WO2006117718A1

WO2006117718A1 - Sound detection device and method of detecting sound

Info

Publication number: WO2006117718A1
Application number: PCT/IB2006/051281
Authority: WO
Inventors: Christophe Macours
Original assignee: Koninklijke Philips Electronics N.V.
Priority date: 2005-05-03
Filing date: 2006-04-24
Publication date: 2006-11-09

Abstract

A device (100) for processing audio data (106, 107), wherein the device (100) comprises a first microphone (103) adapted to detect first audio data (106) emitted by an acoustic source (101), a second microphone (104) adapted to detect second audio data (106) emitted by the acoustic source (101), and a processing unit (108) adapted to combine the first audio data (106) with the second audio data (107) to thereby reduce acoustic noise, wherein the first microphone (103) and the second microphone (104) are positioned essentially symmetrically with respect to the acoustic source (101).

Description

A device for and a method of processing audio data

FIELD OF THE INVENTION

The invention relates to a device for processing audio data. Beyond this, the invention relates to a method of processing audio data. Moreover, the invention relates to a program element. Furthermore, the invention relates to a computer-readable medium.

BACKGROUND OF THE INVENTION

Speech communication nowadays occurs more and more under noisy circumstances as well as during motion intensive activities. Moreover, a user usually wears earphones while on the move or to enjoy her or his favourite music. Some users also use noise suppressing earphones or headphones in order to reduce ambient noise.

It may be problematic in speech communication in a noisy environment to be able to capture the user's speech with sufficiently little background noise.

One possibility is to bring the microphone as close as possible to a user's mouth, for instance via a hanging boom or a stick. These solutions are usually not practical as they may impose usage and fitting constraints to the user. For instance, the hanging boom should always be outside the jacket. Moreover, such solutions usually have a rather high form factor and are not as that aesthetic and practical in use.

Another concept is related to directive recording in the direction of the mouth in order to reject as much as possible of signals that do not come from the mouth, that is to say the ambient noise. This can be done by using a directional microphone or by performing beam forming.

The described solution may be efficient in case that a bone conduction-based system is acceptable that provides noise suppression at the expense of the speech quality.

Furthermore, small form factor noise reduction headsets are known, like Sound ID or Bluetooth headsets, using two microphones placed at the same side of the head of a user and a dual-microphone beam form algorithm to capture the user's voice. A constraint to get a reasonable performance is that one microphone picks up most the speech (so-called primary microphone) and the other microphone picks up mostly noise (so-called secondary microphone). This imposes a physical distance and an acoustical barrier between the two microphones. Consequently, this solution has usually a non-negligible form factor and is not ideal for being permanently used, particularly while on the move.

US 2004/0047464 Al discloses a microphone system using two microphones arranged at a predefined distance from each other. A first and a second input signal resulting from sounds received by the two microphones are received, and the first input signal is enhanced to generate a speech-enhanced signal. Furthermore, the acoustical signal component in the second input signal is suppressed to generate a speech-nulled signal. A noise cancellation filtering circuit is adapted to receive the speech enhanced signal and the speech nulled signal, wherein the noise in the speech enhanced signal is cancelled using the speech nulled signal as reference, thereby generating an output filtered signal representing the desired signal.

US 6,717,991 Bl discloses a dual microphone noise reduction system implementing a far-mouth microphone in conjunction with a near-mouth microphone.

OBJECT AND SUMMARY OF THE INVENTION

It is an object of the invention to provide a system for processing audio data that is capable of providing sound with proper quality.

In order to achieve the object defined above, a device for processing audio data, a method of processing audio data, a program element and a computer-readable medium according to the independent claims are provided.

According to an exemplary embodiment of the invention, a device for processing audio data is provided, wherein the device comprises a first microphone adapted to detect first audio data emitted by an acoustic source, a second microphone adapted to detect second audio data emitted by the acoustic source, and a processing unit adapted to combine the first audio data with the second audio data to thereby reduce acoustic noise. The first microphone and the second microphone may be positioned symmetrically with respect to the acoustic source.

