

A result of the leak is reduced attenuation of potentially dangerous noise. Ideally, the leak should be detected and rectified before the noise exposure. The leak will often not be clearly audible. Consequently, the noise situation could consist of periodic or impulsive components that could damage the hearing almost immediately if a hearing protector were to malfunction or be deficient without the user being aware of this. The invention solves these problems by an on-site acoustic measurement, which is analyzed and reported to the user in an audible form, or to an external device using communication signals. The device required for the measurement is an integral part of the hearing protection. The verification can be activated by the user at any time, or can work continuously when the use is critical.

Alternatively, the verification can be activated by a person (or device) other than the user, for example to verify the functionality of the hearing protection before access to a noise-laden area is permitted.

The above-mentioned problems are solved by the invention according to the accompanying claims.

The invention will be described below with reference to accompanying drawings, which drawings illustrate the invention in the form of examples. Fig. 1 is a simplified vertical section along the central axis of the meatus in the outer ear of an upright human, with an inserted ear terminal according to an embodiment of the invention, also shown in a vertical section along the axis which coincides with the axis of the meatus. Fig. 2 is an electrical connection diagram showing the functional components and the connections between electronic components in a preferred embodiment according to the invention. Fig. 3 is an illustration of a method according to the invention, which shows that spectral analysis of sound captured in the ear is compared to spectral analysis of sound captured by a microphone at a standard distance, for example 1 meter, under otherwise calm conditions. Fig.4 is an illustration of speech sound analysis and subsequent sound source classification with filtering carried out by sound source classification, according to an embodiment of the invention. Fig. 5 is an illustration of another method according to the invention, which illustrates an analysis where sound captured close to the ear is compared with an analysis of sound captured by a microphone placed in the meatus. Fig. 6 illustrates a simplified section through a human's right and left ear with ear terminals according to the invention, illustrated for improved natural sound purposes. Fig. 7 illustrates a process diagram for an embodiment of the invention concerning noise dose measurement, here illustrated by an A-weighting with accumulated noise dose measurements, and also with C-weighting of recordings of peak values of noise. Fig.8 illustrates a second embodiment of the invention, which illustrates a scheme for processing for online verification of the effect of the hearing protection. Fig.9 illustrates an electrical analogy diagram for an acoustic phenomenon, on which an embodiment of online verification of the effectiveness of hearing protection is based.

Description of preferred embodiments of the invention.

The physical design of an embodiment of the present invention enables the construction of a complete in-ear hearing protector and communication terminal with strong passive sound attenuation, strong active sound attenuation, high-quality sound reproduction, high-quality sound capture, small size, low weight and comfortable fit.

An embodiment of this invention is illustrated in Figure 1 and provides the general physical design of a fully in-ear hearing protection and communication terminal, considered as a combination of passive sealing characteristics and placement of electro-acoustic transducers, as well as acoustic filters, electrical circuits and a ventilation system for equalizing pressure.

The ear terminal comprises an outer part 1, arranged to sit on the outward facing end of the sealing part 2 and a part on the inward facing part of the outer part 1 is shaped to fit the outer ear around the outer part of the meatus 3.

The physical design, represented by an embodiment of the invention, enables some or all of the following functions: External sounds are dampened by passive and active noise control. The passive

the attenuation is achieved by means of an earplug 1, 2 with a sealing system 2 inserted in the outer part of the ear canal or meatus 3. The active noise control is provided by one or two microphones Ml, M2 and a loudspeaker SG, together with electrical circuits in an electronics unit 11 which is mounted in the earplug system. The algorithms for noise control are known per se and will not be described in detail here, but may include active noise cancellation by means of returning acoustic signals, converted by at least one of said microphones (Ml, M2) through

the sound generator (SC).

