WO1999014984A1

WO1999014984A1 - Improved directional microphone audio system

Info

Publication number: WO1999014984A1
Application number: PCT/US1998/019107
Authority: WO
Inventors: Matthew G. Anderson
Original assignee: Shure Brothers Incorporated
Priority date: 1997-09-16
Filing date: 1998-09-15
Publication date: 1999-03-25
Also published as: AU9315998A; EP0938830A1; US6137887A; JP2001505396A; EP0938830A4

Abstract

A multiple-microphone actuation control system using direction-sensitive microphones turns ON microphones (72A, 74A) only if a talker's speech originates from within a specified 'acceptable angle' in front of the microphones. Additionally, the invention automatically identifies which microphone best 'hears' the talker, and only turns ON one microphone per talker, while allowing several microphones to turn ON simultaneously for several talkers.

Description

IMPROVED DIRECTIONAL MICROPHONE AUDT SYSTEM

BACKGROUND OF THE INVENTION

The present invention relates to automatic microphone control systems and, more

particularly, to an enhancement of the invention disclosed in U.S. Pat. No. 4,489,442, issued

to Carl R. Anderson, et al. entitled "Sound Actuated Microphone System" and U.S. Patent

4,658,425, issued to Stephen D. Julstrom, entitled "Microphone Actuation Control System

Suitable for Teleconference Systems." U.S. Pat. Nos. 4,489,442 and 4,658,425 are both owned by the same entity as the present application.

The contents of U.S. Pat. Nos. 4,489,442 and 4,658,425 are incorporated herein by

reference, as if fully set forth below. For ease of reference, U.S. Pat. No. 4,489,442 is

hereinafter referred to simply as the "Anderson patent"; U.S. Pat No. 4,658,425 is

hereinafter referred to as "the Julstrom patent".

It is a common practice in audio engineering to use multiple microphones placed at

different locations throughout rooms such as conference rooms, classrooms, or on a stage

wherein multiple talkers voices need to be either amplified and/or recorded. In such a

system, the outputs of the microphones are usually added (combined) in an audio mixer, the

output of which might feed into an amplifier, a recording device, or a transmission link to

a remote location.

Multiple microphones are used to insure that each person's voice can be picked up

by at least one microphone at a relatively close distance to his mouth thereby helping to

insure that the audio quality, including intelligibility, is sufficient for each person. In a conference room, classroom, or on a stage, using only one microphone invariably means that

some talkers will be farther away from the microphone than others. The talkers who are far

from the microphone might not have their voices heard well above the rooms background

noise. Using multiple microphones results in a higher ratio of direct sound from the talker's

voice to room noise and reverberation at each microphone. However, the use of multiple microphones that all pick up the unwanted ambient noise and reverberation as well as the desired talker's voice creates several other problems.

The Anderson patent teaches a method and apparatus for determining if a given microphone should be turned ON or OFF by using two, back-to-back cardioid microphone

elements. If a talker's voice originates from in front of the microphone, then the signal heard

by the front-oriented microphone will be louder than the rear-oriented microphone, and the

microphone should then be turned ON.

The output signal from a cardioid microphone element can be plotted in polar

coordinates which will produce the heart-shaped graph shown in Fig. 3 of the Anderson

patent. A sound wave incident upon a cardioid microphone element at an angle theta, will

have an output level represented by the vector "S". Fig. 3 is a polar coordinate plot of the

cardioid element as a function of the angle of incidence of an acoustic wave. A wave that

impinges upon the element at 0 degrees will produce the highest possible output; a wave that

impinges upon the rear of the element, i.e. at 180 degrees, in theory, produces no output.

The combination of the polar responses of the elements with the circuitry described in the Anderson patent yields a direction-sensitive microphone which will turn ON if a sound

originates within a predetermined angle in front of the microphone; it is spatially selective.

While the invention disclosed in the Anderson patent is effective in providing spatial

selection of microphones, such spatial selection is often insufficient to avoid unwanted

detection of an audio source. When several microphones are placed side-by-side, the spatial

selectivity of the microphones is inadequate to avoid turning ON several of the microphones

if a sound source originates within the sound-sensitive space of more than one of the

microphones.

In applications where multiple microphones are required to be able to hear different

talkers, it would be desirable to be able to ignore microphones that do not best "hear" the

talker's voice.

While the Julstrom patent disclosed a circuit for comparing the outputs of several

microphones in an audio sound system and for turning ON only one microphone per talker,

the Julstrom patent does not provide any means for spatial selection of microphones; a

talker can turn ON a microphone if he is not in front of it.

Accordingly, an audio system that discriminates both on the number of ON

microphones per talker and the location or orientation of the source would be an

improvement over the prior art.

An object of the present invention is to provide an audio system that identifies if a

talker is within some predetermined location with respect to the microphone and identifies

the microphone that best hears the talker. SUMMARY OF THE INVENTION

There is provided an improved multiple-microphone audio system that identifies

which microphone of a plurality of microphones best detects an audio source. The

system employs multiple uni-directional microphones per channel and associated

circuitry to turn OFF a microphone channel for audio signals originating from sources

outside a predetermined geometric angle formed by a normal to the microphone's sensing

element. Additional signal processing evaluates output signal amplitudes from the other

microphones and detects which microphone instantaneously has the largest output signal. The largest-signal determination is logically "AND"ed with the front-of-microphone

signal amplitude test to identify the microphone that best "hears" a talker.

BRIEF DESCRIPTION OF THE DRAWINGS

Fig. 1 shows a block diagram of a multiple-microphone audio system.

Fig.2 A shows a simplified cross-sectional diagram of a uni-directional microphone employed in the preferred embodiment herein.