Furthermore, according to another exemplary embodiment of the invention, a method of processing audio data is provided comprising the steps of detecting, by means of a first microphone, first audio data emitted by an acoustic source, and detecting, by means of a second microphone, second audio data emitted by the acoustic source. The first microphone and the second microphone may be positioned symmetrically with respect to the acoustic source. Furthermore, the first audio data may be combined with the second audio data to thereby reduce acoustic noise.

Beyond this, according to another exemplary embodiment of the invention, a computer-readable medium is provided, in which a computer program of processing audio data is stored, which computer program, when being executed by a processor, may be adapted to control or carry out the above-mentioned method steps.

Moreover, a program element of processing audio data is provided according to still another exemplary embodiment of the invention, which program element, when being executed by a processor, is adapted to control or carry out the above-mentioned method steps.

Processing audio data according to the invention can be realized by a computer program, that is to say by software, or by using one or more special electronic optimization circuits, that is to say in hardware, or in hybrid form, that is to say by means of software components and hardware components. The characterizing features of the invention may particularly have the advantage that two microphones may be arranged in a symmetrical manner with respect to an acoustic source (that is to say essentially at the same distance from the acoustic source), for instance a mouth of a human user carrying a headset comprising the two microphones. By such a symmetrical location of the two microphones, it may be possible to capture audio signals received by the two microphones, wherein the geometric arrangement of the two microphones may allow for an effective cancellation of noise, by simultaneously processing these two audio signals. In other words, a noise-reduced audio output signal may be generated by combinedly evaluating the audio signals captured at symmetrical positions with respect to the signal source. In other words, both microphones may be located at different/opposite sides of the signal source. By taking this measure according to the invention a simple but efficient means of improving audio quality of an audio signal receiving system may be provided.

According to an aspect of the invention, two audio signals both including speech and noise may be captured. By sophisticatedly analyzing these two signals simultaneously, the noise contribution may be filtered out at least partially, thus improving the sound quality of the audio signal resulting from such an analysis.

According to an exemplary embodiment of the invention, an essentially noise-free voice pickup system may be provided, by use of a pair of microphones mounted on earphones. A corresponding method of picking up clean voice may include the generation of a sum and a - A -

difference channel signal from left and right microphones pick-up. Furthermore, a clean audio signal may be generated by spectral subtraction of different signals from the sum signal. Furthermore, a spectral correction of the cleaned signal thus obtained may be performed by applying the inverse of a mouth to ear transfer function to the clean signal. In other words, according to an exemplary embodiment of the invention, a noise- free pickup of a voice signal may be provided by the aid of two microphones, wherein each of these (exactly or at least) two microphones may be positioned close to an ear of the human user. In this context, the following audio signal-processing scheme may be applied.

A summing signal may be produced from the left and right microphone signal (which summing signal will also be denoted in the following as "primary signal"). The primary signal may comprise mostly voice, that is to say speech.

Furthermore, a difference signal of the left and right microphone signal may be produced (which will be denoted as "secondary signal" in the following) . The secondary signal may mostly comprise noise. Subsequently, performing a spectral subtraction by subtracting the frequency components present in the secondary signal from the primary signal may produce a noise cleaned so-called "clean signal". The person skilled in the art knows how to perform a spectral subtraction as such.

Optionally, a "spectral corrected clean speech signal" may be produced by performing a spectral correction in order to compensate for the effect that the microphones may be close to the ear and not in the front of the user. Such a spectral correction may be based on applying the inverse of a mouth to ear transfer function to the "clean signal".

According to one aspect of the invention, a specific noise immune voice capture solution is provided by just using a pair of microphones or earphones equipped with one outer microphone per side. Because the mouth of a user is always at a fixed position with respect to the microphones, specific and low-cost noise reduction solutions can be used to pick up the user's voice. Moreover, a specific spectral correction can also be applied to compensate the mouth to ear transfer function.

