Microphone system comprising a distance sensor and method of operating such microphone system

Abstract

Microphone system (1), comprising: - a microphone capsule (2), having a primary acoustic sensor (4) configured to capture arriving sound (5) from a sound source (6) at least over the entire frequency range between 100 Hz and 16 kHz and to transduce the arriving sound (5) into a first transduced audio signal (7); - at least one distance sensor (8) for measuring a distance (R) to the sound source (6); - a phantom-powered processing unit (9) configured to • compensate for amplitude variations of the first transduced audio signal (7) due to distance changes of the sound source (6) by applying a variable adaption gain (11) to the first transduced audio signal (7), the variable adaption gain (11) being adapted to a measured distance (R) to the sound source (6) and • output an output audio signal (12) comprising the first transduced audio signal (10).

Description

[0001] MICROPHONE SYSTEM COMPRISING A DISTANCE SENSOR AND METHOD OF OPERATING SUCH MICROPHONE SYSTEM

[0002] The invention relates to a microphone system and to a method of operating a microphone system .

[0003] High-quality sound productions , such as those conducted in professional recording studios , usually demand studio quality microphones , trained personnel and controlled acoustic conditions . Such work comprises careful selection of appropriate microphones , practical techniques regarding controlled acoustic environments , precise microphone placement and (post- ) processing . The ultimate goal is to achieve a finely balanced audio signal devoid of unwanted disturbances and signal variations . The final audio product that emerges from this process can be regarded as having studio quality sound .

[0004] Sound source movement can influence the recorded audio signal signi ficantly . I f a singer shi fts his head or positioning, it can introduce fluctuations in the audio signal ' s amplitude . Also maintaining the optimal distance from the microphone is crucial . When a singer gets too close to the microphone , it may result in an ampli fication of low frequencies , whereas being too far from the microphone can lead to a suppression of these same frequencies . Trained vocalists are typically able to maintain a speci fic distance to a microphone in controlled environments . However, in more challenging environments and situations , such as interviews in public places , it can be di f ficult to maintain a certain distance between a sound source and the microphone .

[0005] US 2013 / 0202130 Al discloses a handheld device , in particular a cell phone , comprising a microphone , which is configured to measure the distance between a user' s head and the device . To this end, infrared or ultrasonic distance detection may be used . A gain of an ampli fier of the transduced electrical signal can be repeatedly adj usted to the measured distance . The amount of gain can compensate for the above-described fluctuations in the audio signal ' s amplitude . However, the microphones used in handheld device as disclosed in US 2013 / 0202130 Al typically cannot deliver studio quality sound . Further, with the handheld device presented in US 2013 / 0202130 Al it is not possible to compensate for other physical ef fects .

[0006] AU 2021100005 A4 discloses a microphone system comprising a microphone stand with a movable arm and a movable leg as well as one or more motors . The microphone system further comprises a sensor by means of which the distance between the person and the microphone can be detected . In order to maintain the said distance , the one or more motors may actuate the movable arm and leg to compensate for movements of the person speaking into the microphone .

[0007] Further microphone systems , some of the comprising distance sensors , are known from US 2014 / 112483 Al , US 2013 / 021503 Al , US 9 197 974 Bl , KR 2004 0108492 A, US 11 257 511 Bl and US 2005 / 232447 Al .

[0008] For the typical inexperienced consumer using microphones in uncontrolled environments the production techniques needed to counteract the issues that occur when recording sound with microphones are typically not available , while the demand for recording studio quality sound persists .

[0009] Also , during live performances it is desired to counteract the issues described above .

[0010] In the light of the above remarks , it is thus an obj ective of the present invention to eliminate or at least alleviate at least some of the disadvantages of the prior art . Preferably it is an obj ective of the present invention to provide a compact microphone system and a method of operating a microphone system which enable inexperienced users to record sound with studio quality, thereby compensating physical ef fects that are detrimental to the quality of the recorded sound .

[0011] This obj ective is solved by a microphone system according to claim 1 and a method of operating a microphone system according to claim 14 .

[0012] The microphone system according to claim 1 comprises : - a preferably phantom-powered microphone capsule , preferably a dual-diaphragm condenser microphone capsule , having a primary acoustic sensor configured to capture arriving sound from a sound source , in particular a person, at least over the entire frequency range between 100 Hz and 16 kHz , preferably between 20 Hz and 20 kHz , and to transduce the arriving sound into a first transduced audio signal ;

[0013] - at least one distance sensor for measuring a distance to the sound source ; and

[0014] - a phantom-powered processing unit configured to

[0015] • compensate for amplitude variations of the first transduced audio signal due to distance changes of the sound source by applying a variable adaption gain to the first transduced audio signal , the variable adaption gain being adapted to a measured distance to the sound source and

[0016] • output an output audio signal comprising the first transduced audio signal .

[0017] The method of operating the microphone system according to claim 13 comprises the following steps :

[0018] - capturing arriving sound from the sound source , in particular a person, at least over the entire frequency range between 100 Hz and 16 kHz , preferably between 20 Hz and 20 kHz , with a preferably phantom-powered microphone capsule , preferably a dual-diaphragm condenser capsule , having the primary acoustic sensor ;

[0019] - transducing the arriving sound into the first transduced audio signal with the primary acoustic sensor ;

[0020] - measuring a distance to the sound source with at least one distance sensor ;

[0021] - deriving a variable adaption gain from a measured distance to the sound source ;

[0022] - applying the variable adaption gain of a phantom-powered processing unit to the first transduced audio signal , thereby compensating amplitude variations of the first transduced audio signal due to distance changes of the sound source

[0023] - outputting an output audio signal comprising the first transduced audio signal . Advantageously, with the inventive microphone system, changes in the amplitude level of the output audio signal due to changes in the distance between the sound source and the microphone capsule can be suppressed by measuring said distance and adapting the variable adaption gain to the measured distance and / or orientation . In a more general aspect , the amplitude level of the output audio signal can be kept consistent relative to a baseline amplitude level for each distance . As the amplitude increases / decreases proportional with a decrease / increase of the distance , the variable adaption gain may be preferably selected proportional to the distance . This approach may be referred to as " level balancing" . Alternatively or additionally, also frequency dependent amplitude variations can be compensated with the inventive microphone system . When directional microphones are used, the amplitude of lower frequencies typically increases when the distance of the sound source to the microphone capsule decreases . Conversely, the amplitude of lower frequencies typically decreases when the distance of the sound source to the microphone capsule increases . This frequency dependent behaviour is commonly referred to as the proximity ef fect , which may also be compensated by adapting the variable adaption gain to the measured distance and / or orientation . To this end, the variable adaption gain may be frequency dependent , i . e . , change the amplitude of the transduced audio signal di f ferently for various frequencies . The compensation of the proximity ef fect may be referred to as " low frequency balancing" . The variable adaption gain may be adapted continuously to the measured distance and / or orientation . The variable adaption gain may comprise an ampli fier element . The ampli fier element may ampli fy and / or attenuate the first transduced audio signal . The variable adaption gain may have a gain smaller than 1 for attenuation of the first transduced audio signal or greater than 1 for ampli fication of the first transduced audio signal . I f the variable adaption gain has a gain equal to 1 , the first transduced audio signal may remain unchanged . As described above , the attenuation or ampli fication may be dependent or independent of the frequency of the first transduced audio signal . I f the gain is frequency independent , the first transduced audio signal may be attenuated or ampli fied uni formly over the entire frequency range . I f the variable adaption gain is frequency dependent , the attenuation or ampli fication of the first transduced audio signal may be di f ferent for di f ferent frequencies . For example, lower frequencies may be more attenuated than higher frequencies . This behaviour can be achieved with filters , such as a high-pass filter, which may be included into the variable adaption gain for frequency dependent behaviour . The primary acoustic sensor may be formed by a membrane of the microphone capsule . The microphone capsule is preferably dual-diaphragm condenser microphone capsule . The microphone capsule may have a diameter of at least 8 mm, preferably of at least 15 mm or at least 25 mm . The microphone capsule is configured to capture sound at least over the entire frequency range between 100 Hz and 16 kHz , preferably between 20 Hz and 20 kHz . In this way, sound over the entire frequency range audible to humans may be recorded . In a particularly preferred embodiment of the invention, the microphone capsule and hence the primary acoustic sensor has the following characteristics :