The restoration of the desired sound (external sounds and signals from

the communication system) at the eardrum or tympanum 4 is provided by using the same microphones Ml,

M2, and the loudspeaker SG and the electronics unit 11. Here, too, the algorithms for achieving this are known per se and will not be described in detail here, but may involve amplification of selected frequencies, converted by said microphone (Ml) and generation of a corresponding acoustic signal through said sound generator (SG). The frequencies can, for example, be within the normal range for the human voice. The ear terminal according to claim 1 comprises a sound generator (SG) arranged to be directed towards the meatus and connected to said electronics unit (11), where the electronics unit (11) comprises filtering devices for transmitting active sound, for example by means of amplification of selected frequencies converted by using said external microphone (Ml) and by generating a corresponding acoustic

signal through said sound generator (SG).

Reception of the user's voice is carried out using a microphone M2

with access to the closed area in the meatus 3. This signal is processed using analogue or digital electronics in the electronics unit 11 to make it particularly natural and understandable, either for the user himself or his communicating partners or both parties. This signal has high quality and is well suited for

voice control and speech recognition.

Online control and verification of the function of the hearing protection is achieved by

bring in an acoustic measurement signal, preferably by means of a sound generator or speaker SG in the meatus, and to analyze the signal received by the microphone M2 which has

access to the acoustic signal in the meatus 3.

Measurements of the dose of noise exposure at the tympanic membrane 4 and online

calculation carried out by the electronic circuits, and notification of the risk of hearing damage either by audible or other warning signals, either to the wearer of the hearing protection or

to other relevant personnel.

Equalization of pressure between the two sides of the earplug system is achieved by

use a very thin channel T3, T4 or a valve that balances the static pressure difference, while at the same time maintaining a strong damping of low-frequency sound. A safety valve V, which takes care of rapid decompression, can be incorporated into the pressure equalization system T3, T4.

Figure 1 illustrates an embodiment according to the invention. The earplug comprises a main part 1 which contains two microphones Ml and M2 and a sound generator SG. The head part is designed in such a way as to provide comfortable and secure placement in the concha (the bowl-shaped cavity at the entrance to the ear canal). This can be achieved by using individually molded ear pieces that are held in position by the outer ear or by using a flexible surface material that presses against the tissue in the outer ear. A sealing device 2 is attached to the main part. The sealing device may form an integral part of the earplug, or it may be replaceable. The sound input of the microphone Ml is connected to the outer end of the earplug, in order to pick up the external sounds. The microphone M2 is connected to the inner part of the meatus 3 by means of an acoustic transmission channel Tl. The acoustic transmission channel can optionally contain additional elements for acoustic filtering. The outlet SSG on the sound generator SG is open to the inward-facing part of the sealing part 2. The acoustic transmission channel T2 can optionally contain additional elements for acoustic filtering.

When smaller microphones M2 and sound generators SG are available, it will be possible to mount the microphone M2 and the sound generator SG at the innermost part of the sealing part. Then there is no need for the transmission channels Tl and T2.

The two microphones and the sound generator are connected to an electrical unit 11, which can be connected to other equipment by means of a connecting part 13 which can transmit digital or analogue signals, or both, and as an option, power.

Electronics and a power supply 12, for example a battery, can be incorporated in the part 1 or in a separate part.

In a preferred embodiment, the microphones Ml, M2 can be made up of standard miniature electret microphones similar to those used for hearing aids. Later developed silicone microphones can also be used.

In a preferred embodiment, the sound generator SG can be based on electromagnetic or electrodynamic principles, such as sound generators used in hearing aids.

According to a preferred embodiment of the invention, a safety valve V is incorporated in the ventilation duct, which comprises the ducts T3 and T4. The valve V is arranged to open if the static pressure in the inner part of the meatus3 exceeds the external pressure by a predetermined value, which allows pressure equalization in case of rapid pressure drops. Such pressure drops can occur for military or civilian air personnel who experience a rapid loss of external air pressure. Such pressure drops can also happen to skydivers, divers, or the like. Pressure equalization for slowly varying pressure changes is achieved by using a thin ventilation channel T4 that bypasses the valve V. A proper design of this ventilation channel T4 enables static pressure equalization without having to compromise on the ability to dampen low-frequency noise.

