CN219227773U - Device for receiving sound of multiple participants - Google Patents

Abstract

Sounds of a plurality of participants are received. The device is located in a position substantially surrounded by participants and receives an audio output signal from an output interface (107). A plurality of microphone cartridges is provided wherein each microphone cartridge includes a receiving surface that most readily receives sound waves in substantially normal proximity. The housing includes a plurality of gratings (201, 202), the receiving face of each of the microphone cartridges being aligned behind a corresponding housing grating. The processing system processes the audio input signal from the microphone pod and provides an audio output signal to the output interface. The respective surfaces of each grill and its aligned microphone pod are inclined relative to the horizontal and the grills are offset from each other about a vertical axis.

Description

Device for receiving sound of multiple participants

Technical Field

The present application relates to an apparatus for receiving sound of a plurality of participants.

Background

Microphones for receiving audio signals from a plurality of participants are known. In some cases, as an ideal solution, a separate microphone may be provided for each participant. However, this approach can be cumbersome if the participant does not use a dedicated environment. In a non-dedicated environment, a mobile system may be deployed, but typically this may involve moving the device while a particular participant speaks. However, if a plurality of participants are to be received simultaneously, a problem arises, which in turn may lead to degradation of the received audio signal, thereby making it difficult to listen to or record all the participants due to the quality of the received signal.

Disclosure of Invention

According to a first aspect of the present application, there is provided an apparatus for receiving sound of a plurality of participants. Apparatus for receiving sound of a plurality of participants, comprising: a plurality of microphone cartridges (microphone capsule), wherein each of the microphone cartridges comprises a receiving face and the receiving face most readily receives sound waves substantially normally approaching the receiving face; a housing comprising a plurality of housing grills, wherein a receiving face of each of the microphone cartridges is aligned behind a respective housing grill; a processing system for processing an audio input signal from the microphone pod to provide an audio output signal to an output interface; wherein: the respective surfaces of each of the housing grills and their aligned microphone cartridges are inclined relative to a horizontal plane; and the housing grilles are offset (displaced) from each other about a vertical axis.

Drawings

Embodiments of the present application will now be described, by way of example only, with reference to the accompanying drawings, in which:

FIG. 1 illustrates an apparatus for receiving sound of a plurality of participants;

fig. 2 shows a grill extending from the housing of the microphone shown in fig. 1;

FIG. 3 shows the response of a heart shape;

fig. 4 shows a processing system comprised in the microphone shown in fig. 1;

FIG. 5 shows a device connected to a USB cable;

FIG. 6 shows the application of manual pressure on the device;

fig. 7 shows muting of a device;

FIG. 8 illustrates an operational mode;

FIG. 9 illustrates a functional schematic of operations performed by the processing system shown in FIG. 4;

FIG. 10 illustrates a group (group) mode of operation;

FIG. 11 illustrates the processing system of FIG. 4 configured to respond to a selected single sub-mode (solo sub mode);

FIG. 12 illustrates the processing system of FIG. 4 configured to operate in a vocal sub-mode (vocal sub-mode);

FIG. 13 illustrates the processing system of FIG. 4 configured as a music sub-mode (music sub-mode);

FIG. 14 illustrates the processing system of FIG. 4 modified to operate in an immersive mode (immerse mode);

fig. 15 shows the frequency response of the equalizer shown in fig. 10; and

fig. 16 shows in detail a method of operating the device.

Detailed Description

Fig. 1 shows an apparatus for receiving sound of a plurality of participants. The device comprises a plurality of microphone cartridges (including a first cartridge 101 and a second cartridge 102). Each box is housed behind a protective grille, which is regularly staggered about the central vertical axis 103.

There may be fewer than nine gratings of this type in practice, and the embodiment of fig. 1 has four gratings which are offset from each other by 90 degrees about the central vertical axis 103. Thus, the example of fig. 1 includes a third microphone pod located at the position indicated by arrow 104 and a fourth microphone pod located at the position indicated by arrow 105. The grille will be further described with reference to fig. 2.