One exemplary aspect of the invention is related to an application of a principle of voice pickup with multiple microphones for a case in which the microphones are placed in or close to the ears of the listener. Thus, exemplary fields of application of the invention are headphones and headsets. A corresponding setup may have some advantages with respect to prior art denoising algorithms. Particularly, both or all microphones may be arranged essentially at the same distance to the mouth. Therefore, some parts of general solutions can be frozen and specifically adapted to this setup, such as the decomposition of the signals to generate L+R and L-R signals, wherein L is an audio signal received from a left microphone and R is an audio signal received by a right microphone. One aspect of the invention is thus related to the fact that the transfer function for the two microphones may be the same, and particularly do not vary in time.

Concerning the noise compensation, a filter according to an exemplary embodiment of the invention may give the captured speech the frequency balance that it would have if a microphone would be placed in front of the mouth. According to an aspect of the invention, the two microphones may be worn on the head of a human user. In other words, the two microphones may always be in a fixed position with respect to the audio source (for instance the mouth), so that it may be securely ensured that the left and right ear to mouth transfer functions are identical.

Thus, an aspect of the invention refers to the symmetrical arrangement of the microphones with respect to the signal source, so that an identical transfer function may be achieved. This may allow significantly reducing, suppressing or cancelling out contributions of noise.

Because of the specific geometry of such a system, a simple and efficient solution may be provided. Particularly, a conversion block having a fixed behaviour may benefit from the geometry of the setup.

According to an exemplary embodiment of the invention, a specific noise immune voice capture solution is provided by using a pair of headphones or by using earphones equipped with one outer microphone per side. The inclusion of the microphones into such conventional headphones or earphones may be totally invisible from the user point of view and may be free of any impact on the form factor. In other words, this does not introduce any size constraints. The mouth or any other sound source may be always at a fixed position with respect to the microphones, thus specific and low cost noise reduction solutions can be used. Moreover, a specific spectral correction can also be applied to compensate the mouth to ear transfer function. The device according to the invention may be adapted to capture and/or process a data stream of video data or audio data. However, such media content is not the only type of data that may be processed with the scheme according to the invention. Generating noise- reduced signals may be an issue for both, video (that is to say combined audio and visual data) processing and (pure) audio processing.

A number of microphones may be used to obtain a signal which does contain a reduced amount of ambient noise so that the signal to noise ratio may be improved. In this context, one aspect is related to "beam forming", that is to say shaping an acoustic beam through the control of the geometry and/or the processing capabilities of the audio array.

Referring to the dependent claims, further exemplary embodiments of the invention will be described.

In the following, exemplary embodiments of the device for processing audio data will be described. However, these embodiments also apply for the method of processing audio data, for the computer-readable medium and for the program element.

The device may comprise a support element on which the first microphone and the second microphone may be mounted so that the first microphone and the second microphone may be located symmetrically with respect to the acoustic source in an operation state in which the device is mounted or fixed on the acoustic source. According to this embodiment, a support element like a hanger may be provided receiving or fastening the two microphones, so that such a support element may contribute to the provision and maintenance of the symmetric configuration of the two microphones. For instance, the support element may be a hanger of a headphone, or may be a structural component of a head mounted device.

The device may further comprise a first earphone on which the first microphone may be mounted and may comprise a second earphone on which the second microphone may be mounted. In other words, the device according to the invention may be realized as a combined sound detection and sound reproduction system, particularly as some kind of headset which may, for instance, be implemented within a mobile phone, for instance when a user is driving in a car while using the mobile phone. In such a scenario, a combined sound detection and sound reproduction functionality may be provided. When a user carries the device, for instance on his head, the user may simultaneously listen to sound, for instance generated by a remote communication partner, and may speak, wherein the cleaned speech of the user may be transmitted to the communication partner with low noise.