[0024] • The microphone capsule is capable of capturing and transducing sound over the entire frequency range between 100 Hz and 16 kHz , preferably between 20 Hz and 20 kHz ;

[0025] • The signal to noise ratio of the microphone capsule is at least 70 dB, at least 75 dB or at least 80dB ; and

[0026] • The dynamic range of the microphone capsule is at least l O OdB, at least 105 dB or at least 110 dB .

[0027] With a microphone capsule having these characteristics , the obtained first transduced audio signal , studio quality of sound may be achieved . The first transduced audio signal is preferably an electrical signal . The distance sensor can measure the distance between the distance sensor and the sound source and / or the orientation of the distance sensor with respect to the sound source . Through mathematical calculations , in particular geometrical calculations , a distance between the microphone capsule and the sound source and / or an orientation of the microphone capsule with respect to the sound source can be derived . The distance may be a direct distance , i . e . , a direct line of sight . The orientation may be an angle , for example an angle in a hori zontal plane or a solid angle . The output audio signal may be , for example , an electrical or optical signal . The output signal may be an analogue or digital signal . The microphone system may comprise an output connector, for example an XLR output connector, or a sender unit , for example a Bluetooth sender unit , via which the output audio signal may be sent to , for example , a recording unit or a computer . The processing unit may be configured to mute the microphone system when the distance of the sound source to the distance sensor exceeds a predefined threshold and / or i f the distance sensor does not detect the sound source . The processing unit may be configured to mute the microphone system when a predefined threshold of the amplitude of the arriving sound is exceeded . The processing unit may be configured to mute the microphone system when the amplitude of the arriving sound is below a predefined threshold . Muting may be ef fected by setting the variable adaption gain to zero , for example .

[0028] In one embodiment of the invention, the variable adaption gain is a frequency dependent gain or a frequency dependent gain, in particular a filter, is applied to the first transduced audio signal . The filter is preferably a high-pass filter with a variable cut-of f frequency . To achieve frequency-dependent behaviour, the variable adaption gain may comprise a filter, in particular a high-pass filter, with a variable cut-of f frequency . This is advantageous with regard to " low frequency balancing" and allows for a compensation of the proximity ef fect , which primarily af fects lower frequencies . In one embodiment , the variable adaption gain may be provided by an ampli fier element and the ampli fier element may comprise the filter or be combined with the filter in series . A cut-of f frequency of the high-pass filter may be adapted to the measured distance and / or orientation . I f the filter is implemented as analogue filter, the cut-off frequency may be , for example , adapted by changing a resistance value of a resistor . The resistor may be a JFET , which resistance value can be changed . I f the filter is implemented as digital filter, the cut-of f frequency may be changed by adapting the filter parameters . In a preferred embodiment of the invention, the first transduced audio signal is an analogue signal , in particular an analogue electrical signal , and the processing unit is an analogue processing unit , in particular an analogue electrical processing unit . In other words , the signal is not discreti zed with regard to time and amplitude level . However, the analogue processing unit may be controlled by a digital controller, such as a further processing unit . For example , the variable adaption gain and the above-mentioned filter, in particular the cut-of f frequency, may be adapted by such a digital controller .

[0029] Preferably, the microphone system comprises a main body, in particular a housing, into which the processing unit is included, wherein the microphone capsule is attached to or included into the main body, the microphone capsule preferably being covered with a microphone grill . The main body may comprise an output connector which provides the output audio signal . The main body may be made of plastic and / or metal . The main body may have , for instance , a cylindrical or cuboid form . As the components of the microphone system are included or attached to the main body, the main body can be considered as "microphone" in a general meaning . The microphone may be a handheld microphone .

[0030] The at least one distance sensor may be included into or attached to the main body . The at least one distance sensor may be a digital distance sensor, hence providing digital distance data .

[0031] Preferably, the at least one distance sensor is a time-of- f light sensor, in particular an infrared time-of- f light or ultrasonic time-of- f light sensor, configured to measure the distance of the sound source relative to the microphone system and / or the orientation of the sound source relative to the microphone system . The time-of- f light sensor may be a single- zone or a multi- zone time of flight sensor .

[0032] In one embodiment of the invention, the at least one distance sensor is configured to identi fy an obj ect , in particular a head, and to track the obj ect . To this end, at least one multi- zone distance sensor may be used . Said distance sensor divides the field of view in multiple zones , providing a depth map . This information can be further processed to the detect the orientation and type of the tracked obj ect . The further processing of the information to detect the orientation and type of the tracked obj ect may be carried out a neuronal networks .

[0033] The distance sensor may be configured to turn of f the output audio signal when a predefined obj ect is not present and thus not identi fied by the at least one distance sensor . The distance sensor may be configured to turn of f the output audio signal when the distance to the obj ect exceeds a predefined threshold . This predefined threshold may be set by a user . The predefined threshold may be in a range between 10 cm and 90 cm, for example . This feature may be referred to as "mute by distance" . Thus , i f the predefined obj ect is not detected by the at least one distance sensor or is too far away, the microphone system may be muted . Alternatively, the output audio signal may be attenuated to a predefined amplitude level . The predefined obj ect may be a sound source , such as a head of a human .

[0034] In one embodiment of the invention, the microphone system comprises a secondary acoustic sensor, which is preferably included into the microphone capsule , wherein the secondary acoustic sensor is configured to capture arriving sound from the sound source at least over the entire frequency range between 100 Hz and 16 kHz , preferably between 20 Hz and 20 kHz , and to transduce the arriving sound into a second transduced audio signal , preferably wherein the processing unit is configured to

[0035] • compensate for amplitude variations of the second transduced audio signal due to distance changes and / or orientation changes of the sound source by applying a variable adaption gain to the second transduced audio signal , the variable adaption gain being adapted to a measured distance to the sound source and / or a measured orientation with respect to the sound source and