As further described with reference to fig. 3, each microphone pod (e.g., 101, 102, etc.) includes a receiving surface that most readily receives sound waves approaching in a substantially normal direction.

A processing system as further described with reference to fig. 4-6 is mounted on the circuit board 106 and is configured to process audio input signals received from the microphone pod to provide audio output signals to the output interface; the embodiment of fig. 1 provides a micro USB interface 107. In one embodiment, the input signal is also returned to the USB interface 107.

The respective surfaces of each grill and its aligned microphone pod (101, 102) as described in detail with reference to fig. 2 are inclined with respect to the horizontal, the grills being offset from each other about a central vertical axis 103.

The embodiment of fig. 1 includes a silicone base 108 and may also include threaded holes compatible with photography tripods and the like. The rigid microphone frame 109 supports respective slings of resilient material, including the first slings 110 of each case, to provide mechanical decoupling.

The top portion 111 comprises a second circuit board for supporting a touch sensitive light emitting screen covered by an acrylic light diffuser, and a polycarbonate top that can carry visual indications and possibly logo-type graphics to the user. In one embodiment, the microphone pod has a diameter of 9.5 mm.

The housing for the microphone comprises a plurality of protective grilles (including a first grill 201 and a second grill 202). The base 108 is rotated 180 degrees about the central vertical axis 103 compared to the orientation shown in fig. 1, such that the third microphone pod 104 is behind the first grill 201 and the fourth microphone pod 105 is behind the second grill 202. The first grill 201 comprises a cross section 205 followed by a socket 206 for receiving a headphone plug.

Each housing grid 201, 202 and the shielded third and fourth grids 203, 204 are inclined with respect to the horizontal plane. In one embodiment, the tilt angle 207 is greater than 30 degrees and may be less than 60 degrees. In the embodiment of fig. 2, the inclination angle of each housing grid is substantially 45 degrees with respect to the horizontal.

In the embodiment of fig. 2, the inclined housing grid extends from a substantially flat base 108 defined as a regular shape to a smaller top portion 111 of the same regular shape. Thus, in the embodiment of fig. 2, four inclined grids are proposed in view of this embodiment, these regular shapes being square. However, it will be appreciated that other shapes are possible, such as five gratings exhibiting pentagons, six gratings exhibiting hexagons, etc.

The substantially planar top portion 111 includes a manual receiving input means for providing control data to the processing system and a visual indication means for displaying output data from the processing system. The input means comprise a first manually responsive input means 211 shown as a minus sign and a second input means 212 shown as a plus sign, which is mainly used for adjusting the volume. The volume of the output signal provided to the connected headphones is indicated by the first output indicator 221, and the color changing display 222 indicates the mode of operation. The color changing display 222 is also responsive to manual input.

In one embodiment, the microphone cartridges 101, 102, etc. have a generally heart-shaped response as shown in fig. 3. In fig. 3 a first microphone capsule 101 is schematically shown, having a receiving surface 301 aligned with its corresponding grill 203. As indicated by arrow 302, the receiving surface 301 most readily receives sound waves that are generally normally proximate to the surface.

A polar pattern (polar pattern) 303 is shown which indicates the sensitivity of the microphone pod to sound from different directions. The cardioid polar pattern shows the highest sensitivity to sound coming from directly in front of the microphone (0 degrees) and little sensitivity to sound coming from directly behind (180 degrees) and reduced sensitivity to sound coming from the side (90/270 degrees). In this way, the microphone is most sensitive to receiving speech from a user directly in front of the corresponding inclined surface, the angle of inclination improving the preferred directional axis (as indicated by arrow 302).

When the user is arranged around the microphones, the individual microphones receive different sounds, thereby facilitating a mixing operation which allows the user to be separated in the audio field by generating a suitably mixed stereo signal. In addition, all four signals may be provided separately to a recording device or telephone device (typically a notebook computer), if desired.

