Patent Publication Number: US-9906633-B2

Title: Desktop speakerphone

Description:
TECHNICAL FIELD 
     The present invention relates to a desktop speakerphone. 
     BACKGROUND ART 
     U.S. Pat. No. 5,121,426 discloses a teleconferencing unit with an upwardly aimed loudspeaker and multiple gradient microphones arranged evenly around the loudspeaker. Each microphone has a polar response pattern with a major lobe. The loudspeaker is located in a null adjacent to the major lobe of each microphone. This reduces acoustic coupling between the loudspeaker and the respective gradient microphones. In one embodiment, the speakerphone has four first-order gradient microphones each having a supercardioid polar response pattern. The nulls are aimed at 125° with respect to the main lobe directions. In another embodiment, the speakerphone has six first-order gradient microphones pairwise electrically connected to form three reversible second-order gradient microphones, each having nulls at 90° and 180°. 
     The first- and second-order gradient microphones disclosed in the above patent are relatively expensive to manufacture, which makes the disclosed teleconferencing unit relatively expensive as well. In addition, the achievable reduction of acoustic coupling between the loudspeaker and the gradient microphones is limited due to manufacturing tolerances and changing acoustic behavior of the room. Furthermore, the optimum shape of the disclosed teleconferencing unit depends on the desired directional characteristics of the microphones. Also, the disclosed microphones have a relatively low signal-to-noise ratio (SNR) at lower frequencies. 
     DISCLOSURE OF INVENTION 
     It is an object of the present invention to provide an improved desktop speakerphone without disadvantages of prior art speakerphones. It is a further object to provide a desktop speakerphone that is relatively inexpensive to manufacture. It is a still further object to provide a desktop speakerphone with few constraints on the design of its physical appearance. It is a still further object to provide a desktop speakerphone that provides high-quality sound. 
     These and other objects of the invention are achieved by the invention defined in the independent claims and further explained in the following description. Further objects of the invention are achieved by embodiments defined in the dependent claims and in the detailed description of the invention. 
     Within this document, the term “speakerphone” refers to an audio communication device that can be connected directly or indirectly to an audio communication network and that allows a local party comprising a plurality of party members (users) to simultaneously communicate orally with one or more remote parties via the audio communication network. A speakerphone generally comprises an acoustic input device configured to pick up voices of local party members and an acoustic output device configured to provide an acoustic output signal simultaneously to a plurality of the local party members. An acoustic input device generally comprises one or more acoustic input transducers, such as one or more microphones, and an acoustic output device generally comprises one or more acoustic output transducers, such as one or more loudspeakers or sound drivers. A plurality of local party members may thus simultaneously use a speakerphone as an audio interface to an audio communication network. The above definition includes such speakerphones that comprise circuitry, e.g. landline telephone circuitry, mobile phone circuitry or computer circuitry, which enable the speakerphone to connect directly to an audio communication network, as well as such speakerphones that do not comprise such circuitry and therefore require the use of gateway devices, e.g. landline telephones, mobile phones or personal computers, for connecting to audio communication networks. 
     A “desktop speakerphone” refers to a speakerphone that is configured to be arranged and used in a stable operating position on a horizontal desktop. Where orientations or directions in space, such as e.g. “vertical”, “horizontal”, “up”, “down”, etc., are mentioned herein without further specification, such orientations and directions shall be read as referring to a desktop speakerphone arranged in its operating position for normal use on a horizontal desktop. 
     Furthermore, when an element or entity is referred to as being “connected” or “coupled” to another element or entity, this includes direct connection (or coupling) as well as connection (or coupling) via intervening elements or entities, unless expressly stated otherwise. Also, unless expressly stated otherwise, when a signal is referred to as being “provided” by a first entity to a second entity, this includes directly or indirectly transmitting the signal in its original form as well as any direct or indirect transmission that modifies the original signal and/or converts the signal into another domain and/or representation before it arrives at the second entity, provided that the information comprised by the signal received by the second entity is sufficient for the second entity to perform the specified actions with respect to the signal. 
     Within this document, the singular forms “a”, “an”, and “the” are intended to include the plural forms as well (i.e. to have the meaning “at least one”), unless expressly stated otherwise. Correspondingly, the terms “has”, “includes”, “comprises”, “having”, “including” and “comprising” specify the presence of respective features, operations, elements and/or components, but do not preclude the presence or addition of further entities. The term “and/or” generally includes any and all combinations of one or more of the associated items. The steps or operations of any method disclosed herein need not be performed in the exact order disclosed, unless expressly stated so. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       The invention will be explained in more detail below in connection with preferred embodiments and with reference to the drawings in which: 
         FIG. 1  shows an embodiment of a desktop speakerphone according to the invention, 
         FIG. 2  shows a block diagram of the desktop speakerphone of  FIG. 1 , 
         FIG. 3  shows details of an output path shown in  FIG. 2 , 
         FIG. 4  shows details of an input path shown in  FIG. 2 , 
         FIG. 5  shows details of a cluster input processor shown in  FIG. 4 , 
         FIG. 6  shows details of a high-frequency array processor shown in  FIG. 5 , 
         FIG. 7  shows a frequency-domain block diagram of the high-frequency array processor shown in  FIG. 6 , and 
         FIG. 8  shows details of a filter controller shown in  FIGS. 6 and 7 . 
     