[0036] • include the second transduced audio signal into the output audio signal . The secondary acoustic sensor may be formed by a membrane of the microphone capsule , which may be a dual-diaphragm condenser microphone capsule . In other words , the primary acoustic sensor may be a first membrane of the microphone capsule and the secondary acoustic sensor may be a second membrane of said microphone capsule . The primary and the secondary acoustic sensor both may have essentially the same characteristics . The primary and the secondary acoustic sensor may be directional sensors , each having a directivity with a main direction of directivity . In a preferred embodiment , the first transduced audio signal and the second transduced audio signal are being combined and the combination of the first and second transduced audio signal is being adapted by the variable adaption gain by applying the variable adaption gain to the combination of the first and second transduced audio signal . In this way, both, the first and second transduced audio signal receive the same adaptions , i . e . , level balancing and low frequency balancing . It may be assumed that the distance to the sound source is the same for the first and second acoustic sensor . In the described preferred embodiment , the variable adaption gain being applied to the second transduced audio signal is thus the variable adaption gain that is also being applied to the first transduced audio signal . In other words , one single variable adaption gain may be used for the first and second transduced audio signal or a combination thereof . However, alternatively, another similar variable adaption gain may be applied to the second transduced audio signal . The first and the second transduced audio signal may be combined by summation or subtraction prior to , at or after the output . In one embodiment , the output signal may consist of two separate signals , namely the first transduced audio signal and the second transduced audio signal , preferably provided at di f ferent electrical output contacts of the output . The first transduced audio signal and the second transduced audio signal may be provided as individual signals at the output . The output is preferably an XLR output , in particular according to standard IEC 61076-2- 103 . One output contact of the output may trans fer the first transduced audio signal , one output contact of the output may trans fer the second transduced audio signal and one output contact of the output may be connected to ground potential . The output contacts of the output may be pins . The first transduced audio signal and the second transduced audio signal may be provided as symmetric signals at the output . In one embodiment of the invention, the first transduced audio signal and the second transduced audio signal may be combined to one single signal by symmetric summation or subtraction . The summation or subtraction may be carried out prior to , at or after the output . By summing or subtracting the first and the second transduced audio signal , an overall directivity of the microphone system may be changed . To change the overall directivity, the first and the second transduced audio signals may be weighted .

[0037] The primary acoustic sensor and the secondary acoustic sensor each may have a directivity, said directivities being oriented preferably essentially opposite to each other . In particular, the main direction of directivity of the primary acoustic sensor may point opposite to the main direction of directivity of the secondary acoustic sensor . In other words : The primary and the secondary acoustic sensor may be aligned such that the main directions of directivity point in opposite directions .

[0038] In order to change an overall directivity of the microphone system, in particular of the microphone capsule , the processing unit may be configured to

[0039] • apply a first gain to the first transduced audio signal and a second gain to the second transduced audio signal and combine the first transduced audio signal and the second transduced audio signal to the output audio signal . Preferably, the combination of the first transduced audio signal and the second transduced audio signal may be implemented as symmetric summation or subtraction at or after the output of the microphone . The summation or subtraction of the signals may be carried out by a device connected to output via a cable , for example in an ampli fier, an audio interface or a mixing console . The directivity of the first acoustic sensor may be approximated by a cardioid, for example . Also , the directivity of the second acoustic sensor may be approximated by a cardioid, for example . The overall directivity of the microphone system (which may be described by two essentially back- facing cardioids i f the first and second acoustic sensor are included in the microphone capsule ) may depend solely on the first gain and the second gain and the directivities of the first and second acoustic sensors . In one embodiment of the invention, the first gain and the second gain may each be included into variable adaption gains for the first and second transduced audio signal as described above . Alternatively, the first and the second gain may be separate from the one or more variable adaption gains and may be set between - 1 and 1 .

[0040] I f the first gain is equal to the second gain, the microphone system is essentially omni-directional . I f the first gain is set to , for example , 1 while the second gain is set to 0 , the overall directivity of the microphone system is front cardioid . I f first gain is set to 1 and second gain is set to 0 . 5 , the overall directivity of the microphone system is super-cardioid . I f the first gain is set to 1 and the second gain is set to - 1 , the overall directivity of the microphone system is bi-directional . By adapting the first and the second gain, the overall-directivity of the microphone system can be changed . To this end, the first transduced audio signal and the second transduced audio signal may be summed or subtracted from each other, for example in an ampli fier, an audio interface , or a mixing console .

[0041] In one embodiment of the invention, the first gain and the second gain may be ef fected by adapting a polari zation voltage of the condenser capsule , for example . In this embodiment , the first and second gain may be considered included in the first and second acoustic sensor .

[0042] The first gain and the second gain may be included into variable adaption gains for the first and second transduced signal , respectively, as described above .

[0043] In one embodiment of the invention, the microphone system comprises a further processing unit configured to convert the output audio signal from the processing unit into a digital output signal and to convert the distance to the sound source and / or the orientation with respect to the sound source into a digital representation .

[0044] Optionally, the method of operating a microphone system may further comprise the following steps :

[0045] - capturing the arriving sound from the sound source at least over the entire frequency range between 100 Hz and 16 kHz , preferably between 20 Hz and 20 kHz , with a secondary acoustic sensor, which is preferably included into the microphone capsule ;

[0046] - transducing the arriving sound into a second transduced audio signal with the secondary acoustic sensor ;

[0047] - preferably applying a variable adaption gain to the second transduced audio signal , thereby compensating amplitude variations of the second transduced audio signal due to distance changes and / or orientation changes of the sound source ;

[0048] - applying a first gain to the first transduced audio signal and a second gain to the second transduced audio signal ;

[0049] - outputting the output audio signal comprising the first transduced audio signal and / or the second transduced audio signal , and in this way adapting an overall directivity of the microphone system, in particular of the microphone capsule .

[0050] The adaption of the second transduced audio signal is preferably similar to the adaption of the first transduced audio signal . Thus , the above remarks regarding the variable adaption gain and the first transduced audio signal may be also applied to variable adaption gain and the second transduced audio signal . Thus , " level balancing" and " low frequency balancing" may also performed with the second transduced audio signal or, in another embodiment , a combination of the first and second transduced audio signal . The first transduced audio signal and / or the second transduced audio signal may be included into the output audio signal provided at the output . Preferably, the first and second transduced audio signal are output separately at the output , which may be an XLR output . By combination, in particular by summation, of the first transduced audio signal and the second transduced audio signal , the overall directivity of the microphone system may be adapted . The variable adaption gain applied to the second transduced audio signal may be identical to the variable adaption gain that is applied to the first transduced audio signal . The second transduced audio signal may be an analogue signal . Therefore , the second gain may also be implemented as an analogue gain . The first and second gain may be included into variable adaption gains for the first and second transduced audio signal . However, alternatively, the first and second gains may be separate from the variable adaption gain ( s ) . In a preferred embodiment , the first gain is applied to the first transduced audio signal and the second gain is applied to the second transduced audio signal ; however, the variable adaption gain for level balancing and low frequency balancing may be applied to a combination of the first and second transduced audio signal . The first and second gain may be implemented by polari zation adaption of the microphone capsule .

[0051] In one embodiment of the invention, the microphone system comprises a further processing unit configured to convert the output audio signal from the processing unit into a digital output signal and to convert the distance (R) to the sound source and / or the orientation (cp) with respect to the sound source into a digital representation . The further processing unit may be configured to adapt the variable adaption gain and / or the variable filter, in particular its cut-of f frequency . Furthermore , the further processing unit may be configured to adapt the beam creation by adapting the weights of the first gain and the second gain . This can be achieved by adapting the polari zation of the microphone capsule . The further processing unit may be configured to derive control parameters based on the output signal . At least one or more of the variable adaption gain, and the filter may be adapted according to the control parameters , for examp 1 e .

[0052] For example , the microphone system with the further processing unit may comprise a dynamic range compressor . A dynamic range compressor is preferably configured to estimate the signal envelope of the output audio signal or of the first and / or second transduced audio signal and applies an attenuation to the output audio signal i f the signal envelope exceeds a predefined threshold . The further processing unit may be configured to estimate the signal envelope of the output audio signal or of the first and / or second transduced audio signal and to apply an attenuation accordingly, in case the signal envelope exceeds a predefined threshold. The attenuation may be applied by using the variable adaption gain(s) .