The main part of the earmold can be made of standard polymer materials used in normal hearing aids. The sealing part may be made of an elastic, slowly re-expanding, shape-retaining polymer foam, such as PVC, PUR or other materials suitable for earplugs.

In some applications (less extreme noise levels), the earplug can be molded in one piece 1, 2, thereby combining the main part 1 and the sealing part 2. The material for this design can be a typical material used for passive earplugs (Elacin, acrylic).

It is also possible to make the earplug in one piece, comprising the main part 1 and the sealing part 2, where everything is made of a polymer foam as indicated above, but where the channels Tl, T2, T3, T4 are made of a wall material that prevents the channels T1,T2 , T3,T4 in collapsing when the sealing part 2 is introduced into the meatus 3 .

All the features described above can be achieved by an electrical circuit which is represented by the block diagram in figure 2.

The microphone Ml picks up the external sound. A signal from the microphone Ml is amplified in El and sampled and digitized in an analog to digital converter E2 and fed to a processing unit E3 which can be a digital signal processor (DSP), a microprocessor (/iP) or a combination of both. A signal 51 from the microphone M2, which picks up the sound in the meatus between the insulating part 2 and the tympanum 4, is amplified in the amplifier E4 and sampled and digitized in the analog to digital converter E5 and fed to the processing unit E3.

A desired digital signal DS is generated in the processing unit E3. This signal is converted to analog form in the analog to digital converter E7 and fed to the analog output amplifier E6 which drives the loudspeaker SG. The sound signal produced by the loudspeaker SG is fed to the eardrum (tympanum) 4 via the channel T2 and into the meatus as described above.

The processing unit E3 is connected to memory elements RAM (random access memory) E8, ROM (read only memory) E9 and EEPROM (electrically erasable programmable read only memory) ElO. In a preferred embodiment of the invention, the memories E8, E9 and E10 are used to store computer programs, filter coefficients, analysis data and other relevant data.

The electronic circuits 11 can be connected to other electronic devices through a two-way digital interface E12. Communication with other electrical devices can be carried out via a cable or wirelessly through a digital radio link. The Bluetooth standard for digital short-range radio (Specification of the Bluetooth System, version 1.0 B, 01 December 1999, Telefonaktiebolaget LM Ericsson) is a possible candidate for wireless communication for this digital interface E12.

In a preferred embodiment of the invention, the signals transmitted through this interface can be: program code for the processing unit E3

analysis data from the processing unit E3

synchronization data when two ear terminals 1, 2 are used in a binaural (two-ear) mode

digitized audio signals in both directions to and from an ear terminal 1, 2

control signals for controlling the operation of the ear terminal

digital measurement signals for diagnosis of the performance of the ear terminal.

A manual control signal can be generated in Ell and fed to the processing unit E3. The control signal can be generated by means of operating knobs, switches, etc., and can be used to turn the device on or off, to change the operating mode, etc. In an alternative embodiment, a predetermined voice signal can constitute the control signal of the processing unit E3.

The electric current circuit is powered by a power supply 12a which can be constituted by a primary battery or a rechargeable battery arranged in the earplug or in a separate unit, or it can be powered via a connection to other equipment, such as for example a communication radio.

An embodiment of the invention relates to the use of an ear terminal as an "in-the-ear voice receiver". The sound of a person's own voice as heard in the meatus is not identical to the sound of the same person's voice as heard by another listener. The present embodiment of the invention solves this problem. The microphone M2 illustrated in Figure 3 picks up the sound in the inner part of the meatus 3 which is closed off by means of the sealing unit 2 in an ear protection communication device of the earplug type. The signal is amplified by the amplifier E4 illustrated in figure 2, A/D converted by the A/D converter E5 and processed in the digital signal processing unit (DSP) or in the microcomputer unit E3. The processing can be seen as a signal-dependent filter that takes into account the signal properties as well as calculated estimates of the location of the sound generator for different speech sounds. Thereby, both speech understanding and naturalness can be improved.