    
    
     The figures are schematic and simplified for clarity, and they just show details essential to understanding the invention, while other details may be left out. Where practical, like reference numerals and/or names are used for identical or corresponding parts. 
     MODE(S) FOR CARRYING OUT THE INVENTION 
     The desktop speakerphone  1  shown in a top view in  FIGS. 1 a  and 1 n    a section-like side view in  FIG. 1 b    comprises a housing  2  with a support surface  3 . The housing  2  has a shape generally as an elongate disc, and the support surface  3  is located at one of the main surfaces of the elongated disk, so that the support surface  3  can support the desktop speakerphone  1  in a stable operating position on a horizontal surface, such as e.g. a desktop  4 . The desktop speakerphone  1  further comprises an upwardly directed sound driver  5  mounted centrally at the upper side of the housing  2 , so that the sound driver  5  can emit speaker sound A e  to multiple users of the desktop speakerphone  1  simultaneously. The desktop speakerphone  1  further comprises two microphone clusters  6 ,  7  mounted at the upper side of the housing  2  closer towards respective longitudinal ends  8  of the latter, so that each microphone cluster  6 ,  7  can receive voice sound A v  from one or more of the users. Each microphone cluster  6 ,  7  comprises three pressure microphones  10 ,  11 ,  12 , each fluidly connected to receive voice sound A v  from the environment through a respective sound inlet  13 ,  14 ,  15  arranged at the housing  2 . 
     An imaginary center line  16  is defined so that it extends perpendicularly to the support surface  3  through the acoustic center  17  of the sound driver  5 . For each microphone cluster  6 ,  7 , an imaginary median plane  18  is defined so that it comprises the center line  16  and further extends through the first sound inlet  13  of the respective microphone cluster  6 ,  7 . In the desktop speakerphone  1  shown in  FIG. 1 , the sound inlets  13 ,  14 ,  15  of the first microphone cluster  6  are arranged symmetrically to the corresponding sound inlets  13 ,  14 ,  15  of the second microphone cluster  7  with respect to the center line  16 , and the median planes  18  for the two microphone clusters  6 ,  7  therefore coincide in space and further are rotationally symmetric with respect to the center line  16 . With the desktop speakerphone  1  placed in its operating position on a horizontal surface  4 , both the center line  16  and the median planes  18  extend vertically. 
     All sound inlets  13 ,  14 ,  15  are arranged at equal distance from the support surface  3 , i.e. in the same horizontal plane when the desktop speakerphone  1  is in its operating position. Furthermore, within each microphone cluster  6 ,  7 , the second and third sound inlets  14 ,  15  are arranged symmetrically on opposite sides of the respective median plane  18 . Within each microphone cluster  6 ,  7 , the first and second microphones  10 ,  11  constitute a first microphone pair  10 ,  11 , while the first and third microphones  10 ,  12  constitute a second microphone pair  10 ,  12 . 
     Within each microphone cluster  6 ,  7 , the relative arrangement of the three sound inlets  13 ,  14 ,  15  defines a respective microphone axis  9 ,  19  for each of the microphone pairs  10 ,  11 ,  10 ,  12 . The microphone axis  9  of the first microphone pair  10 ,  11  extends through the first and the second sound inlet  13 ,  14 , while the microphone axis  19  of the second microphone pair  10 ,  12  extends through the first and the third sound inlet  13 ,  15 . The three sound inlets  13 ,  14 ,  15  are arranged such that the first and the second microphone axes  9 ,  19  are perpendicular to each other and so that each of the first and the second microphone axis  9 ,  19  has an angle α of 45° with the median plane  18 . The first sound inlet  13  is arranged with a larger distance to the center line  16  than each of the second and third sound inlets  14 ,  15 . 
     In the block diagram in  FIG. 2 , the desktop speakerphone  1  is shown connected to an audio communication network  20  through a gateway device  21 . The gateway device  21  serves as an interface between the desktop speakerphone  1  and the audio communication network  20 , such that the desktop speakerphone  1  may receive an audio output signal A o  from the audio communication network  20  and provide an audio input signal A i  to the audio communication network  20 . The gateway device  21  may convey, convert and/or adapt any of the audio output signal A o  and the audio input signal A i , and may further provide call signaling and/or other control functions, as known from prior art gateway devices, in order to enable users of the desktop speakerphone  1  to communicate orally with remote parties through the audio communication network  20 . In some embodiments, a gateway device  21 , such as e.g. a desktop telephone, a mobile phone, a personal computer with a softphone, or the like, may be comprised by the desktop speakerphone  1 . In some embodiments, the desktop speakerphone  1  may be directly connectable to an audio communication network  20 . 
     The desktop speakerphone  1  comprises a transceiver  22  that through a bidirectional connection  23  receives the audio output signal A o  from the audio communication network  20  and/or the gateway device  21 , transmits the audio input signal A i  to the audio communication network  20  and/or the gateway device  21  and further handles control functions associated therewith as known from prior art speakerphones. The desktop speakerphone  1  further comprises an output path  24  that provides a driver signal A d  to the sound driver  5  in dependence on the audio output signal A o  that is received through the transceiver  22 . The sound driver  5  emits speaker sound A e  to the environment in dependence on the driver signal A d . The desktop speakerphone  1  further comprises an input path  25  that provides the audio input signal A i  through the transceiver  22  in dependence on microphone signals A m  received from the microphones  10 ,  11 ,  12  of the two microphone clusters  6 ,  7 , which provide the microphones signals A m  in response to voice sound A v  received from the environment through the respective sound inlets  13 ,  14 ,  15 . The input path  25  further receives the audio output signal A o  from the transceiver  22  for use in acoustic feedback reduction and a level-controlled signal A u  from the output path  24  for use in noise reduction as explained further below. The desktop speakerphone  1  further comprises a rechargeable battery or other suitable power supply  26  for supplying electric energy to components of the desktop speakerphone  1 , such as e.g. the transceiver  22 , the output path  24  and the input path  25 . The transceiver  22  may be implemented as a wired or as a wireless transceiver and may further be implemented to connect with the audio communication network  20  and/or the gateway device  21  through an analog connection  23  or preferably a digital connection  23 , such as e.g. a Bluetooth connection, an IrDA connection, a DECT connection or a USB connection. 