[0053] In one embodiment of the invention, the microphone system may further comprise an additional filter stage to form a de-esser. The de-esser (in particular a further processing unit of the de- esser) is configured to split the output audio signal into at least two frequency bands (i.e., at least an upper frequency band and a lower frequency band) and to apply compression to the upper band, which typically contains sibilance (e.g., "s", "z", typically above 2kHz) .

[0054] Preferably, the processing unit and / or the further processing unit are adapted to be powered by a phantom power source. The phantom power source provides a voltage supply. In one embodiment, the phantom power source may be considered as part of the microphone system. The phantom power source may, for example, provide a voltage of essentially 48 V. However, also other voltage levels are possible, such as, for example, 9 V, 12 V or 24 V. The voltage may be provided from an amplifier or an audio interface. The phantom power source may thus be included into the amplifier or audio interface.

[0055] Optionally, the microphone capsule is a phantom-powered microphone capsule. Any type of capsule may be phantom-powered. In particular, a condenser microphone capsule may be phantom powered. For example, a polarization of the condenser microphone capsule may be generated by the phantom power voltage. To this end, the phantom power voltage may be directly applied or via an electrical circuit that may generate a voltage for polarization. The microphone system may comprise a phantom power source providing a voltage supply to the primary acoustic sensor. In particular, a P48 (or 48 V) phantom power may be utilized. Optionally, the processing unit may be powered by the phantom power source in addition to the microphone capsule. Optionally, also the further processing unit may be powered by the phantom power source. Optionally, the entire microphone system may be powered by the phantom power source.

[0056] In more general aspects, the invention may be described with the following embodiments :

[0057] Embodiment 1 : Microphone system, comprising :

[0058] - a microphone capsule , preferably a dual-diaphragm condenser microphone capsule , having a primary acoustic sensor configured to capture arriving sound from a sound source , in particular a person, at least over the entire frequency range between 100 Hz and 16 kHz , preferably between 20 Hz and 20 kHz , and to transduce the arriving sound into a first transduced audio signal ;

[0059] - at least one distance sensor for measuring a distance to the sound source and / or an orientation with respect to the sound source ; and

[0060] - a processing unit configured to

[0061] • compensate for amplitude variations of the first transduced audio signal due to distance changes and / or orientation changes of the sound source by applying a variable adaption gain to the first transduced audio signal , the variable adaption gain being adapted to a measured distance to the sound source and / or a measured orientation with respect to the sound source and

[0062] • output an output audio signal comprising the first transduced audio signal .

[0063] Embodiment 2 : Microphone system according to embodiment 1 , wherein the variable adaption gain is a frequency dependent gain or a frequency dependent gain, in particular a high-pass filter, is applied to the first transduced audio signal .

[0064] Embodiment 3 : Microphone system according to embodiment 1 or 2 , wherein the first transduced audio signal is an analogue signal , in particular an analogue electrical signal , and the processing unit is an analogue processing unit , in particular an analogue electrical processing unit .

[0065] Embodiment 4 : Microphone system according to any one of embodiments 1 to 3 , wherein the microphone system comprises a main body, in particular a housing, into which the processing unit is included, wherein the microphone capsule is attached to or included into the main body, the microphone capsule preferably being covered with a microphone grill .

[0066] Embodiment 5 : Microphone system according to embodiment 4 , wherein the at least one distance sensor is included into or attached to the main body .

[0067] Embodiment 6 : Microphone system according to any one of embodiments 1 to 5 , wherein the at least one distance sensor is a time-of- f light sensor, in particular an infrared time-of- f light or ultrasonic time-of- f light sensor configured to measure the distance of the sound source relative to the microphone system and / or the orientation of the sound source relative to the microphone system .

[0068] Embodiment 7 : Microphone system according to any one of embodiments 1 to 6 , wherein the at least one distance sensor is configured to identi fy an obj ect , in particular a head, and to track the obj ect .

[0069] Embodiment 8 : Microphone system according to embodiment 7 , wherein the distance sensor is configured to turn of f the output audio signal when a predefined obj ect is not present and thus not identi fied by the at least one distance sensor .

[0070] Embodiment 9 : Microphone system according to any one of embodiments 1 to 8 , where the microphone system comprises a secondary acoustic sensor, which is preferably included into the microphone capsule , wherein the secondary acoustic sensor is configured to capture arriving sound from the sound source at least over the entire frequency range between 100 Hz and 16 kHz , preferably between 20 Hz and 20 kHz , and to transduce the arriving sound into a second transduced audio signal , preferably wherein the processing unit is configured to

[0071] • compensate for amplitude variations of the second transduced audio signal due to distance changes and / or orientation changes of the sound source by applying a variable adaption gain to the second transduced audio signal , the variable adaption gain being adapted to a measured distance to the sound source and / or a measured orientation with respect to the sound source and • include the second transduced audio signal into the output audio signal .

[0072] Embodiment 10 : Microphone system according to embodiment 9 , wherein the primary acoustic sensor and the secondary acoustic sensor each have a directivity, said directivities being oriented preferably essentially opposite to each other .

[0073] Embodiment 11 : Microphone system according to embodiment 9 or 10 , wherein in order to adapt an overall directivity of the microphone system, in particular of the microphone capsule , the processing unit is configured to

[0074] • apply a first gain to the first transduced audio signal and a second gain to the second transduced audio signal and

[0075] • combine the first transduced audio signal and the second transduced audio signal to the output audio signal .

[0076] Embodiment 12 : Microphone system according to any one of embodiments 1 to 11 , characterized in by a further processing unit configured to convert the distance to the sound source and / or the orientation with respect to the sound source into a digital representation and to preferably convert the output audio signal from the processing unit into a digital output signal .

[0077] Embodiment 13 : Microphone system according to any one of embodiments 1 to 12 , wherein the microphone capsule is a phantom-powered microphone capsule .

[0078] Embodiment 14 : Method of operating a microphone system with the following steps :

[0079] - capturing arriving sound from a sound source , in particular a person, at least over the entire frequency range between 100 Hz and 16 kHz , preferably between 20 Hz and 20 kHz , with a microphone capsule , preferably a dual-diaphragm condenser capsule , having a primary acoustic sensor ;

[0080] - transducing the arriving sound into a first transduced audio signal with the primary acoustic sensor ;

[0081] - measuring a distance to the sound source and / or an orientation with respect to the sound source with at least one distance sensor ; - deriving a variable adaption gain from a measured distance to the sound source and / or a measured orientation with respect to the sound source ;

[0082] - applying the variable adaption gain to the first transduced audio signal , thereby compensating amplitude variations of the first transduced audio signal due to distance changes and / or orientation changes of the sound source ; and

[0083] - outputting an output audio signal comprising the first transduced audio signal .

[0084] Embodiment 15 : The method according embodiment 14 , with the following further steps :

[0085] - capturing the arriving sound from the sound source at least over the entire frequency range between 100 Hz and 16 kHz , preferably between 20 Hz and 20 kHz , with a secondary acoustic sensor, which is preferably included into the microphone capsule ;

[0086] - transducing the arriving sound into a second transduced audio signal with the secondary acoustic sensor ;

[0087] - preferably applying a variable adaption gain to the second transduced audio signal , thereby compensating amplitude variations of the second transduced audio signal due to distance changes and / or orientation changes of the sound source ;

[0088] - applying a first gain to the first transduced audio signal and a second gain to the second transduced audio signal ;

[0089] - outputting the output audio signal comprising the first transduced audio signal and / or the adapted transduced audio signal , and in this way adapting an overall directivity of the microphone system, in particular of the microphone capsule .