As an alternative to the use of two earphones with attached microphones, two headphones may be used at which the two microphones may be arranged. The microphones may be attached to the headphones in a symmetrical manner, in accordance with the symmetric anatomy of a human user. Even an implementation with two loudspeakers instead of earphones or headphones may be possible. The first microphone may be arranged in a vicinity of a first ear, and the second microphone may be arranged in a vicinity of a second ear of a human user, in a scenario in which the device is mounted on a head of the human user. According to this configuration, the microphones may be easily attached close to the ears of the user. Thus, the microphones may be brought and maintained automatically in a symmetric arrangement with respect to a human user.

The processing unit of the device may be adapted to process the first audio data in combination with the second audio data to generate an audio output signal having reduced noise compared to the first audio and the second audio data. In other words, noise reduction algorithms may be applied when processing the signals received from the two microphones in order to improve sound quality. Many noise reduction algorithms are suitable which may benefit from the fact that the location at which the sound is detected is symmetric with respect to the sound source.

For instance, the processing unit may be adapted to generate a third audio data by summing an amplitude of the first audio data and an amplitude of the second audio data. Experiments have shown that such a summing signal in accordance with the symmetric configuration of the two microphones may particularly reflect voice components, wherein noise components may be suppressed in such a sum signal.

Furthermore, the processing unit may be adapted to generate a fourth audio data by subtracting an amplitude of the second audio data from an amplitude of the first audio data, or vice versa. Experiments have shown that such a subtraction signal may essentially contain noise contributions and may thus serve as a reference signal reflecting uncharacteristic underground audio content.

Moreover, the processing unit may be adapted to generate a fifth audio data by subtracting frequency components of the fourth audio data from frequency components of the third audio data. Such a frequency analysis of the sum signal and the difference signal generated in accordance with the above description may provide a clean voice signal that may have an enormously reduced noise contribution and may thus allow improving the audio quality significantly. The combination of forming a sum signal, forming a difference signal and the frequency specific subtraction of the noise reflecting signal from the voice reflecting signal may be a suitable output signal of the audio system. Such a spectral subtraction may include performing a Fourier analysis.

Concerning spectral subtraction, an appropriate technique is the Boll method, which is described in "http://www.lss.uni-stuttgart.de/matlab/adaptive/bolltheory.en.htmr', for instance. In "http://www.ind.rwth- aachen.de/researcMnoise_reduction.htod#Spectral%20Weighting", spectral subtraction is actually used as a way of performing "spectral weighting", which may be seen as a more general term. Therefore, spectral subtraction or any equivalent noise reduction technique may be used in the frame of the invention.

However, in order to further refine the output sound quality, the processing unit may be adapted to generate an audio output signal by processing the fifth audio data by applying an inverse of a transfer function reflecting a distance between the acoustic source on the one hand and the first microphone and/or the second microphone on the other hand. More particularly, the processing unit may be adapted to generate an audio output signal by processing the fifth audio data by applying an inverse of a mouth to ear transfer function.

This processing step may mainly benefit from the symmetrical arrangement of the microphones and may allow producing a high quality output signal. For instance, the device according to the invention may be realized as one of the group consisting of a portable audio player, a portable video player, a head mounted display, a mobile phone comprising earphones or headphones, a medical communication system between patient and operator (for instance a physician), a body-worn device, a DVD player, a CD player, a harddisk-based media player, an internet radio device, a public entertainment device and an MP3 player. However, these applications are only exemplary, and other applications in many fields of the art are possible and in the frame of the invention.

Particularly, the invention may be advantageously applied in all voice communication systems where small form factors are required. For instance, a combined system of a mobile phone and earphones, or a medical system communication system between operator and patient are advantageous fields of application. One particular application of the device according to the invention is a mobile phone for use in conjunction with a stereo headset or stereo earpieces. Another application is a device offering a combined functionality of recording audio signals from the environment and reproducing audio content, like the recorded data or media content stored on storage medium like a CD, a DVD or a harddisk. The aspects defined above and further aspects of the invention are apparent from the examples of embodiment to be described hereinafter and are explained with reference to these examples of embodiment. BRIEF DESCRIPTION OF THE DRAWINGS

The invention will be described in more detail hereinafter with reference to examples of embodiment but to which the invention is not limited.