As shown in fig. 4, the processing system 401 is configured to process the audio signal from the microphone pod to provide an audio output signal to the output interface 107. Processing system 401 may be implemented substantially around an XU212 processor manufactured by Xmos limited, uk, which includes a plurality of processing cores and memory for storing executable instructions.

Although the device is primarily concerned with receiving the sound of a plurality of participants, embodiments may also be configured to receive the sound of a single participant, such that the normal mode is identified as group mode and when the sound of a single participant is received is identified as single person mode. When operating in a single person mode, the individual participant should be aligned with the tilted first grill 201 (indicated by the position of the headphone jack 204) considering that the individual participant may be wearing headphones. Thus, as shown at 105L, 102R, 104F, and 101B of fig. 4, microphone 104 is identified as front, microphone 102 is identified as right, microphone 105 is identified as left, and microphone 101 is identified as rear.

The processing system 401 receives four input signals from respective microphone capsules via respective microphone preamplifiers, soft clipping circuitry, and analog-to-digital converters. Thus, the output from the first microphone box 101B is provided to the first preamplifier 411, the first soft clipping circuit 421 and the first analog-to-digital converter 431, and similar means are provided for the other three inputs.

In one embodiment, analog-to-digital converters (431, 432, 433, and 434) are implemented using devices provided by texas instruments, each individual device accepting two analog inputs, requiring two physical devices of this type. Microphone preamplifiers (411, 412, 413, 414) ensure that the sensitivity of the analog signal at the respective analog-to-digital converter 431 to 434 is optimized. The corresponding soft clipping circuits 431 to 434 prevent overload of the analog-to-digital converters 431 to 434.

The amplifier gains of the preamplifiers 411 to 414 are controlled by the processing system 401, in combination with the soft clipping circuitry, again ensuring that the analog to digital converter does not receive an overvoltage that would cause undesirable distortion (undesirable distortion). As will be described with reference to fig. 8 to 15, the control line 415 supplies control signals from the processing system 401 to the amplifiers 411 to 414 depending on the operation mode.

The output signal is supplied to an external device through the USB cable 441. The external device may be an audio recorder, a video recorder, an amplifier, a public address system, a telecommunication system or a computer. The external device indicated at 442 may be implemented, for example, as a notebook computer.

The USB interface 107 receives the stereo output signal and provides a left channel 443L and a right channel 443R. Further, the USB interface 107 may receive separate outputs from a single microphone cartridge (shown as 444L, 444R, 444F, and 444B).

The USB interface 107 is also configured to receive input signals from the external device 442, which may also be in the form of stereo pairs implemented as left signal 445L and right signal 445R. These signals are supplied to the headphone jack 204 via the first digital-to-analog converter 446L and the second digital-to-analog converter 442R.

The device receives power from an external device 442 via a USB cable 441. To activate the device, a USB plug 501 is inserted into the USB interface 107. Upon receipt of power, color changing display 222 lights up and displays a color representing the mode used when the device was previously deactivated.

Typically, when receiving audio from multiple participants, the microphone is placed substantially centrally, allowing it to pick up speech from all participants present without the need to move again.

As shown in fig. 6, the color changing display 222 is responsive to manually applied pressure. When operating in the selected mode, the color-changing display screen 222 will display a fixed (solid) green, white, magenta, or cyan (when operating in stereo mode). When operating in the immersive mode, the display will flash between one of the colors and yellow. Pressing the color changing display as shown in fig. 6, the microphone will mute and the display will display red.

The headset may be plugged into the headset receptacle 206. The volume of the signal provided to the headphones may be adjusted by applying pressure to the first manual response input device 211 to decrease the volume, or pressure may be applied to the second manual response input device 212 to increase the volume of the signal provided to the headphones. The first output indicator 221 also reflects increasing or decreasing the volume by changing the brightness, which becomes brighter when a signal of a greater volume is provided, and darker when the volume is decreased. If the second manual response input device is repeatedly operated such that the volume is reduced to zero, the first output indicator 221 will also display a fixed red color.