[0090] In the following, the invention will be described with figures , to which the invention shall not be restricted to .

[0091] The figures show : Fig . 1 a microphone system;

[0092] Fig . 2 a block diagram of a processing unit for a microphone capsule with a primary acoustic sensor

[0093] Fig . 3 a block diagram of a processing unit for a microphone capsule with a primary acoustic sensor and a secondary acoustic sensor ;

[0094] Fig . 4A, Fig . 4B and Fig . 4C bode diagrams that illustrate level balancing;

[0095] Fig . 5A, Fig . 5B and Fig 5C bode diagrams that illustrate low frequency balancing;

[0096] Fig . 6A-D several directivity patterns .

[0097] Fig . 1 shows a microphone system 1 , comprising a microphone capsule 2 , in this exemplary embodiment a dual-diaphragm condenser microphone capsule 3 , having a primary acoustic sensor 4 configured to capture arriving sound 5 from a sound source 6 , in particular a person ( see fig . 4A, for example ) , at least over the entire frequency range between 100 Hz and 16 kHz , preferably between 20 Hz and 20 kHz . Furthermore , the primary acoustic sensor 4 is configured to transduce the arriving sound 5 into a first transduced audio signal 7 . The microphone system 1 comprises at least one distance sensor 8 ( in this example exactly one distance sensor 8 ) for measuring a distance R to the sound source 6 and / or an orientation cp with respect to the sound source 6 .

[0098] The microphone system 1 furthermore comprises a processing unit 9 configured to compensate for amplitude variations of the first transduced audio signal 7 due to distance changes and / or orientation changes of the sound source 6 by applying a variable adaption gain 11 to the first transduced audio signal 7 . The variable adaption gain 11 is adapted to a measured distance R to the sound source 6 and / or a measured orientation cp with respect to the sound source 6 . This may be referred to as " level balancing" ( see figures 4A, 4B and 4C for details ) . Furthermore , the processing unit 9 is configured to output an output audio signal 12 comprising the first transduced audio signal 7. The first transduced audio signal 7 and the output audio signal 12 containing the first transduced audio signal 7 are preferably analogue signals.

[0099] In the embodiment shown, the microphone capsule 2 has a diameter of 25 mm. The primary acoustic sensor 4 is formed by a membrane of microphone capsule 2.

[0100] The processing unit 9, which is preferably an analogue processing unit, furthermore comprises a variable filter 26. By means of the variable filter 26 the output audio signal 12 at the output 23 of the microphone system 1 may be kept consistent over the whole frequency range. The adjustment of the variable filter 26 may be based on the measured distance R and / or orientation cp. This may be referred to as "low frequency balancing" (see figures 5A, 5B and 5C for details) .

[0101] As already mentioned, the first audio signal 7 is an analogue signal, in this example an analogue electrical signal. The processing unit 9 is an analogue processing unit, in this exemplary embodiment an analogue electrical processing unit.

[0102] The microphone system 1 comprises a main body 13, in particular a housing, into which the processing unit 9 is included. The microphone capsule 2 is attached to the main body 13. The microphone capsule 2 is attached to an upper side 19 of the cylindrical main body 13, which is made of plastic and / or metal. The microphone capsule 2 is mounted to the main body 13 (i.e., the housing) by means of a preferably oblong retaining element 22. The microphone capsule 2 may optionally be covered by a grill (not shown) . In the shown embodiment, the microphone system 1 has the size of a handheld device.

[0103] The (at least one) distance sensor 8 is included into the main body 13. The main body 13, i.e., the housing, comprises an opening 21 for the distance sensor 8, such that the distance sensor 8 may have a direct line of sight to the sound source 6.

[0104] The processing unit 9 and the distance sensor 8 are both included into the housing in the shown embodiment .

[0105] The output audio signal 12 may be provided at an output 23 of the main body 13 . By means of the distance sensor 8 , a distance R between the microphone system 1 and the sound source 6 and / or an orientation cp of the sound source 6 with respect to the microphone system 1 can be obtained . The distance sensor 8 is a time-of- f light sensor 24 , in particular a multi- zone time-of- f light sensor . In this exemplary embodiment the time-of- f light sensor is a 8x8 multi- zone ranging sensor with wide field of view, in particular a VL53L8CX manufactured by STMicroelectronics . In a preferred embodiment , multiple distance sensors 8 are provided . The distance R and / or orientation cp may be obtained, for example , with a frequency of 5 measurements per second .

[0106] The at least one distance sensor 8 may be configured to identi fy an obj ect , for example a head 25 ( see fig . 4A) , and to track the obj ect , which is preferably the sound source 6 . The microphone system 1 , in particular the processing unit 9 , can be configured to turn of f the output audio signal 12 when a predefined obj ect ( in this example the head 25 ) is not present and thus not identi fied by the distance sensor 8 or exceeds an adj ustable predefined threshold . In other words , the microphone system 1 is muted when the distance sensor 8 does not detect the presence of the predefined obj ect or the obj ect is too far away from the microphone system 1 .

[0107] In this exemplary embodiment , the microphone system 1 comprises a secondary acoustic sensor 14 , which in this example is included into the microphone capsule 2 . The secondary acoustic sensor 14 is configured to capture arriving sound 5 from the sound source 6 at least over the entire frequency range between 100 Hz and 16 kHz , preferably between 20 Hz and 20 kHz , and to transduce the arriving sound 5 into a second transduced audio signal 15 . Similar to the first transduced audio signal 7 , also the second transduced audio signal 15 may be included into the output audio signal 12 . The secondary acoustic sensor 14 is formed by a membrane of the microphone capsule 2 (which in this example is a dual-diaphragm condenser microphone capsule ) . The primary acoustic sensor 4 and the secondary acoustic sensor 14 each have a directivity, said directivities being oriented essentially opposite to each other . Otherwise , the primary acoustic sensor 4 and the secondary acoustic sensor 14 both have essentially the same characteristics .

[0108] In order to adapt an overall directivity of the microphone system 1 , in particular of the microphone capsule 2 , the processing unit 9 is configured to generate an intermediate audio signal 16 by applying a first gain 17 and a second gain 20 to the transduced audio signals 7 and 15 accordingly and summing the transduced audio signals 7 and 15 using a summation stage 37 after applying the gains 17 , 20 . The first 17 and second gain 20 are adapted to the distance R to the sound source 6 and / or the orientation cp with respect to the sound source 6 in order to achieve a desired overall directivity of the microphone system 1 that is directed to the sound source . The overall directivity is achieved by a combination of the weighted transduced audio signals 7 and 15 . After summation, the intermediate audio signal 16 is fed into the variable filter for " low frequency balancing" and into the variable adaption gain 11 for " level balancing" .

[0109] The cut-of f frequency of the filter 26 and the variable adaption gain both depend on the distance R to the sound source 6 . In a far- field approximation it may be assumed that the distance for both acoustic sensors 4 and 14 are the same . After passing the variable adaption gain 11 , the signal may be referred to as output signal 12 , which may be fed to the output 23 and to a further processing unit 18 . The output is preferably an XLR output . First 17 and second gains 20 may be implemented by regulating the polari zation of the microphone capsule 2 .