Figures 1 and 3 show examples of embodiments of the invention, with the microphone M" as an integral part of a hearing protective communication earplug. The acoustic transmission channel Tl connects the microphone M2 to the inner part of the meatus 3. The microphone M2 picks up the sound field created by a person's own voice. The signal can be amplified in the amplifier E4, A/D converted in the A/D converter E5 and processed in the digital signal processing unit (DSP) or in the microcomputer unit E3 A processed signal from E3 can be transmitted in digital form through a digital interface E12 to other electronic devices In an alternative embodiment, the processed signal from E3 can be D/A converted and transmitted in analog form to other electronic devices.

Figure 4 illustrates a possible arrangement for signal processing according to the invention. The figure illustrates an example of the type of signal-dependent filters that can be used on the signal from a microphone M2 to achieve a good reconstruction of the speech signal, which provides very good speech understanding, even in extremely noisy environments.

After the amplification in E4 and the A/D conversion E5, the signal of the microphone M2 is analyzed in the DSP//zP processing unit E3. The analysis represented by block 21 in Figure 4 may comprise a short-term estimate of the spectral power in the microphone signal, an auto-correlation estimate of the microphone signal, or a combination of both. Based on these estimates, a continuous classification with corresponding determination, represented by block 22, can be made in the processing unit E3 for selection of the most suitable conditioning filter for the signal from the microphone M2. In the example shown in Figure 4, the selection can be made between, for example, three filters Hl(f), H2(f) and H3(f), represented by blocks 23, 24 and 25, suitable for vowel sounds, nasal sounds and fricative sounds respectively. The processed signal is present at the output 26 and the block 22. Other sound classifications that use more sophisticated sub-divisions between sound classifications and corresponding sound filters and analysis algorithms can be used. The selection algorithm can include a gradual transition between the filter outputs to thereby avoid audible artefacts. Filtering and selection is carried out in the processing unit E3 at the same time as the sound analysis and classification.

The basis for the filter characteristics and associated analysis and classification processing unit E3 can be derived from an experiment with the form shown in figure 3. An earplug with microphone M2, which has the same characteristics as the one used for voice capture, is used to capture the voice of a test subject from meatus 3, illustrated in the upper part of figure 3. At the same time, the voice is recorded by a high quality

microphone M3 in front of the test subject, at a normal distance of 1 meter, under anechoic conditions. Estimation of power spectrum

densities can be calculated for the two signals using the analyzes represented by boxes 27 and 28, and the corresponding levels LI(f) and L2(f) are compared in the comparison unit 29. The output of the comparison unit is represented by the transfer function H(f). The analyzes can be short-term spectral estimates, for example 1/9 octave spectra in the frequency range 10 0 Hz to 14 0 0 Hz. The test sequences that the test subject utters can include speech sounds that are kept constant for around 1 second. For voiced sounds, the subject person can add the pitch

vary during the analysis period. The transfer function of the filters described in connection with figure 4 can be based on diagrams of H(f), the spectral density level of the free-field microphone M3, subtracted from the corresponding level of the in-ear microphone M2.

A simplest embodiment of the invention can reduce the system in Figure 4 to a simple time-invariant filter. The analysis and selection processing can then be omitted. The transfer function of the single filter will still be based on the diagrams of the spectral density level of a free-field microphone subtracted from the corresponding levels of the in-ear microphone, described in connection with Figure 3. The transfer function may be a combination of the results for the different voice sounds, weighted in accordance with their importance for audibility and naturalness for the processed speech.

Another embodiment of the invention is best understood under the term "Natural Own Voice", which indicates that a person wearing an ear terminal should experience his own voice as natural while the person has a meatus blocked by an earplug.