As shown in fig. 7, the setting mode can be selected by simultaneously pressing the first manual response input device 211 and the second manual response input device 212 for more than two seconds. In the setup mode, pressing the color changing display screen 222, as described with reference to fig. 6, may select a sub-mode in the stereo mode and the immersive mode.

In the setup mode, switching between the stereo mode and the immersive mode is achieved by pressing the color changing display screen 222 for more than 5 seconds. If nothing is pressed for more than 20 seconds, the device will return to normal mode.

Fig. 8 shows an operation mode. The possibility of muting depicted in fig. 6 is illustrated at 801. When not muted, the color-changing display screen 222 displays the operation color of green, white, magenta, or cyan, but enters a mute state after the operation described in fig. 7 is performed, the display screen turns red, thereby confirming to the participant that the microphone has been muted.

As described with reference to fig. 6, maintaining pressure on color-changing display screen 222 for more than 5 seconds when in the set mode results in switching between stereo mode 802 and immersive mode 803. While in the stereo mode 802, in the set mode, the color-changing display screen 222 is pressurized such that the sub-mode (or operation mode) cycles between the group sub-mode 802G, the single sub-mode 802S, the human voice sub-mode 802V, and the music sub-mode 802M.

While in group sub-mode 802G, display 222 will flash between green and black, and further clicking will select single person sub-mode 802S. In the single person sub-mode 802S, the display 222 will flash between white and black. Further application of pressure on the display screen 222 will select the voice sub-mode 802V and the display screen 222 will flash between magenta and black. Again, applying less than two seconds of pressure on the display 222 will select the music sub-mode 802M and the display will flash between cyan and black. A similar operation would then return the selected sub-mode to group sub-mode 802G. Thus, cycling of sub-mode selection will allow the participant to select the desired mode of operation.

For a group of participants in a conversation, group sub-mode 802G is most appropriate. After selection, pressure is again applied to the first manual response input device 211 and the second manual response input device 212 to return the apparatus to its operational mode during which a non-blinking, fixed color (group sub-mode being a fixed green color) is displayed on the color changing display screen 222.

In the stereo mode, after the combining operation, a stereo signal is received through the left channel 443L and the right channel 443R.

In the set mode, when pressure is applied on color-changing display screen 222 for more than 5 seconds, an immersive mode is selected from which similar sub-modes can be selected, such as 803G for a group, 803S for a single person, 803V for a human voice, 803M for music. In the immersive mode, the sub-modes are substantially similar to those described with respect to the stereo mode 802, but for the group sub-mode, the display screen does not flash between green and black, but between green and yellow. Similarly, for the single person sub-mode, flicker between white and yellow (instead of white and black), while for the human voice sub-mode, flicker between magenta and yellow. Also, for the music sub-mode, flicker between cyan and yellow, rather than between cyan and black.

As previously described, brief pressure on the display 222 will cause the sub-mode selection cycle to proceed until the desired sub-mode is selected. When the pressure remains for more than 5 seconds, the mode selection returns to stereo mode. Applying pressure to both devices 211 and 212 returns the apparatus to its operational mode. Alternatively, if nothing is pressed for more than 20 seconds, the device will return to this (normal) mode of operation.

Fig. 9 shows a functional schematic of operations performed by the processing system 401. The system comprises a first set of switches 901 and a second set of switches 902. As shown in fig. 9, these switches are normally closed when the system is running, but open when a mute condition is selected (as described with reference to fig. 8).

Fig. 10 shows the operation of the group sub-mode in the stereo mode, when mute switches 901 and 902 are closed. Similarly, mute switches 901 and 902 are closed in the operation of the single person sub-mode described with reference to fig. 11, the operation of the human voice sub-mode described with reference to fig. 12, the operation of the music sub-mode described with reference to fig. 13, and the operation of the immersive sub-mode described with reference to fig. 14.