Figure 1 schematically illustrates a device of processing audio data according to an exemplary embodiment of the invention,

Figure 2 schematically illustrates a processor according to an exemplary embodiment of the invention of the device of processing audio data shown in Figure 1,

Figure 3 illustrates a diagram showing a measured mouth to ear transfer function for different listeners.

DESCRIPTION OF EMBODIMENTS

The illustration in the drawing is schematically. In different drawings, similar or identical elements are provided with the same reference signs.

In the following, referring to Fig. 1, an audio data processing system 100 according to an exemplary embodiment of the invention will be described.

The audio data processing device 100 is adapted for processing audio data 106, 107 as will be described in the following.

The audio data processing device 100 comprises a first microphone 103 which is adapted to detect the first audio data 106 emitted by the mouth of a human user 101 as an acoustic source, that is to say as a source of acoustic waves when the human user 101 speaks. Furthermore, a second microphone 104 is provided which is adapted to detect the second audio data 107 emitted by the mouth of the human user 101.

As can be taken from Fig. 1, when the human user 101 speaks, a speech signal 105 is emitted which is also transmitted to the microphones 103, 104 which are arranged close to the ears and thus essentially symmetrical with respect to the mouth of the user 101.

Particularly, the microphones 103, 104 are provided close to the ears of the human user 101 so that the distance between the mouth of the human being 101 and each of the microphones 103, 104 is essentially the same.

Further referring to Fig. 1, acoustic noise 102 is present in the environment of the device 100, for instance noise generated by an engine of a car when the user 101 uses the audio data processing device 100 during driving.

Furthermore, the device 100 comprises a hanger (not shown in Fig. 1) having an essentially half circular shape and connecting both ears of the user 101 above the head of the user 101. The microphones 103, 104 are arranged close to the ends of the hanger. With the use of this hanger as a support element, the symmetric configuration of the two microphones 103, 104 with respect to the mouth of the human 101 is ensured and stabilized even when the user 101 moves, for instance when using the device 100 during jogging. A first headphone (not shown in Fig. 1) is mounted close to the first microphone

103, and a second headphone (not shown in Fig. 1) is mounted close to the second microphone 104. Accordingly, the first microphone 103 is arranged in a vicinity of a first ear of the human 101, and the second microphone 104 is arranged in a vicinity of a second ear of the human 101. As can further be taken from Fig. 1, a microprocessor 108 (for instance a central processing device, CPU) is provided in the device 100, which microprocessor 108 is supplied with a left microphone audio data 106 captured by the first microphone 103 and with a right microphone audio data 107 captured by the second microphone 104. The processing unit 108 then processes the left microphone audio data 106 and the right microphone audio data 107, as will be described below in more detail.

The microprocessor 108, which may also be denoted as a clean voice pickup unit, processes the speech plus noise signals 106, 107 as received by the first and the second microphones 103, 104. By processing the signals, the microprocessor 108 generates a clean speech signal 109 that is provided at an output of the audio data processing device 100. Thus, Fig. 1 illustrates the clean voice pickup principle according to one aspect of the invention.

In the following, referring to Fig. 2, the functionality of the microprocessor 108 will be described in more detail.

The left microphone audio data 106 and the right microphone audio data 107 are supplied to a conversion block 200. By processing the data 106, 107 simultaneously and in combination, the conversion block 200 generates a primary audio data 201 and a secondary audio data 202. These signals 201, 202 are provided to a spectral subtraction block 203. By processing the data 201, 202, the spectral subtraction block 203 generated and provides, at an output thereof, a clean speech audio data 204 that is supplied to an input of a spectral correction block 205. After having accordingly processed the signal 204, the spectral correction block 205 outputs, at an output thereof, the corrected clean speech signal 109 that is essentially free of noise. The primary audio data 201 is generated by the conversion block 200 by summing an amplitude of the left microphone audio data 106 and an amplitude of the right microphone audio data 107.