The inner microphone M2, or the outer microphone Ml, or a combination of both, captures the audio signal representing the user's voice signal. The signal is amplified, A/D converted and analyzed in the digital signal processor E3. Based on previously measured transfer functions from the user's speech to the microphone M2 (and/or M1), the microphone signal can be filtered to restore the naturalness of the user's speech. The signal is then D/A converted, amplified and reproduced by an internal loudspeaker SG. The internal loudspeaker can be arranged in a similar ear terminal 1, 2 in the wearer's other ear to prevent local feedback in the earplug. In a more acoustically demanding arrangement, the loudspeaker SG, which is arranged in the same meatus as the one where the internal pick-up microphone M2 is located, can be used, and in this way require feedback cancellation. The desired signal to the loudspeaker SG in the other ear can be transmitted via electrical conductors on the outside of the wearer's head, or via radio signals. Figure 6 shows a preferred embodiment of the invention where the natural voice characteristics are integrated into two active hearing protective communicating earplugs. Each earplug may comprise a main part 1 containing two microphones, an outer microphone M1 and an inner microphone M2, and a sound generator SG. The right and left earplugs are each generally symmetrical, and otherwise identical for each ear. Part 2 is an acoustic seal on the hearing protection. An acoustic transmission channel Tl connects the microphone M2 to the inner part of the meatus 3. The microphone M2 picks up the sound from the meatus 3. When the user speaks and the ear canal is blocked, this signal will essentially be the user's own voice signal. This signal is filtered and reproduced in the loudspeaker SG in the other ear. An acoustic transmission channel T2 connects the sound generator SG with the inner part of the meatus 3. A block diagram of the electrical system is shown in Figure 2. Figure 4 shows an example of the type of signal-dependent filters that can be used for the microphone signal to achieve a good reconstruction of the voice.

After amplification in E4 and the A/D conversion E5, the signal from the microphone M2 is analyzed in the DSP//iP processing unit E3. The analysis represented by block 21 in Figure 4 may comprise a short-term estimate of the spectral power in the microphone signal, a short-term auto-correlation estimate of the microphone signal, or a combination of both. Based on these estimates, a continuous classification with associated determination represented by block 22 can be made in the processing unit E3 for selection of the most suitable filter for the signal from the microphone M2. In the example shown in Figure 4, the selection is made between, for example, three filters Hl (f), H2(f) and H3(f), represented by blocks 23, 24 and 25, suitable for vowel sounds, nasal sounds and fricative sounds respectively. The processed signal is present at the output port 26 of the block 22. Other sound classifications that use more sophisticated subdivisions between sound classifications and associated sound filters and analysis algorithms can be used. The selection algorithm can include gradual transition between filter output ports to thereby avoid audible artifacts. Filtering and selection is carried out in the processing unit E3 at the same time as the sound analysis and classification.

The basis for the filter characteristics and the associated analysis and classification in the processing unit E3 can be derived from an experiment in a form as shown in figure 5. An earplug with a microphone M2 with generally the same characteristics as that used for capturing voices can be used to capture pick up the voice of a test subject from the meatus 3 illustrated in the upper part of figure 5. At the same time, the voice is recorded by a high-quality microphone M4 close to the subject's ear, under anechoic conditions. Estimates of power spectral densities can be calculated for the two signals using the analysis represented by blocks 37 and 38 respectively. Corresponding levels LI(f) and L2(f) are compared in the comparator 39. The output data from the comparator is represented by the transfer function H(f). The analyzes can be short-term spectral estimates, for example 1/9 octave spectra in the frequency range 100 Hz to 14000 Hz. The test sequences expressed by the test subject may include speech sounds, which are held for approximately 1 second. For voiced sounds, the test subject can vary the pitch during the analysis period. The transfer function of the filters described in connection with figure 4 can be based on digrams of H(f), the levels of the spectral density of the free-field microphone M4 subtracted from the corresponding levels of the in-ear microphone M2.