In all modes of operation, digital input signals are received from analog-to-digital converters 431 through 434. The input signals are initially processed by respective equalizers 1001, 1002, 1003, and 1004. Providing a first type of equalization when in group sub-mode and single sub-mode, and performing a second type of equalization when in human voice sub-mode or music sub-mode; further details of which will be described with reference to fig. 15.

An output signal is provided on control line 415 which is used to control the preamplifiers 411 to 414. In group, single person and person sub-modes, the maximum sound pressure level allowed is 104dB. In the music mode, the maximum sound pressure level allowed is 120dB by the control of line 415.

A first signal path 1011 connects the outputs of the equalizers 1001 to 1004 to a stereo output 443. A second signal path 1012 connects the outputs of the equalizers 1001 to 1004 to the four immersive outputs 444. In stereo mode 802, no immersive output is required, and the switches in the third set of switches 1013 are open (as shown in fig. 10 and 11-13).

The system includes a fourth switch 1014, a fifth pair of switches 1015, and a sixth pair of switches 1016. In group sub-mode, the fourth switch 1014 is in an up position to receive the output from the auto-mixer 1017, which in turn receives all four outputs from the respective equalizers 1001-1004. The fifth pair of switches 1015 are in an up position to provide stereo signals to the respective gates (gates) 1017L and 1017R. If the gate 1017L/1017R receives a signal below a predetermined threshold, the gate is closed such that the signal is completely blocked and no signal is present at the output. Such gating is performed under the assumption that a signal below a predetermined level is only from among the background noise, whereby the background noise is effectively blocked.

In one embodiment, the gating elements are used for group sub-mode (green) and single person sub-mode (white). The door has an open threshold of-42 dB, a closed threshold of-48 dB, an attenuation of 4.5dB, a start-up time (attack) of 5ms, a hold time of 100ms, and a release time of 200ms.

The sixth pair of switches 1616 is also in an upward position so that the output of the gate 1017L/1017R is provided to the stereo compression circuit 1018. For each channel, circuit 1018 includes a compressor 1019L/1019R followed by a compensation gain amplifier 1020L/1020R. The compressor 1019 attenuates high signal levels. The compensation gain amplifier then amplifies all signals (most pronounced in amplifying low level signals). Thus, the low level signal appears louder and the high level signal appears quieter, with the overall dynamic range being compressed.

In one embodiment, the compressor is set in group sub-mode with a threshold of-12 dB, a ratio of 4:1, a knee point (knee) of 12dB, a start-up duration of 5ms, and a release duration of 100ms.

Fig. 11 shows a processing system 401 configured according to a selected single person sub-mode 802S. Here, the fourth switch 1014 is placed in a lower position such that only the input 432 from the preceding microphone is processed through the first signal path. The control signal on control line 415 remains unchanged, the maximum sound pressure level allowed is 104dB, and no modification is made to the second equalizer 1002. However, in one embodiment, the compression circuit 1018 has a threshold of-22 dB, a ratio of 4:1, a knee of 22dB, a start-up duration of 5ms, and a release duration of 60ms. A stereo signal is then provided at stereo output 443.

In the single person mode, a single user is likely to wear headphones. A loop back signal is derived from the output 443 and provided to each of the amplifiers including the first amplifier 1101 and the second amplifier 1102 to control the level of the loop back. The return signals (comprised of left signal 445L and right signal 445R) received at 445 from external device 442 are combined with the loopback signals in first combiner 1111 and second combiner 1112. The volume of the earphone is then controlled by the third and fourth amplifiers 1123, 1124, which in turn are provided to digital-to- analog converters 446L, 446R.

Fig. 12 shows processing system 401 configured to operate in a vocal submode 802V. Inputs 431 through 434 are received from all four microphones and stereo output is produced at 443. In this way, the signal is processed again through the first signal path 1011. In this case, however, the fifth pair of switches 1015 has been placed in a lower position. This effectively disables the auto-mixer 1017 while utilizing the seventh and eighth amplifiers 1207 and 1208 and the third and fourth combiners 1213 and 1214.