The conversion block 200 generates the secondary audio data 202 by subtracting the amplitude of the right microphone audio data 107 from the amplitude of the left microphone audio data 106.

The spectral subtraction block 203 then generates the clean speech audio data 204 by subtracting frequency components of the secondary audio data 202 from frequency components of the primary audio data 201, that is to say by processing the data 201, 202 in the frequency domain.

The spectral correction block 205 then generates the corrected clean speech data 109 by processing the clean speech audio data 204 by applying an inverse of a mouth to ear transfer function.

The left and right microphone signals 106, 107 are passed through the conversion block 200 which is specifically converts these two inputs 106, 107 into the signal 201 which contains as much speech as possible (primary signal), and into the signal 202 which contains as few speech as possible (secondary signal, which may also be denoted "noise reference"). In the specific case of the microphones 103, 104 placed in the ears and thus equidistant to the mouth of user 101, it has been found that: - summing the left and right microphone inputs 106, 107 may already provide a 3dB speech to noise ratio improvement because of the diffuse and decorrelated characteristics of the ambient noise. Indeed, both microphones 103, 104 may capture the same speech signal which may lead to a gain of 6dB for the speech signal (amplitude summation). Because of the left and right noise components may be most of the time decorrelated, summing is equivalent to an energy summation, leading to a gain of only 3dB instead of 6dB for the speech.

- the difference between the left and right microphone signals 106, 107 may show a significant reduction of the speech to noise ratio and can often be considered as a good noise reference for subsequent processing. The primary signal 201 and the secondary signal 202 are then passed through the spectral subtraction algorithm realized in the spectral subtraction unit 203 which basically subtracts the frequency components present in the noise reference from the primary signal 201 in order to retrieve the clean speech signal 204. The block 205 of Fig. 2 is also denoted as the spectral correction unit 205. Experiments have shown that the transfer function between the mouth of the user 101 and each of the microphones 103, 104 follows generic characteristic, as can be seen in Fig. 3.

Fig. 3 shows a diagram 300 having an abscissa 301 along which the frequency in Hz is plotted on a logarithmic scale. Along an ordinate 302 of the diagram 300, an amplitude of the signals in dB is plotted. In other words, Fig. 3 shows a measured mouth to ear transfer function for different listeners, wherein a particular one of the curves plotted in Fig. 3 is assigned to each listener. Fig. 3 shows individual spectra of all recordings and a regression curve 303. By applying the inverse of this mouth to ear transfer function 300 to the clean speech signal 204, the same spectral characteristics can be recovered as what a microphone 103, 104 in front of the user's mouth would have picked up. It can either be modelled by a first order shelving treble boost cutting at 1 kHz and stopping at 10 kHz (the inverse of the thick curve 303 on Fig. 3), or either individualized for each listener. It should be noted that the term "comprising" does not exclude other elements or steps and the "a" or "an" does not exclude a plurality. Also elements described in association with different embodiments may be combined.

It should also be noted that reference signs in the claims shall not be construed as limiting the scope of the claims.

Claims

Claims:

1. A device (100) for processing audio data (106, 107), wherein the device (100) comprises a first microphone (103) adapted to detect first audio data (106) emitted by an acoustic source (101); a second microphone (104) adapted to detect second audio data (107) emitted by the acoustic source (101); a processing unit (108) adapted to combine the first audio data (106) with the second audio data (107) to thereby reduce acoustic noise; wherein the first microphone (103) and the second microphone (104) are positioned essentially symmetrically with respect to the acoustic source (101).

2. The device (100) according to claim 1, comprising a support element on which the first microphone (103) and the second microphone (104) are mounted in such a manner that the first microphone (103) and the second microphone (104) are positioned symmetrically with respect to the acoustic source (101) when the device (100) is mounted on the acoustic source (101).