[0110] Preferably, the first 17 and the second gain 20 has an ampli fication between - 1 and 1 , respectively . Likewise , the variable adaption gain 11 preferably has an ampli fication between 0 and 1 .

[0111] The further processing unit 18 is configured to convert the output audio signal 12 from the processing unit 9 into a digital output signal and to convert the distance R to the sound source 6 and / or the orientation cp with respect to the sound source 6 into a digital representation . In this exemplary embodiment , the microphone capsule 2 is a phantom-powered microphone capsule . A phantom power source (not shown) is used to power the microphone capsule 2 .

[0112] A method of operating the microphone system 1 comprises the following steps :

[0113] - capturing arriving sound 5 from a sound source 6 , in particular a person, at least over the entire frequency range between 100 Hz and 16 kHz , preferably between 20 Hz and 20 kHz , with the microphone capsule 2 , preferably a dual-diaphragm condenser capsule 3 , having the primary acoustic sensor 4 ;

[0114] - transducing the arriving sound 5 into a first transduced audio signal 7 with the primary acoustic sensor 4 ;

[0115] - measuring a distance R to the sound source 6 and / or an orientation cp with respect to the sound source 6 with at least one distance sensor 8 ;

[0116] - deriving a variable adaption gain 11 from a measured distance R to the sound source 6 and / or a measured orientation cp with respect to the sound source 6

[0117] - applying the variable adaption gain 11 to the first transduced audio signal 7 ( contained in the intermediate audio signal 16 ) , thereby compensating amplitude variations of the first transduced audio signal 7 due to distance changes and / or orientation changes of the sound source 6

[0118] - outputting an output audio signal 12 comprising the first transduced audio signal 7 .

[0119] In the embodiment shown, the method comprises the following steps :

[0120] - capturing the arriving sound 5 from the sound source 6 at least over the entire frequency range between 100 Hz and 16 kHz , preferably between 20 Hz and 20 kHz , with the secondary acoustic sensor 14 , which is preferably included into the microphone capsule 2

[0121] - transducing the arriving sound 5 into a second transduced audio signal 15 with the secondary acoustic sensor 14 ;

[0122] - applying a variable adaption gain ( in this case the variable adaption gain 11 contained in the intermediate audio signal 16 ) to the second transduced audio signal 7 , thereby compensating amplitude variations of the first transduced audio signal 7 due to distance changes and / or orientation changes of the sound source 6

[0123] - applying a first gain 17 to the first transduced audio signal 7 and a second gain 20 to the second transduced audio signal 15 ;

[0124] - outputting the output audio signal 12 comprising the first transduced audio signal 7 and / or the second transduced audio signal 15 , and in this way adapting an overall directivity of the microphone system 1 , in particular of the microphone capsule 2 .

[0125] The overall directivity of the microphone system 1 , which may also be referred to as directivity pattern 33 , may be directed to the sound source 5 and approximate a beam ( see Fig . 6D) that can be adapted in directionality . Said beam can be more directional to suppress reverberation and ambient noise 34 when the sound source is far away . The described method may be also referred to as "beam forming" .

[0126] Fig . 2 shows a block diagram of a processing unit 9 of a microphone capsule 2 with a primary acoustic sensor 4 . Compared to the embodiment of Fig . 1 , no secondary acoustic sensor 14 is provided . The processing unit 9 may be at least partially implemented in discrete electrical parts and units . The processing unit 9 is capable of performing " level balancing" and " low frequency balancing" . The implementation comprises one distance sensors 8 which measures a ( average ) distance R between the microphone system 1 and the sound source 6 and / or an ( average ) orientation cp of the microphone system 1 with respect to the sound source 6 . The orientation cp may be also referred to as direction .

[0127] From the measured distance R and orientation cp a distance between the microphone capsule 2 , and the sound source 6 and / or an orientation cp of said capsule 2 can be derived by using geometrical calculations , as the positions and arrangements of the capsules 2 are known . For this purpose , a digital controller 36 as further processing unit 18 , is provided, which in this example receives both the measured distance R and the measured orientation cp from the distance sensor 8 as well as the first transduced audio signal 7 from the primary acoustic sensor 4 . The microphone capsule 2 provides the first transduced audio signal 7 as an analogue signal . In between the microphone capsule 2 and the digital controller 36 is an analogue-to-digital (A / D) converter 50 , which is configured to convert the ( analogue ) first transduced audio signal 7 form the microphone capsule 2 in a digital signal and provide the digital signal to the digital controller 36 . The digital controller 36 may be configured as a further processing unit 18 , i . e . , to convert the output audio signal 12 from the processing unit 9 into a digital output signal and to convert the distance R to the sound source

[0128] 6 and / or the orientation cp with respect to the sound source 6 into a digital representation .

[0129] The processing unit 9 receives the first transduced audio signal

[0130] 7 from the microphone capsule 2 , i . e . , from the primary acoustic sensor 4 . The processing unit 9 applies the variable filter 26 and then the variable adaption gain 11 to the first transduced audio signal 7 in order to compensate for amplitude variations of the first transduced audio signal 7 due to distance changes and / or orientation changes of the sound source 6 . However, the roles may also be exchanged, i . e . , the variable adaption gain 11 may be applied at first and the filter 26 at second . The variable adaption gain 11 is being adapted by the digital controller 36 based on the measured distance R and / or measured orientation cp . In the embodiment shown, the variable adaption gain 11 and the variable filter 26 are depicted as separate units . However, in an alternative embodiment of the invention, the filter 26 may be included into the variable adaption gain 11 , thereby achieving a frequency dependent variable adaption gain 11 .

[0131] The variable filter 26 is a variable high-pass filter . This is advantageous with regard to " low frequency balancing" and allows for a compensation of the proximity ef fect , which primarily affects lower frequencies . A cut-of f frequency of the (high-pass ) filter 26 is adapted to the measured distance and / or orientation . The filter 26 is implemented as an analogue filter in this example . The cut-of f frequency may be , for example , adapted by changing a resistance value of a resistor of the filter 26 . The resistance may be formed by a JFET . The processing unit 9 generates the output audio signal 12 by applying the variable adaption gain 11 as well as the filter 26 to the first transduced audio signal 7 . This is done in order to compensate for amplitude variations of the first transduced audio signal 7 due to distance changes and / or orientation changes of the sound source 6 . The output audio signal 12 comprises the first ( adapted) transduced audio signal 7 .

[0132] The first transduced audio signal 7 and the output audio signal 12 are analogue signals , in particular analogue electrical signals , and the processing unit 9 is an analogue processing unit , in particular an analogue electrical processing unit . In other words , the audio signals are not discreti zed with regard to time an amplitude level within the processing unit 9 . However, the analogue processing unit 9 is controlled by the digital controller 36 of the further processing unit 18 . The variable adaption gain 11 and the filter 26 are adapted by the digital controller 36 . The variable adaption gain 11 and the filter 26 are adapted based on the measured distance R and / or orientation cp with respect to the sound source 6 .