     As shown in  FIG. 3 , the output path  24  comprises an emphasis filter  31 , a volume control  32 , a limiter  33 , a digital-to-analog converter  34  and a power amplifier  35  connected in series to receive the audio output signal A o , modify the audio output signal A o  and provide the modified signal as the driver signal A d . The emphasis filter  31  applies a frequency-dependent gain to the audio output signal A o  to emphasize frequency regions important for the understanding of speech and/or to compensate, at least partly, for frequency dependencies in the audio communication network  20 , the gateway device  21  and/or the sound driver  5 . The volume control  32  applies a variable output gain to the filtered signal to provide the level-controlled signal A u . The volume control  32  controls the output gain in dependence on a volume control signal C v  received from a user interface  36  and indicating user input detected by the user interface  36 . The limiter  33  applies a frequency-dependent level compression, level attenuation and/or level limitation to the level-controlled signal A u , to prevent the sound driver  5  from emitting too loud sound A e , such as sound A e  with unpleasant or harmful sound pressure levels. The digital-to-analog converter  34  converts the limited signal into an analog signal that is amplified by the power amplifier  35  to provide the driver signal A d . 
     As shown in  FIG. 4 , the input path  25  comprises, for each of the two microphone clusters  6 ,  7 , a cluster input processor  41  that provides two beam signals A b  in dependence on the microphone signals A m  received from the microphones  10 ,  11 ,  12  of the respective microphone cluster  6 ,  7  as well as on the audio output signal A o  and the level-controlled signal A u . The input path  25  further comprises a speech detector  42 , a speech level normalizer  43  and a beam selector  44 . The speech detector  42  receives the beam signals A b  from the cluster input processors  41 , for each beam signal A b  estimates whether or not voice signals are present in the respective beam signal A b  and provides a speech detection signal C s  comprising an indication of the result of this estimation. The speech detector  42  further estimates the levels of voice signals present in the beam signals A b  and provides in the speech detection signal C s  an indication of the estimated speech levels. The speech level normalizer  43  receives the beam signals A b  from the cluster input processors  41  and the speech detection signal C s  from the speech detector  42 , applies an individual beam gain to each beam signal A b  to provide a respective normalized signal A n  and controls the individual beam gains in dependence on the speech levels indicated in the speech detection signal C s  such that differences in speech levels between the normalized signals A n  are reduced compared to differences in speech levels between the beam signals A b . The speech level normalizer  43  may e.g. increase the level of beam signals A b  with lower speech levels and/or decrease the level of beam signals A b  with higher speech levels among the estimated speech levels. The beam selector  44  receives the normalized signals A n  from the speech level normalizer  43  as well as the speech detection signal C s  from the speech detector  42 , selects a preferred signal among the normalized signals A n  in dependence on the speech levels indicated in the speech detection signal C s , such that the preferred signal corresponds to the beam signal A b  having the higher speech level among the estimated speech levels, and provides the preferred signal as the audio input signal A i . 
     As shown in  FIG. 5 , each cluster input processor  41  comprises two high-frequency array processors  51 ,  52 , a low-frequency array processor  53 , two high-pass filters  54 , a low-pass filter  55 , two adders  56  and two residual-echo cancellers  57 . 
     The first high-frequency array processor  51  provides a first array signal A a1  in dependence on a first pair of microphone signals A m1 , A m2  from a first microphone array  10 ,  11 , which comprises the first microphone  10  and the second microphone  11 , and in further dependence on the audio output signal A o . The second high-frequency array processor  52  provides a second array signal A a2  in dependence on a second pair of microphone signals A m1 , A m3  from a second microphone array  10 ,  12 , which comprises the first microphone  10  and the third microphone  12 , and in further dependence on the audio output signal A o . 
     For ease of reading, the following will be adhered to in the following text: The sound inlet  13  of the first microphone  10  will be referred to as a front sound inlet, while the sound inlets  14 ,  15  of the second and the third microphones  11 ,  12  will be referred to as rear sound inlets. Correspondingly, the first microphone  10  will be referred to as a front microphone, while the second and the third microphones  11 ,  12  will be referred to as rear microphones. Also, the microphone signal A m1  from the first microphone  10 , which is received by both high-frequency array processors  51 ,  52 , will be referred to as a front microphone signal, while the microphone signals A m1 , A m2  from the second and the third microphones  11 ,  12 , which is each received by only one of the high-frequency array processors  51 ,  52 , will be referred to as rear microphone signals. Also, for each microphone array  10 ,  11 ,  10 ,  12 , the direction from the respective rear sound inlet  14 ,  15  along the respective microphone axis  9 ,  19  towards the front sound inlet  13  will be referred to as the front direction. 
     Thus, each high-frequency array processor  51 ,  52  receives a front microphone signal A m1  as well as a respective one of the rear microphone signals A m2 , A m3  and provides a respective one of the first and the second array signal A a1 , A a2  in dependence hereon. As explained in further detail further below, each high-frequency array processor  51 ,  52  controls the directivity pattern of the respective array signal A a1 , A a2  such that the directivity pattern has a main lobe generally oriented towards the front direction of the respective microphone array  10 ,  11 ,  10 ,  12  and such that the directivity pattern further exhibits reduced sensitivity towards the sound driver  5 . 