A simplest embodiment of the invention can reduce the system in Figure 4 to a simple time-invariant filter. The analysis and selection processing can then be omitted. The transfer function of the single filter will still be based on the diagrams of the spectral density level of a free-field microphone, subtracted from the corresponding levels of the in-ear microphone, described in connection with Figure 5. The transfer function may be a combination of the results for the different voice sounds, weighted according to their importance to the accuracy of the processed speech.

Another embodiment of the invention is a so-called "personal noise exposure dosimeter". Similar to the embodiments described above, a microphone M2 captures the sound in the meatus 3. One of the new features is that this noise exposure is measured in the meatus, even when the ear is already protected from noise. The signal from the microphone M2 is amplified, A/D converted and analyzed in a digital signal processing (DSP) unit or a microcomputer unit E3 in the same way as described above. According to a preferred embodiment of the invention, the analysis covers both stationary and semi-stationary noise, and sudden noise. The result from the analysis is compared with the criteria for damage risk and the user receives an audible warning signal or other forms of warning signals when certain limits are about to be exceeded and measures must be initiated. The warning signal can also be transmitted to other devices, such as systems for monitoring industrial health protection. According to a preferred embodiment, time recording of the analysis can be stored in a memory, for example in a RAM E8 for later reading and processing.

Figure 1 shows a preferred embodiment of the invention with a dosimeter for personal noise exposure, integrated in an active hearing protection communication earplug, comprising a main part 1 containing two microphones, an outer microphone M1 and an inner microphone M2 and a sound generator SG. Since this embodiment forms part of a hearing protection earplug, a sealing device 2 is attached to the main element. An acoustic transmission channel Tl connects the microphone M2 to the inner part of the meatus 3. The microphone M2 therefore picks up the sound present in the meatus 3, just outside the eardrum (tympanum) 4. An acoustic transmission channel T2 connects the sound generator SG to the inner part by meatus 3. The sound generator SG can provide audible information to the user, in the form of warning signals or synthetic speech.

All the electronics, as well as the battery, are arranged in the main part 1.

A block diagram of a possible implementation of this embodiment is shown in figure 2. The sound is captured by the microphone M2, amplified and AD-converted before being fed to the processing unit E3 with DSP or/xP (or both) as the central processing unit. The memory units E8 with RAM, E9 with ROM and E10 with EEPROM can store programs, configuration data and analysis results. The information for the user is generated in the central processing unit E3, DA-converted, amplified and can be presented as audible information via the loudspeaker SG. The digital interface is used for programming, management and reading out the results.

The signal processing for calculating noise exposure is shown in the flow diagram in figure 7. The signal from the microphone M2 is amplified, converted to digital form and analyzed by the algorithms in the processing unit E3. Sample-by-sample equalization, represented by block 41, is first used to compensate for irregularities in the response area of the microphone, the transmission channel Tl and the missing ear canal response caused by blockage of the channel due to the earplug. According to the invention, the processed samples can be evaluated in at least two ways. To obtain the stationary or semi-stationary noise doses, an A-weighting represented by block 42 is used. Standards for this A-weighting exist: IEC 179 and the sample values are squared and accumulated in blocks 43 and 44 respectively. To obtain the peak value for to estimate suddenly occurring noise, a C-weighting represented by block 45 is used according to internationally accepted standards, also IEC 179, and peak valuers (regardless of sign). The noise dose and peak values are finally compared against predetermined limits in a decision algorithm represented by block 47, so that warning signals can be given. The audible information to the user can be provided in the form of warning signals or synthetic speech. The warning signal can also be transmitted to other devices, such as, for example, devices for monitoring industrial health protection. The time recording of the two can also be stored in the memory of the processing unit E3 for later reading and further evaluation.

In addition to use as a device for passive hearing protection, this embodiment of the invention can be used as hearing protection when the terminal is used as a headphone connected to CD players or the like, to control the noise dose sent from the headphones to the ear over time, or as a peak value.