[0133] In order to overcome unwanted fluctuations of the amplitude level of the first transduced audio signal 7 , the measured distance R between the microphone system 1 , in particular the microphone capsule 2 and the sound source 6 may be used by the digital controller 36 to automatically compensate varying amplitude level of the first transduced audio signal 7 by adj usting the variable gain 11 , such that the output audio signal 12 is consistent in amplitude relative to a desired amplitude level . The adj ustment of the variable adaption gain 11 may be based on the measured distance R . For example , i f the distance R is reduced to the 1 / 2 , the variable adaption gain 11 may be reduced by 1 / 2 . In general , the variable adaption gain 11 may be proportional to ~R . This procedure may be referred to as " level balancing" . Fig . 3 shows a block diagram of a processing unit 9 of a microphone capsule 2 with a primary acoustic sensor 4 as well as a secondary acoustic sensor 14 . The block diagram basically corresponds to the embodiment shown in Fig . 1 . The first 4 and the second acoustic sensor 14 . The first transduced audio signal 7 and the second transduced audio signal 15 are fed into a first gain 17 and a second gain 20 . By weighing the first 7 and second transduced audio signal 15 and summing them in a summation block 37 , the overall directivity of the microphone system can be changed so that reverb may be suppressed . The signal containing the weighted transduced audio signals 7 and 15 may be referred to as intermediate audio signal 16 . The intermediate audio signal 16 is further fed into the filter 26 for " low frequency balancing" and then into a variable adaption gain 11 for " level balancing" . After passing the variable adaption gain 11 , the intermediate audio signal 16 may be output as output audio signal 12 at the output 23 of the microphone system 1 . The processing unit 9 is an analogue processing unit 9 , which means that the audio signals 7 , 15 , 16 and 12 are analogue signals . The intermediate signal 16 may be converted into a digital signal by a A / D converter 50 and fed into a digital controller of a further processing unit 18 . In contrast to this , in Fig . 1 the output audio signal 12 is fed into the further processing unit . The further processing unit 18 may be configured to control the first gain 17 , the second gain 20 , the filter 26 and the variable adaption gain 11 based on distance R and / or orientation cp . The further processing unit 18 may also analyse the intermediate audio signal 16 ( or the output audio signal 12 ) and further adapt the gains 17 , 20 , the filter 26 and / or the variable adaption gain 11 to achieve further adaptions of the output audio signal 12 , such as the function of a dynamic range compressor or a de-esser . The further processing unit 18 may also analyse the intermediate audio signal 16 ( or the output audio signal 12 ) with respect to reverb and adapt further the gains 17 , 20 accordingly . The further processing unit 18 may also output a digital output audio signal 51 .

[0134] In the following, ef fects that can negatively influence the recorded sound will be described with reference to a generic microphone 28 : As sound 5 propagates from the sound source 6 to the generic microphone 28 ( see Fig . 4A-C ) , the amplitude level 29 of the sound 5 reduces essentially inversely proportionally . For this reason, when recording a sound source 6 , it is necessary for the distance R to be kept constant i f a constant amplitude level 29 of a transduced audio signal is desired . However, in practice , the distance R may vary for a number of reasons and the amplitude level 29 may also vary . This is depicted in the plots of Fig . 4A-C showing di f ferent amplitude levels 29 with respect to a desired amplitude level 30 , wherein the abscissa represents a frequency f and the ordinate represents an amplitude A. In Fig . 4A, the amplitude level 29 is higher than the desired amplitude level 30 due to the short distance R between the sound source 6 and the generic microphone 28 . In Fig . 4B, the amplitude level 29 corresponds to the desired amplitude level 30 . In Fig . 4C, the amplitude level 29 is lower than the desired amplitude level 30 due to the far distance R between the sound source 6 and the generic microphone 28 .

[0135] For audio productions in which a consistent amplitude level 29 is desired, such as speech recording or popular music, techniques can be applied in the recording studio to counteract this . Such techniques may include vocalists adj usting their microphone distance R in real time to balance baseline amplitude level variation, or the use of volume automation and / or compression to reduce amplitude level variation . For untrained users in uncontrolled environments such variation may occur and will directly result in variation of the amplitude level of a transduced audio signal .

[0136] In order to overcome unwanted fluctuations of the amplitude level 29 of the first transduced audio signal 7 , the distance R between the microphone system 1 , in particular the microphone capsule 2 and the sound source 6 may be used by the digital controller 36 to automatically compensate varying amplitude level 29 of the first transduced audio signal 7 , the second transduced audio signal 15 or a combination of the first 7 and the second transduced audio signal 15 , i . e . , the intermediate audio signal 16 , by adj usting the variable adaption gain 11 , such that the output audio signal 12 is consistent in amplitude relative to a desired amplitude level 30. The adjustment of the variable adaption gain 11 may be based on the measured distance R. For example, if the distance R is reduced to the 1 / 2, the variable adaption gain 11 may be reduced to 1 / 2 (or essentially -6 dB) . In general, the variable adaption gain 11 may be proportional to ~R. The described procedure may be referred to as "level balancing".

[0137] In close proximity to a sound source 6, known as the near-field 31, the nature of the sound wavefronts is more spherical in shape, while at greater distance, known as the far-field 32, these wavefronts are essentially planar. The transition from near to far field is proportional to ~l / f, leading to near-field behaviour existing at greater source distance as frequency f decreases. Directional microphones typically achieve directional behaviour by measuring the difference in sound pressure between two points in space, the near-field wave propagation results in greater variation in low frequency sensitivity relative to the rest of the frequency range. In practice this means that the overall spectral balance of a recording using a directional microphone is often not the same for different distances, at close distances there will be greater bass level (see Fig. 5A) and at far distances R there will be less bass level (see Fig. 5C) . In Fig. 5B, the bass level corresponds to a desired amplitude level 30.

[0138] In practice this means that a directional microphone does not have a consistent frequency response with respect to distance R, resulting in variation of bass if a sound source varies in distance R to the microphone. This can be overcome in a studio environment through controlled source positioning, selection of appropriate microphone patterns and through automated post processing techniques to ensure a consistent final balance of the sound spectrum. The end user may not have access to such tools or to be able to limit movement in their recording environment .

[0139] In order to achieve a consistent frequency response, the distance R between the microphone system 1, in particular the microphone capsule 2, and the sound source 6 may be used by the digital controller 36 to automatically compensate the variation in bass of the first transduced audio signal 7, the second transduced audio signal 15 or a combination of the first 7 and the second transduced audio signal 15, i.e., the intermediate audio signal 16, by adjusting the variable filter 26 (see Fig. 2 or 3) , such that the output audio signal 12 is consistent in amplitude relative to a desired amplitude level 30 over the entire frequency range. The adjustment of the filter 26 may be based on the measured distance R and the overall directivity. This may be referred to as "low frequency balancing".

[0140] For example, if the distance R is reduced, the filter cutoff frequency fc is proportionally increased (such that fc = 0.5*c / (2*pi*R) , where c is the speed of sound) . The damping of the filter is proportional to the directivity of the microphone system 1. For a cardioid, a first order high-pass filter may be used that doubles the signal amplitude, every time the frequency doubles, or 6dB / octave. More directional microphones, such as super-cardioid or bidirectional microphones, may require higher order filters.

[0141] Another major challenge in sound recording is minimising the impact of unwanted ambient noise 34 and reverberation. In general, ambient noise 34 and reverberation are present in all rooms. In recording studios, ambient noise 34 is reduced by controlled acoustic conditions, optimal source placement and post processing. However, these measures are typically not available for the end user.

[0142] By means of combination of the first transduced audio signal 7 of the primary acoustic sensor 4 and the second transduced audio signal 15 provided by the secondary acoustic sensor 14, it is possible to change a directivity pattern 33 of the microphone system 1. This is shown in Fig. 6A-D.