     The first microphone signal A m1  provided by the first microphone  10  is used for providing both the first and the second array signal A a1 , A a2 , which may make the desktop speakerphone  1  less space-consuming and less expensive to manufacture than prior art speakerphones. Also, the use of pressure microphones, i.e. omnidirectional microphones, may make the desktop speakerphone  1  less expensive to manufacture than prior art speakerphones and may further provide greater versatility with respect to the over-all design of the housing  2  of the desktop speakerphone  1  without compromising the effectiveness of the directional microphone system  6 ,  7 . 
     The low-frequency array processor  53  provides a mainly non-directional array signal A a3  by adding the microphone signals A m1 , A m2 , A m3  from all of the three microphones  10 ,  11 ,  12 , which thus form a third microphone array. The non-directionality is achieved through in-phase adding of the microphone signals A m1 , A m2 , A m3  and subsequent low-pass filtering in the low-pass filter  55  (see below). 
     The two high-pass filters  54  each receives and high-pass filters a respective one of the first and the second array signal A a1 , A a2  to provide a respective high-pass filtered signal A f1 , A f2 . The low-pass filter  55  receives and low-pass filters the mainly non-directional array signal A a3  to provide a low-pass filtered signal A f3 . Each of the two adders  56  receives a respective one of the high-pass filtered signals A f1 , A f2  as well as the low-pass filtered signal A f3  and adds the respective high-pass filtered signal A f1 , A f2  to the low-pass filtered signal A f3  to provide a respective combined array signal A c1 , A c2 . Each of the two residual-echo cancellers  57  receives a respective one of the combined array signals A c1 , A c2 , the front microphone signal A m1 , the audio output signal A o  as well as the level-controlled signal A u  from the output path  24  and provides a respective beam signal A b1 , A b2  in dependence hereon. 
     Each residual-echo canceller  57  may employ any know method for cancelling or otherwise suppressing residual feedback from the sound driver  5  in the respective beam signal A b1 , A b2 . One such known method is based on processing the respective combined array signal A c1 , A c2  in multiple frequency bands and attenuating the combined array signal A c1 , A c2  in those frequency bands wherein its signal level correlates with the signal level of the audio output signal A o  in the same frequency band. 
     As shown in  FIG. 6 , each high-frequency array processor  51 ,  52  comprises a controllable filter  61 , a subtractor  62 , an equalizer  63  and a filter controller  64 . The controllable filter  61  receives the rear microphone signal A m2 , A m3  from the respective microphone array  10 ,  11 ,  10 ,  12 , filters the rear microphone signal A m2 , A m3  using a first set of filter coefficients C w  received from the filter controller  64  and provides the filtered signal A w  to the subtractor  62 . The subtractor  62  subtracts the filtered signal A w  from the front microphone signal A m1  and provides the resulting difference signal A z  to the equalizer  63 . The equalizer  63  filters the difference signal A z  using a second set of filter coefficients C q  to provide an equalized signal A q . The main purpose of the equalizer  63  is to compensate for some of the level distortion caused by the subtractor  62 . The equalizer  63  is preferably configured for a reference situation wherein the front microphone  10  and the rear microphone  11 ,  12  solely receive voice sound A v  from a user located at a reference location in the far field and in the front direction of the respective microphone array  10 ,  11 ,  10 ,  12 . The second set of filter coefficients C q  may thus be fixed at design or production time and may preferably be configured to reduce or minimize, within one or more predefined frequency ranges, the level difference between the equalized signal A q  and the front microphone signal A m1  in the reference situation. The high-frequency array processor  51 ,  52  provides the equalized signal A q  as the respective array signal A a1 , A a2 . Each array signal A a1 , A a2  thus constitutes an output signal of a differential microphone array  10 ,  11 ,  10 ,  12  comprising a front microphone  10  and a respective rear microphone  11 ,  12 . 
     The filter controller  64  receives the front microphone signal A m1 , the rear microphone signal A m2 , A m3  as well as the audio output signal A o  and adaptively determines the first set of filter coefficients C w  such that in the array signal A a1 , A a2 , sound A e  emitted by the sound driver  5  is suppressed or attenuated relative to voice sound A v  arriving from the front direction of the microphone array  10 ,  11 ,  10 ,  12 . The filter controller  64  thus controls the directivity pattern of the microphone array  10 ,  11 ,  10 ,  12  such that the directivity pattern has reduced sensitivity towards the sound driver  5 , at least when compared to the sensitivity in the front direction, preferably also when compared to the average sensitivity across all directions. 
     The filter controller  64  preferably determines the first set of filter coefficients C w  according to an adaptation algorithm that provides a reduction in the coherence between the array signal A a1 , A a2  and the audio output signal A o  under the constraint that voice sound A v  received from the front direction is substantially maintained in the array signal A a1 , A a2 . Thus, the directivity pattern of the microphone array  10 ,  11 ,  10 ,  12  is adaptively controlled to reduce acoustic feedback from the sound driver  5  in the array signal A a1 , A a2  and thus also in the audio input signal A i . Numerous such adaptation algorithms are known from the prior art and may be used for this purpose. Preferred algorithms are described in the following. 
     The block diagram shown in  FIG. 7  is substantially a frequency-domain version of  FIG. 6 . Thus, the rear microphone spectrum Sr is the frequency spectrum of the rear microphone signal A m2 , A m3 , the front microphone spectrum Sf is the frequency spectrum of the front microphone signal A m1 , the difference spectrum Sz is the frequency spectrum of the difference signal A z  from the subtractor  62 , the equalized spectrum Sq is the frequency spectrum of the equalized signal A q —and of the array signal A a1 , A a2  provided by the high-frequency array processor  51 ,  52 , and the audio output spectrum So is the frequency spectrum of the audio output signal A o . The transfer function W is the transfer function of the controllable filter  61 , and the transfer function Q is the transfer function of the equalizer  63 . In addition to the front microphone signal A m1 , also the difference signal A z  from the subtractor  62  is provided to the filter controller  64 . As will be understood from the following description, the filter controller  64  may determine the first set of filter coefficients C w  in dependence on any of these signals. 
     In the shown embodiment of the high-frequency array processor  51 ,  52 , the equalized spectrum, i.e. the spectrum of the of the array signal A a1 , A a2 , thus equals:
 