The output from the front microphone 432 is amplified by the seventh amplifier 1207, and the output from the rear microphone 431 is amplified by the eighth amplifier 1208. The third and fourth combiners combine the signals from the left and right microphones 434, 433 to form a mixture, which is then provided to the compression circuit 1018.

In one embodiment, the compression circuit deploys side chain compression to reduce tooth pitch (sibilance). The threshold is set at-28 dB, the ratio is 1.7:1 (the compression of the introduced dynamic range is much smaller), the inflection point is 28dB, the starting time is 5ms, and the releasing time is 100ms.

As described with reference to fig. 15, the output on the control line 415 remains unchanged, but the characteristics of the equalizers 1001 to 1004 change.

The voice sub-mode 802V modifies the characteristics of the microphones so that they are more suitable for singing sounds than speaking sounds. In this way, devices that are primarily used to receive speech may also be deployed in more creative environments.

It will also be appreciated that in the vocal mode and the music mode (described with reference to fig. 13), there may be a relatively quiet channel, so the sixth pair of switches 1016 moves to a lower position, thereby disengaging the gate 1017 from the circuit.

The processing system 401 as shown in fig. 13 is configured as a music sub-mode 802M. From the switching point of view, the topology is substantially the same as described with reference to fig. 12. However, a new control signal is provided on control line 415 to increase the maximum sound pressure level to 120Db. The response of the equalizers 1001 to 1004 is also substantially the same as the response specified for the human phonon mode.

In one embodiment, the compression circuit 1018 is changed. The threshold is set at-22 bD, with a ratio of 10:1, thereby introducing significantly more compression for the dynamic range. The inflection point is 22dB, the starting time is 5ms, and the releasing time is 60ms.

Fig. 14 illustrates a processing system 401 modified according to a selected immersive mode 803. Where the third group switch 1013 is placed in a lower position to be output at 444 through the second signal path 1012. In the immersive mode 803, the following four sub-modes may be selected: group sub-mode 803G, single sub-mode 803S, human voice sub-mode 803V, and music sub-mode 803M.

Although the switching operation within the first signal path 1011 has no effect, other modifications previously described continue to be implanted as described with reference to fig. 10 to 13.

Thus, when a sub-mode within immersive mode 803 is selected, the signal on control line 415 changes, equalizers 431 through 434 change, and compression circuit 1018 changes.

As shown in fig. 15, characteristics of the equalizers 1001 to 1004 may be expressed as frequency responses. In this embodiment, a first frequency response 1501 is selected for the group sub-mode and the single sub-mode. A second frequency response 1502 is selected for the human phonon mode and the musical sub-mode.

The first frequency response 1501 provides a +6dB low shelf (low shell) 1503 at 200Hz with a Q value of 0.8. Unlike a low pass filter, the low stage 1503 boosts the signal but has a flat response. Followed by a boost (boost) 1504 at 13kHz with a q value of 0.71. The frequency response may be followed by a high shell.

These frequency responses are intended to optimize the intelligibility (intellisitivity) of the received speech, but in view of their main purpose to compensate for the nonlinearities of the microphone cartridges 101 to 104, the actual nature of the compensation required will be determined empirically.

In the second frequency response 1502, there is a similar low shelf 1513 at 1.4kHz followed by a peak 1514, an attenuation 1515, and another higher peak 1516 at 12.3 kHz.

Therefore, when a solo sound is received or music of an ensemble is received, an artistic balance level is introduced in addition to the compensation balance. This effectively increases the low and high frequencies while attenuating the intermediate frequency.

The described apparatus facilitates deployment of a method of receiving sound from multiple participants. Initially, the device is located in a position substantially surrounded by participants; the need to move the microphone when a particular participant speaks is reduced. An audio signal is received from the output interface 107 and provided to the external device 442. A plurality of microphone cartridges is provided, each microphone cartridge including a receiving face that most readily receives sound waves that are substantially normally accessed.