[0143] In Fig. 6A, the sound source 6, a person, is located far away from the microphone system 1 so that the output audio signal 12 will contain ambient noise 34. The directivity pattern 33 shown in Fig. 6A is the directivity pattern 33 of the primary acoustic sensor 4 without further measures . By changing the directivity pattern 33 of the microphone system 1 ( see Fig . 6B ) , thereby directing it to the sound source 6 , the microphone system 1 can be focused to the sound source 6 . It can be seen in the upper part of Fig . 6B that the ambient noise 34 could be signi ficantly reduced . A similar situation is depicted in Fig . 6C and Fig . 6D, where the sound source 6 is even farther away than in Fig . 6A and Fig . 6B . Accordingly, a more directional beam is used ( Fig . 6D) to suppress the reverb . The process of directing the directivity pattern 33 to a sound source 6 , which may be movable , can be referred to as "ambiance balancing" . By means of the distance sensor 8 , the sound source 6 may be tracked and the directivity pattern 33 directed to the tracked sound source 6 , thereby suppressing ambient noise 34 .

[0144] To adapt an overall directivity of the microphone system 1 , in particular of the microphone capsule 2 , the processing unit 9 is configured to apply a first gain 17 to the first transduced audio signal 7 and a second gain 20 to the second transduced audio signal 15 and to combine the first transduced audio signal 7 and the second transduced audio signal 15 to an intermediate audio signal 16 and further to the output audio signal 12 .

Claims

Claims :

1. Microphone system (1) , comprising:- a preferably phantom-powered microphone capsule (2) , preferably a dual-diaphragm condenser microphone capsule (3) , having a primary acoustic sensor (4) configured to capture arriving sound (5) from a sound source (6) , in particular a person, at least over the entire frequency range between 100 Hz and 16 kHz, preferably between 20 Hz and 20 kHz, and to transduce the arriving sound (5) into a first transduced audio signal ( 7 ) ;- at least one distance sensor (8) for measuring a distance (R) to the sound source (6) ;- a phantom-powered processing unit (9) configured to• compensate for amplitude variations of the first transduced audio signal (7) due to distance changes of the sound source (6) by applying a variable adaption gain (11) to the first transduced audio signal(7) , the variable adaption gain (11) being adapted to a measured distance (R) to the sound source (6) and• output an output audio signal (12) comprising the first transduced audio signal (7) .

2. Microphone system according to claim 1, wherein the variable adaption gain (11) is a frequency dependent gain, in particular a high-pass filter, that is applied to the first transduced audio signal ( 7 ) .

3. Microphone system (1) according to claim 1 or 2, wherein the first transduced audio signal (7) is an analogue signal, in particular an analogue electrical signal, and the processing unit (9) is an analogue processing unit, in particular an analogue electrical processing unit.

4. Microphone system (1) according to any one of claims 1 to 3, wherein the microphone system (1) comprises a main body (13) , in particular a housing, into which the processing unit (9) is included, wherein the microphone capsule (3) is attached to or included into the main body (13) , the microphone capsule (2) preferably being covered with a microphone grill.

5. Microphone system (1) according to claim 4, wherein the at least one distance sensor (8) is included into or attached to the main body (13) .

6. Microphone system (1) according to any one of claims 1 to 5, wherein the at least one distance sensor (8) is a time-of-f light sensor, in particular an infrared time-of-f light or ultrasonic time-of-f light sensor configured to measure the distance (R) of the sound source (6) relative to the microphone system (1) .

7. Microphone system (1) according to any one of claims 1 to 6, wherein the at least one distance sensor (8) is configured to identify an object, in particular a head, and to track the ob- j ect .

8. Microphone system (1) according to claim 7, wherein the distance sensor is configured to turn off the output audio signal (12) when a predefined object is not present and thus not identified by the at least one distance sensor (8) .

9. Microphone system (1) according to any one of claims 1 to 8, where the microphone system (1) comprises a secondary acoustic sensor (14) , which is preferably included into the microphone capsule (2) , wherein the secondary acoustic sensor (14) is configured to capture arriving sound (5) from the sound source (6) at least over the entire frequency range between 100 Hz and16 kHz, preferably between 20 Hz and 20 kHz, and to transduce the arriving sound (5) into a second transduced audio signal (15) , preferably wherein the processing unit (9) is configured to• compensate for amplitude variations of the second transduced audio signal (15) due to distance changes of the sound source (6) by applying a variable adaption gain (11) to the second transduced audio signal (15) , the variable adaption gain (11) being adapted to a measured distance (R) to the sound source (6) and• include the second transduced audio signal (15) into the output audio signal (12) .

10. Microphone system (1) according to claim 9, wherein the primary acoustic sensor (4) and the secondary acoustic sensor (14) each have a directivity, said directivities being oriented preferably essentially opposite to each other.

11. Microphone system (1) according to claim 9 or 10, wherein in order to adapt an overall directivity (33) of the microphone system (1) , in particular of the microphone capsule (2) , the processing unit (9) is configured to• apply a first gain (17) to the first transduced audio signal (7) and a second gain (20) to the second transduced audio signal (20) and• combine the first transduced audio signal (7) and the second transduced audio signal (15) to the output audio signal(12) .

12. Microphone system (1) according to any one of claims 1 to 11, characterized in by a further processing unit (18) configured to convert the distance (R) to the sound source (6) into a digital representation and to preferably convert the output audio signal (12) from the processing unit (9) into a digital output signal ( 51 ) .

13. Method of operating a microphone system (1) with the following steps:- capturing arriving sound (5) from a sound source (6) , in particular a person, at least over the entire frequency range between 100 Hz and 16 kHz, preferably between 20 Hz and 20 kHz, with a preferably phantom-powered microphone capsule (2) , preferably a dual-diaphragm condenser capsule (3) , having a primary acoustic sensor (4) ;- transducing the arriving sound (5) into a first transduced audio signal (7) with the primary acoustic sensor (4) ;- measuring a distance (R) to the sound source (6) with at least one distance sensor (8) ;- deriving a variable adaption gain (11) from a measured distance (R) to the sound source (6) ;- applying the variable adaption gain (11) of a phantom-powered processing unit (9) to the first transduced audiosignal (7) , thereby compensating amplitude variations of the first transduced audio signal (7) due to distance changes of the sound source (6) ; and- outputting an output audio signal (12) comprising the first transduced audio signal (10) .

14. The method according to claim 13, wherein the entire microphone system (1) is powered by the phantom power source.

15. The method according to any one of claims 13 or 14, with the following further steps:- capturing the arriving sound (5) from the sound source (6) at least over the entire frequency range between 100 Hz and 16 kHz, preferably between 20 Hz and 20 kHz, with a secondary acoustic sensor (14) , which is preferably included into the microphone capsule (2) ;- transducing the arriving sound (5) into a second transduced audio signal (15) with the secondary acoustic sensor (14) ;- preferably applying a variable adaption gain (11) to the second transduced audio signal (7) , thereby compensating amplitude variations of the second transduced audio signal (7) due to distance changes of the sound source (6) ;- applying a first gain (17) to the first transduced audio signal (7) and a second gain (20) to the second transduced audio signal (15) ;- outputting the output audio signal (12) comprising the first transduced audio signal (7) and / or the second transduced audio signal (15) , and in this way adapting an overall directivity of the microphone system (1) , in particular of the microphone capsule (2) .

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