Another embodiment of the invention, called "online verification/control of hearing protection function", utilizes the fact that the sound field that is locally generated in the cavity near the eardrum is influenced by leakage in the hearing protection. A small electro-acoustic transducer (sound source) SG and a microphone M2 are placed in a sealing part 2, arranged to dampen the sounds entering the meatus cavity 3. A digital signal processing unit (DSP) or a micromachine unit E3 in the main part 1 or in the sealing part 2, is used to generate a predetermined signal which is D/A converted by the D/A converter E7, amplified by the amplifier E6 and supplied to the sound source SG, which generates a sound field in the closed part of the meatus 3. The microphone M2 captures up the sound in the meatus cavity 3. This signal is amplified by the amplifier E4, A/D converted by the A/D converter E6, and analyzed in the digital signal processing unit or microprocessor E3. The result of the analysis is compared with stored results from previous measurements of the same type for the same situations with good tetnin conditions. The user can receive audible confirmations or other reported confirmations that the leakage is acceptably low, or a warning signal that the leakage is unacceptably high. In the same way, a signal can be sent to other units, such as, for example, a management unit for external industrial health, with information about the leak. An example could be that an ear terminal according to the invention is used to check for leakage in the hearing protection but the wearer is still in a gate that controls access to a noise-prone area. If a leak occurs, a signal will be transmitted from an ear terminal to an associated signal receiver at the gate, where equipment is arranged to block access until the leakage conditions are addressed and verified. Figure 1 illustrates an embodiment of the invention where the device for verification is integrated in a hearing-protective earplug. This embodiment comprises an outer part 1 containing a microphone M2 and a sound generator SG. An inner sealing part 2 is attached to the outer part, but can be designed as an integral outer part/sealing part 1, 2. An acoustic transmission channel T2 connects the sound generator SG with the inner part of the meatus 3. The sound generator SG produces a predetermined signal, which generates a sound field in the meatus 3. An acoustic transmission channel Tl connects the microphone M2 to the inner part of the meatus 3. The microphone M2 captures the sound field set up by the sound generator SG. The signal generation and analysis is carried out in a digital signal processing unit (DSP) or a microcomputer unit E3 with suitable amplifiers and converters as described in the previous section. All electronics 11, as well as power supply 12, are provided in the outer part 1. Figure 8 illustrates an arrangement for signal processing according to a preferred embodiment of the invention. This embodiment uses a signal that produces reliable characterizations of sound fields in the cavity, preferably without causing annoyance to the user. The signal may comprise one or more sinusoidal components which are presented simultaneously or in sequence. Alternatively, a pseudo-random sequence can be used. In both cases, both in-phase and out-of-phase conditions in the sound field are preferably analyzed and used in the verification algorithm.

An example of signal processing is shown in the flow diagram in Figure 8. In this example, two pure tones with different frequencies fx and f2 are generated by algorithms in the processing unit E3. The generators are represented by blocks 81 and 82 respectively. The generators generate in-phase (sin) and out-of-phase (cos) components. The I-phase components are summed in block 83, converted to analog form, amplified and applied to the sound generator SG. The resulting sound field is captured by the microphone M2, amplified, converted to digital form and analyzed by algorithms in the processing unit E3 for a series of detectors represented by blocks 84, 85, 86 and 87. In-phase and out-of-phase the components for the microphone's M2 signal are analyzed for each of the two frequencies. The detector algorithms perform a sample-on-sample multiplication of the two input signals and equalize the result with a low-pass filter. The output signal of the four detectors is used in a polling algorithm represented by block 88, where they are compared with stored values. The decision result can be a digital "go" / "no go" real-time signal indicating acceptable noise protection attenuation or unacceptable protection conditions. The results of the analysis are compared with stored results from previous measurements of the same type in a situation with good sealing conditions.

Stored values for the decision algorithm can, according to a preferred embodiment, be based on previous laboratory experiments. However, the values for the decision algorithm can also be determined, for example, by making an average and by determining a lower acceptance limit for a general-purpose embodiment of the invention.