 Sq=Q·Sz=Q ·( Sf−W·Sr )  (1)
 
     The sound A e  emitted by the sound driver  5  will be received by each of the front and the rear microphone  10 ,  11 ,  12  and will thus also appear in the front and the rear microphone spectrum Sf, Sr. In the following, the portion of the front microphone spectrum Sf that originates from the sound driver  5  is referred to as Sfe, the portion of the rear microphone spectrum Sr that originates from the sound driver  5  is referred to as Sre, and the portion of the difference spectrum Sz that originates from the sound driver  5  is referred to as Sze. Applying equation (1), the portion of the equalized spectrum Sq that originates from the sound driver  5  thus equals:
 
 Sqe=Q·Sze=Q ·( Sfe−W·Sre )  (2)
 
     Acoustic feedback in the array signal A a1 , A a2  may therefore be reduced or eliminated by controlling W such that Sqe is reduced, ideally to zero. The latter may be achieved by controlling W according to:
 
 W=Sfe/Sre   (3)
 
provided that Sre does not contain any spectral zeroes.
 
     The sound A e  emitted by the sound driver  5  is derived from the audio output signal A o , and thus, equation (3) can be expanded to:
 
 W =( Sfe/So )/( Sre/So )= Hfo/Hro   (4)
 
wherein Hfo and Hro are the transfer functions from the audio output signal A o  to respectively the front microphone signal A m1  and the rear microphone signal A m2 , A m3 . In the general case wherein a signal y dependent on another signal x is contaminated by noise uncorrelated to the other signal x, the transfer function Hyx from x to y may be estimated as:
 
 Hyx= Pyx / Pxx     (5)
 
wherein  Pxx  is the average auto-power spectrum of x and  Pyx  is the average cross-power spectrum of x and y. Assuming that the sound A e  emitted by the sound driver  5  is not correlated with the voice sound A v , equation (4)/(5) may thus be further expanded to:
 
 W=Hfo/Hro =(   Pfo / Poo   )/(   Pro / Poo   )=   Pfo / Pro     (6)
 
wherein  Pfo  is the average cross-power spectrum of the audio output signal A o  and the front microphone signal A m1 ,  Pro  is the average cross-power spectrum of the audio output signal A o  and the rear microphone signal A m2 , A m3 , and  Poo  is the average auto-power spectrum of the audio output signal A o .
 