3. The device (100) according to claim 1 or 2, comprising a first earphone on which the first microphone (103) is mounted and comprising a second earphone on which the second microphone (104) is mounted.

4. The device (100) according to claim 1, comprising a first headphone on which the first microphone (103) is mounted and comprising a second headphone (104) on which the second microphone (104) is mounted.

5. The device (100) according to claim 1, wherein the first microphone (103) and the second microphone (104) are positioned symmetrically with respect to the mouth of a human user (101) being the acoustic source when the device (100) is mounted on a head of the human user (101).

6. The device (100) according to claim 1, wherein the first microphone (103) is arranged in vicinity of a first ear and the second microphone (104) is arranged in vicinity of a second ear of a human user (101).

7. The device (100) according to claim 1, wherein the processing unit (108) is adapted to combine the first audio data (106) with the second audio data (107) to generate an audio output signal (109) having reduced acoustic noise compared to the first audio data (106) and the second audio data (107).

8. The device (100) according to claim 1, wherein the processing unit (108) is adapted to generate a third audio data (201) by summing an amplitude of the first audio data (106) and an amplitude of the second audio data (107).

9. The device (100) according to claim 1, wherein the processing unit (108) is adapted to generate a fourth audio data (202) by subtracting an amplitude of the second audio data (107) from an amplitude of the first audio data (106).

10. The device (100) according to claims 8 and 9, wherein the processing unit (108) is adapted to generate a fifth audio data (204) by subtracting frequency components of the fourth audio data (202) from frequency components of the third audio data (201).

11. The device (100) according to claim 10, wherein the processing unit (108) is adapted to generate an audio output signal (109) by processing the fifth audio data (204) by applying an inverse of a transfer function reflecting a distance between the acoustic source (101) on the one hand and the first microphone (103) and the second microphone (104) on the other hand.

12. The device (100) according to claims 5 and 6 and 10, wherein the processing unit (108) is adapted to generate an audio output signal (109) by processing the fifth audio data (204) by applying an inverse of a mouth to ear transfer function.

13. The device ( 100) according to claim 1 , realized as at least one of the group consisting of a portable audio player, a portable video player, a head mounted display, a mobile phone comprising earphones or headphones, a medical communication system, a body- worn device, a DVD player, a CD player, a harddisk- based media player, an internet radio device, a public entertainment device, and an MP3 player.

14. A method of processing audio data (106, 107), wherein the method comprises the steps of detecting, by means of a first microphone (103), first audio data (106) emitted by an acoustic source (101); detecting, by means of a second microphone (104), second audio data (107) emitted by the acoustic source (101), wherein the first microphone (103) and the second microphone (104) are positioned essentially symmetrically with respect to the acoustic source (101); combining the first audio data (106) with the second audio data (107) to thereby reduce acoustic noise.

15. A computer-readable medium, in which a computer program of processing audio data (106, 107) is stored, which computer program, when being executed by a processor (108), is adapted to carry out or control the following method steps detecting, by means of a first microphone (103), first audio data (106) emitted by an acoustic source (101); detecting, by means of a second microphone (104), second audio data (107) emitted by the acoustic source (101), wherein the first microphone (103) and the second microphone (104) are positioned essentially symmetrically with respect to the acoustic source (101); combining the first audio data (106) with the second audio data (107) to thereby reduce acoustic noise.

16. A program element of processing audio data (106, 107), which program element, when being executed by a processor (108), is adapted to carry out or control the method steps of detecting, by means of a first microphone (103), first audio data (106) emitted by an acoustic source (101); detecting, by means of a second microphone (104), second audio data (107) emitted by the acoustic source (101), wherein the first microphone (103) and the second microphone (104) are positioned essentially symmetrically with respect to the acoustic source (101); combining the first audio data (106) with the second audio data (107) to thereby reduce acoustic noise.