The number and values of frequencies and equalization characteristics for the detector can be weighted as a compromise between audibility and reaction time. Should a continuous ongoing verification be necessary, low frequencies with sufficiently low levels, such as for example in the range 10 - 20 Hz, can be used to avoid irritation. The pure tones can then be partially or completely audibly masked by the residual noise transmitted by the hearing protection.

The acoustic phenomenon on which the embodiment of the invention is based is illustrated by the electrical analog diagram shown in Figure 9. In the diagram, the sound generator SG is modeled by its acoustic Thevenin equivalent, represented by blocks 91 and 92. The sound pressure pl is generated by the Thevenin generator 91, resulting in a volume velocity through the Thevenin impedance Zl(f) 92. The microphone M2 is modeled by its acoustic impedance Z3(f), represented by block 93. The sound pressure p2 at the microphone's input is converted to a electrical signal of the microphone. For the solution in the present illustration, all acoustic elements exposed to sound pressure generated by the sound generator SG, apart from the microphone, are lumped together in an acoustic impedance Z2(f) represented by block 95. A leak in the hearing protector can be modeled by changing the variable acoustic the impedance Z2(f). The change will usually affect both the frequency-dependent modulus and a frequency-dependent phase of Z2(f). This change leads to a change in the ratio between the sound pressure p2 and pl, which is analyzed as described in connection with Figure 8.

Claims (15)

1. Device for ear protection for placement in the meatus of the ear (3) and comprising an internal microphone (M2), where the internal microphone (M2) has an audio input (S2) directed towards the meatus (3) for converting a received audio signal into an electrical signal, characterized in that the device also comprises an electronics unit (11, E3) containing sound analysis means designed to analyze the internal microphone's input signal and compare the signal with predetermined values for the signal corresponding to acceptable noise levels, and for activating an indicator (72 ) in the event of a detected exceedance of these levels. 2. Device according to claim 1 where the sound analyzer (5) is provided with a discriminator (55) to distinguish between periods with a sound level above or below given limits (71) for sound level. 3. Device according to claim 1, where the indicator (72) comprises a time recording (73) for the sound levels. 4. Device according to claim 1, where the indicator (72) comprises a registration (75) of undertaken risk. 5. Device according to claim 4, where the indicator (72) comprises a registration of accumulated noise dose. 6. Device according to claim 1, comprising a sound generator connected to the indicator with a sound outlet directed towards the meatus. 7. Device according to claim 1, where the indicator comprises a radio transmitter. 8. Device according to claim 1, comprising a sealing section which defines the inner surface of the device against the meatus, which sealing section comprises a channel (Tl) between the sound inlet (S2) and the internal microphone (M2). 9. Device according to claim 1, comprising an electronic memory for recording the analyzed signal. 10. Device according to claim 1, comprising an external microphone (Ml) connected to the electronics unit for acoustic measurements in the surroundings. 11. Device according to claim 1, comprising a pressure equalization channel (T3) for slow pressure passage to and from the meatus (3) through the sealing section (2). 12. Device according to claim 11, where the pressure equalization channel (T3) comprises a pressure-releasable valve (V) arranged to open if the pressure difference between the meatus and the surroundings exceeds a given limit. 13. Device according to claim 12 comprising a leakage channel (T3) in the pressure equalization channel past the valve (V). 14. Device according to claim 6, comprising a connecting part (13, E12), for example a radio transmitter or electrical connector, connected to the electronic unit (11) where the electronic unit is provided with conversion means (E7) for converting signals received from the connecting part and connected to the sound generator (SG) for transmitting information to the user. 15. Device according to claim 1, comprising a connecting part (13, E12), for example a radio transmitter or electrical connector, connected to the electronics unit (11) where the electronics unit is provided with conversion means (E5) for converting signals received from the internal microphone ( M2) electrical or electromagnetic information from the user.