     The filter controller  64  may thus preferably repeatedly perform a cross-power analysis based on the audio output signal A o , the front microphone signal A m1  and the rear microphone signal A m2 , A m3  and determine the transfer function W of the controllable filter  61  in dependence on the result of the cross-power analysis. The filter controller  64  may e.g. repeatedly estimate the average cross-power spectrum  Pfo  of the audio output signal A o  and the front microphone signal A m1  as well as the average cross-power spectrum  Pro  of the audio output signal A o  and the rear microphone signal A m2 , A m3  and determine the transfer function W of the controllable filter  61  in dependence on a quotient between the two estimated average cross-power spectra  Pfo ,  Pro , e.g. according to equation (6). 
     The filter controller  64  may preferably repeat the determination of the transfer function W of the controllable filter  61  at a rate fast enough to ensure that typically encountered changes in the acoustic path between the sound driver  5  and the microphones  10 ,  11 ,  12  do not cause artifacts in the audio input signal A i . Such changes may occur e.g. when users relocate or reorient the desktop speakerphone  1 , or when users move themselves, their hands or other objects in the vicinity of the desktop speakerphone  1 . This adaptation of the transfer function W may enable the desktop speakerphone  1  to provide a more robust suppression of acoustic feedback from the sound driver  5  compared to prior art speakerphones. The adaptation may be made at different speeds dependent on the intended use scenarios for a particular desktop speakerphone  1 . The filter controller  64  may e.g. repeat the determination of the transfer function W of the controllable filter  61  once per frame or less frequently. Within the present document, the term “frame” bears the meaning it commonly has in connection with frequency-domain signals, namely a set of frequency bin values provided in a single step of converting a time-domain signal into a frequency-domain signal. 
     In a more robust embodiment, the filter controller  64  may iteratively determine the transfer function W of the controllable filter  61  by repeatedly determining and applying a frequency-dependent adjustment term dW to the transfer function W to counteract acoustic feedback in the difference signal A z . An advantage of this approach is that the filter controller  64  may halt or slow down the adaptation of the transfer function W when adverse conditions for adaptation prevail, e.g. when local users speak, when the transfer function W is close to its optimum value and/or when Sre does contain spectral zeroes. Also, where or when the adaptation of W is to be made less frequently than once per frame, this may be achieved simply by setting the adjustment term dW equal to zero for intermediate frames, i.e. frames for which no adaptation shall be made. 
     The filter controller  64  may preferably determine the transfer function W according to:
 
 W   k+1   =W   k   +U   k   ·dW   k   (7)
 
wherein the index k represents the current frame number of the involved frequency-domain signals, W k  is the current value of the transfer function W, W k+1  is the subsequent value of the transfer function W, dW k  is the adjustment term, and U k  is a frequency-dependent moderation factor between 0 and 1. The filter controller  64  may preferably determine the adjustment term dW k  such that if it were applied in the current frame, the portion Sze of the difference spectrum Sz that originates from the sound driver  5  would become zero. This value of the adjustment term dW k  may be derived from equation (2). First, applying frame indices k to equation (2) and omitting the effect of the equalizer  63  yields:
 
 Sze   k   =Sfe   k   −W   k   ·Sre   k   (8)
 
     Inserting the adjustment term dW k  and the condition that Sze k  be zero into equation (8) yields:
 
0= Sfe   k −( W   k   +dW   k )· Sre   k   (9)
 
     Solving the equation set (8) (9) for the adjustment term dW k  yields:
 
 dW   k   =Sze   k   /Sre   k   (10)
 
which following the reasoning further above from equation (3) through equation (6) may be expanded to:
 
 dW   k =   Pzo   k   /   Pro   k     (11)
 
wherein  Pzo k    is the current value of the average cross-power spectrum of the audio output signal A o  and the difference signal A z  and  Pro k    is the current value of the average cross-power spectrum of the audio output signal A o  and the rear microphone signal A m2 , A m3 .
 