The housing includes a plurality of grills with the receiving face of each microphone pod aligned behind the grills. The processing system processes the audio input signal received from the microphone pod to provide an audio output signal to the output interface. The respective surfaces of each grill and its aligned microphone pod are inclined relative to the horizontal and the grills are offset from each other about a vertical axis.

In operation, the USB plug 501 is inserted and then initialized in step 1601. In response to manual interaction as previously described, a mode selection is made in step 1602, such that the device is then operational.

In a preferred embodiment, processing system 401 operates in response to executing stored instructions, which may depend on a degree of processing performed on a single central processing unit in a multi-tasking environment, or in a preferred embodiment, a degree of multiprocessing (multiprocessing) to provide substantially real-time signal processing.

In step 1603, the input signal is subjected to equalization processing, and in step 1604, it is queried whether another input signal is present. When the answer is affirmative, the next input signal is equalized and when all inputs are considered, the question asked in step 1604 will be answered in the negative. In the previously described embodiment, a total of four input signals are received.

In step 1605, the maximum input level is controlled and an output control signal is sent on line 415 to analog amplifiers 411-414.

In step 1606, gate control and compression are performed, and in step 1607, an output is provided to USB interface 107 via first signal path 1011 or via second signal path 1012.

If desired, the earphone output is processed in step 1608, including further amplifying (1101, 1102) the mix (111, 112) and further amplifying the mix (1123, 1124). The process may then receive further input data for mode selection or for processing further audio samples.

In one embodiment, the equalization step 1603 may compensate for the frequency response of the microphone and may also help remove unwanted sounds. Equalization may also be used to enhance musical performance when selecting either a vocal sub-mode or a musical sub-mode. In one embodiment, the individual microphone output signals are combined to form a stereo mix. In addition, when recorded, the received earphone signal and the local loop may be combined with an external signal received from a remote source (at the time of the conference) or in the form of a loop back signal at the time of recording.

Claims (12)

1. An apparatus for receiving sound of a plurality of participants, comprising:

a plurality of microphone cartridges, wherein each of the microphone cartridges includes a receiving face that most readily receives sound waves substantially normally approaching the receiving face;

a housing comprising a plurality of housing grills, wherein a receiving face of each of the microphone cartridges is aligned behind a respective housing grill;

a processing system for processing an audio input signal from the microphone pod to provide an audio output signal to an output interface; wherein:

the respective surfaces of each of the housing grills and their aligned microphone cartridges are inclined relative to a horizontal plane; and

the housing grids are offset from each other about a vertical axis.

2. The apparatus of claim 1, wherein the microphone pod has a heart-shaped response.

3. The apparatus of claim 1, further comprising a frame for supporting the microphone cartridges, wherein each of the microphone cartridges is supported in the frame by an elastic material.

4. The apparatus of claim 1, comprising more than two and less than nine housing grids.

5. The apparatus of claim 4, comprising four housing grids offset from each other by 90 degrees about the vertical axis.

6. The apparatus of claim 1, wherein each of the housing grills is inclined at an angle greater than 30 degrees relative to the horizontal plane.

7. The apparatus of claim 1, wherein each of the housing grills is inclined at an angle of less than 60 degrees relative to the horizontal plane.

8. The apparatus of claim 1, wherein each of the housing grills is inclined at an angle of approximately 45 degrees relative to the horizontal plane.

9. The apparatus of claim 1, wherein the inclined housing grid extends from a substantially flat base defined as a regular shape to a smaller substantially flat top of the same regular shape.

10. The apparatus of claim 9, wherein the regular shape is square.

11. The apparatus of claim 9, wherein the substantially flat top comprises:

a manual receiving input device for providing control data to the processing system; and

visual indication means for displaying output data from said processing system.

12. The apparatus of claim 1, wherein the processing system comprises an analog-to-digital converter for digitizing the audio input signal such that the audio output signal is a digital signal.

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