     As shown in  FIG. 8 , the filter controller  64  may comprise a first spectral analyzer  81  that repeatedly estimates the average cross-power spectrum  Pzo k    of the audio output signal A o  and the difference signal A z , a second spectral analyzer  82  that repeatedly estimates the average cross-power spectrum  Pro k    of the audio output signal A o  and the rear microphone signal A m2 , A m3 , an adjustment controller  83  that repeatedly determines the adjustment term dW, preferably in dependence on a quotient between the two estimated cross-power spectra  Pzo k   ,  Pro k   , e.g. according to equation (11), a filter estimator  84  that repeatedly determines the transfer function W in dependence on the adjustment term dW, e.g. according to equation (7), and a converter  85  that repeatedly determines the first set of filter coefficients C w  in dependence on the determined transfer function W, e.g. by Inverse Fast Fourier Transformation (IFFT), such that the transfer function of the controllable filter  61  becomes equal to the determined transfer function W. 
     It may be difficult to prevent the sound driver  5  from exitating spurious resonances in the housing  2  and other mechanical structures of the speakerphone  1 . Such spurious resonances may cause substantial changes in the sound field surrounding the speakerphone  1  and thus also affect the microphone signals A m  and eventually the determination of the transfer function W. Since such resonances are not correlated with the voice sound S v , the filter controller  64  may treat the disturbances as feedback from the sound driver  5  and thus cause the transfer function W to deviate from its optimum. Spurious resonances may thus indirectly cause audible artefacts in the audio input signal A i  provided to the audio communication network  20 , in particular with a fast adaptation of the transfer function W. The filter controller  64  may preferably apply a spectral-domain low-pass filter function G to the determined transfer function W to reduce the effect of such spurious resonances. The spectral-domain low-pass filter function G acts to reduce differences between neighboring bins in the determined transfer function W. In other words, the spectral-domain low-pass filter function G smoothes the spectral shape of the transfer function W. The smoothing reduces the influence of narrow-band excursions in the spectrum of the acoustic feedback path from the sound driver  5  to the microphones  10 ,  11 ,  12 , and since such narrow-band excursions are typically caused by resonances, this may generally improve the sound quality perceived by a remote party and/or allow for applying a faster adaptation of the transfer function W without deteriorating the sound quality. 
     The filter controller  64  may preferably apply the spectral-domain low-pass filter function G according to:
 
 W   k+1   =G ( W   k   +U   k   ·dW   k )  (12)
 
which is a modified version of equation (7). Alternatively, the filter controller  64  may apply the spectral-domain low-pass filter function G according to:
 
 W   k+1   =W   k   =G ( U   k   ·dW   k )  (13)
 
such that the spectral-domain low-pass filter function G works on the moderated adjustment term U k ·dW k .
 
     The filter estimator  84  may thus comprise a spectral-domain low-pass filter  86  that operates to reduce differences between neighboring bins in the determined transfer function W. The spectral-domain low-pass filter  86  may e.g. be configured to apply the spectral-domain low-pass filter function G by passing a sliding average window across the spectrum of each instance of the determined transfer function W and/or each instance of the moderated adjustment term U k ·dW k . Instead of a sliding average window, the spectral-domain low-pass filter  86  may apply one or more other suitable filters selected among low-pass filters generally known in the art. 
     The filter estimator  84  may preferably adaptively determine the moderation factor U k  in a manner that favors reliable values of the adjustment term dW k  over unreliable values, e.g. as described in further detail below. 
     The reliability of the adjustment term dW k  generally decreases when the amount of acoustic feedback from the sound driver  5  in the microphone signals A f , A r  decreases relative to other signals, which typically is the case when local users speak. The filter estimator  84  may thus preferably adaptively monitor at least one of the microphone signals A f , A r  and increase the moderation factor U k  in frequency bins wherein acoustic feedback from the sound driver  5  in a monitored microphone signal A f , A r  increases relative to other signals and adaptively decrease the moderation factor U k  in frequency bins wherein acoustic feedback from the sound driver  5  in the monitored microphone signal A f , A r  decreases relative to other signals. To achieve this, the filter estimator  84  may e.g. determine a frequency-dependent coherence Cmo between the audio output signal A o  and one of the front and the rear microphone signal A f , A r  and determine the moderation factor U k  in dependence on the determined coherence Cmo. For each frequency bin, the coherence Cmo approaches 1 when acoustic feedback from the sound driver  5  dominates the respective microphone signal A f , A r  and drops towards 0 when other signals are mixed into the microphone signal A f , A r . The above approach may thus result in improved values of the transfer function W and thus in increased reduction of acoustic feedback in the audio input signal A i . 
     The reliability of the adjustment term dW k  further generally decreases when the amount of acoustic feedback from the sound driver  5  in the difference signal A z  decreases relative to other signals, which typically is the case when the transfer function W is close to optimum. The filter estimator  84  may thus preferably, additionally or alternatively, adaptively increase the moderation factor U k  in frequency bins wherein acoustic feedback from the sound driver  5  in the difference signal A z  increases relative to other signals and adaptively decrease the moderation factor U k  in frequency bins wherein acoustic feedback from the sound driver  5  in the difference signal A z  decreases relative to other signals. To achieve this, the filter estimator  84  may e.g. determine a frequency-dependent coherence Czo between the audio output signal A o  and the difference signal A z  and determine the moderation factor U k  in dependence on the determined coherence Czo. For each frequency bin, the coherence Czo approaches 1 when acoustic feedback from the sound driver  5  dominates the difference signal A z  and drops towards 0 when other signals are mixed into the microphone signal A z . The above approach may thus result in improved values of the transfer function W and thus in increased reduction of acoustic feedback in the audio input signal A i . 
     The filter estimator  84  may preferably repeatedly determine the moderation factor U k  in dependence on the coherence Cmo between the audio output signal A o  and one of the front and the rear microphone signal A f , A r  as well as in dependence on the coherence Czo between the audio output signal A o  and the difference signal A z , e.g. according to:
 
 U   k   =Cmo   k ·( Czo   k +α)/(1+β)  (14)
 
wherein the index k is the current frame number, Cmo k  is the current value of the frequency-dependent coherence Cmo between the audio output signal A o  and one of the front and the rear microphone signal A f , A r , Czo k  is the current value of the frequency-dependent coherence Czo between the audio output signal A o  and the difference signal A z , and β is a small, non-zero, non-negative convergence term that may prevent the adaptation of the transfer function W to stop prematurely when approaching the optimum.
 
     In other embodiments, the filter estimator  84  may apply variants of equation (14). For instance, the convergence term β may be set to zero and/or the factor Cmo k  may be set to unity. In other embodiments, the filter estimator  84  may apply other, preferably similar functions for computing the moderation factor U k . 
     The filter controller  64  is preferably further configured to determine the transfer function W in a manner that is robust against spectral zeroes in the portion Sre of the rear microphone spectrum Sr that originates from the sound driver  5 . This may e.g. be achieved by configuring the second spectral analyzer  82  to enforce a lower limit on the individual bin values of the average cross-power spectrum  Pro k    of the audio output signal A o  and the rear microphone signal A m2 , A m3 . 
     In the desktop speakerphone  1 , the transceiver  22  preferably exchanges the audio output signal A o  and the audio input signal A i  in digital form with the audio communication network  20  and/or the gateway device  21 , e.g. through a USB connection or a Bluetooth connection. Also, the output path  24  and the input path  25  are preferably configured as digital circuits operating on digital signals, possibly except for portions thereof that interface to the sound driver  5  and/or the microphones  10 ,  11 ,  12 . Also, the output path  24  and the input path  25  are preferably configured to operate on spectral signals, in particular in order to facilitate the adaptation of the transfer function W. Most portions of the transceiver  22 , the output path  24  and the input path  25  may, however, alternatively or additionally be configured to operate on time-domain signals and/or as analog circuits operating on analog signals. Accordingly, the transceiver  22 , the output path  24  and/or the input path  25  may comprise any number of signal domain converters, i.e. analog-to-digital, digital-to-analog, time-to-spectral-domain (FFT) and/or spectral-to-time-domain (IFFT) converters, as well as any number of signal encoders and/or signal decoders to perform any required signal conversions, signal encoding and/or signal decoding. 
     Functional blocks of digital circuits may be implemented in hardware, firmware or software, or any combination hereof. Digital circuits may perform the functions of multiple functional blocks in parallel and/or in interleaved sequence, and functional blocks may distributed in any suitable way among multiple hardware units, such as e.g. signal processors, microcontrollers and other integrated circuits. 
     The detailed description given herein and the specific examples indicating preferred embodiments of the invention are intended to enable a person skilled in the art to practice the invention and should thus be seen mainly as an illustration of the invention. The person skilled in the art will be able to readily contemplate further applications of the present invention as well as advantageous changes and modifications from this description without deviating from the scope of the invention. Any such changes or modifications mentioned herein are meant to be non-limiting for the scope of the invention. 
     Examples of further changes or modifications include: the desktop speakerphone  1  may comprise further sound drivers  5 , the housing  2  may have various shapes, the sound driver  5  may be mounted off-center with respect to the housing  2 , the number of microphone clusters  6 ,  7  may be e.g. 1, 3, 4, 5 or 6 and the input path  25  may be modified accordingly, the sound inlets  13 ,  14 ,  15  of multiple microphone clusters  6 ,  7  may be arranged asymmetrically, the output path  24  and/or the input path  25  may comprise further functional blocks known from prior art speakerphones, such as e.g. decoders, audio filters, circulators and the like, the emphasis filter  31 , the volume control  32  and/or the limiter  33  may be omitted, the user interface  36  may be omitted or arranged remotely, e.g. in a gateway device  21 , the speech detector  42 , the speech level normalizer  43  and/or the beam selector  44  may be omitted, the beam selector  44  may employ other or further criteria for selecting the preferred signal, the low-frequency array processor  53  and the low-pass filter  55  may be omitted, the residual-echo cancellers  57  may be omitted, the subtractor  62  may be replaced with an adder if the filtered signal A w  and the front microphone signal A m1  have opposite phases, etc. 
     The invention is not limited to the embodiments disclosed herein, and the invention may be embodied in other ways within the subject-matter defined in the following claims. As an example, features of the described embodiments may be combined arbitrarily, e.g. in order to adapt the devices according to the invention to specific requirements. 
     Any reference numerals and names in the claims are intended to be non-limiting for their scope.