Fig.2B shows a simplified plot of the relative output level of the cardioid

microphone elements used in the microphone shown in Fig. 2 A as a function of an audio signal's angle of incidence upon the included microphone elements.

Fig.2C shows the two plots shown in Fig. 2B overlaid to show the difference in output signal level from the front cardioid element versus the rear cardioid element.

Fig.3A shows a functional block diagram of the preferred embodiment of the

invention.

Fig.3B shows an alternate implementation of the invention and the functional

elements of a digital signal processor implementation thereof.

Fig.3C shows an alternate implementation of the invention and the functional elements of a microprocessor implementation thereof.

DESCRIPTION OF THE PREFERRED EMBODIMENT

Figure 1 shows a multiple-microphone sound system (10) contemplated by the

embodiment described herein. A talker (12), whose voice is to be amplified or broadcast for

other distribution, is generally in front of and within the acoustic detection range of three

microphones (14, 16 and 18). As would occur in real experiences, the talker (12) is

preferably proximate to at least one of the microphones (14, 16, and 18) but in reality all three microphones "hear" the talker's voice.

Outputs from the microphones (14, 16 and 18) are input (20, 22, and 24) to

microphone mixer (26), which sums the inputs (20, 22 and 24). The mixer's output (27)

feeds an amplifier (29) which drives a loudspeaker (30). While each of the microphones (14,

16, and 18) hear the talker (12), one of the microphones will always hear the talker better

than the others. The microphone that is best located or positioned to detect the talker's voice,

is preferably the only microphone that should be enabled; its output should be the only signal

heard from the loudspeaker (30). The invention contemplated herein uses "direction-

sensitive" microphones and audio signal amplitude discrimination circuitry to selectively

amplify a talker's voice detected from the microphone that best "hears" the talker.

Direction-sensitive microphones are well-known and described in U.S. Patent No.

4,489,442, the "Anderson patent" For ease of reference, Fig. 2A shows a simplified block

diagram of a direction-sensitive microphone (50) and is prior art. In the embodiment shown in Fig. 2A, and in the Anderson patent, a housing (51)

which in the preferred embodiment is an elongated tube, has mounted within it a first

cardioid directional microphone element (54) and a second cardioid directional microphone

element (52).

It should be understood that the elongated tube (51) is constructed such that audio

waves can readily pass through it. A wire or plastic mesh or screen might support the two

microphone elements. In the preferred embodiment the tube (51) is constructed from

columnar frame members that hold the two microphone elements with the orientations shown

in Fig. 2 A. The top and bottom outlines of the tube (51) shown in Fig. 2A depict placement

of the columnar frame members that hold the directional microphone elements in place. The

microphone elements might also be supported by a plurality of rigid or semi-rigid wires

maintaining the orientation of the microphone elements inputs as shown. The front, or first,

cardioid directional microphone elements (54) has a front audio, or acoustic, input port (54A)

and a rear audio input port (54B). The rear, or second, directional microphone element (52)

also has a front audio, or acoustic, input port (52A) and a rear input acoustic port (52B).

Again, with reference to the Anderson patent, Figure 3 therein shows a polar

coordinates plot of the relative output signal level from a cardioid microphone element as a

function of an acoustic signal's angle of incidence upon the microphone. In Figure 2B, the

plot of the relative output amplitude of the first cardioid element (54) is identified by

reference numeral 64; the plot of the relative output amplitude of the second cardioid element

(52) is identified by reference numeral 66. As set forth in the Anderson patent, the cardioid elements (52 and 54) can be considered as directional elements in that their output signals

are greatest when an audio wave is incident upon the front audio input port at an angle that

is substantially normal to the plane of the front audio input port. The response of cardioid

elements is well known and the polar coordinate plot shown in Fig. 2B is also prior art.

With reference to Fig. 2A, the first and second microphone elements (52 and 54) are

mounted within the elongated tube (51) and are positioned such that the front audio input

port (54A) of the first cardioid directional microphone element (54) faces or is oriented to one end of the tube (51) that can be considered to be the front (56) of the microphone (50).

The opposite end of the tube (51) is considered the rear (58) of the direction-sensitive

microphone (50).

As set forth in the Anderson patent, audio signals incident upon the front 56 end of

the microphone (50) produce an output signal from the first microphone element (54) at its

output terminals (62) that will be substantially greater than the amplitude of the signal output

from the second microphone element (52) from its output terminals (60).

Figure 2B shows a polar plot of the output levels (64 and 66) produced by the front

or first microphone element (54) and the rear or second microphone element (52) for a given

angle of acoustic incidence, theta. Vector (65) has a length La_o,,, that represents the output

level from the front microphone element (54). Vector (67) has a length L^that represents

the output level from the rear microphone element (52). Figure 2C shows the superposition

of the plots (64 and 66) and illustrates that for a sound source positioned at the angle theta, vector (65) L^is substantially greater than vector (67) L-_. Fig.2C is also disclosed in the aforementioned Anderson patent and is also prior art.

As set forth in the Anderson patent, when the angle of incidence theta is equal to

approximately 60 degrees, the output level of the front microphone element (54) would be

approximately 9.5 decibels greater than the output level of the rear microphone element (52).

It can be seen in Figure 2A, that the first microphone element (54) and the second microphone element (52) are both directional microphone elements mounted within the

substantially elongated housing (51) which, of course, has a center axis. The angle of

incidence of audio signals is measured with respect to the center axis of the microphone

elements, which in Fig. 2A is substantially the center axis of the tube (51). In alternate

embodiments, the directional microphone elements (52 and 54) can be mounted in housings

other than tubes, such as cubes, cones, or other geometrically shaped housings. The directional microphone elements are preferably collinear and kept proximate to each other

so as to be able to accurately measure differences in audio signal amplitudes incident upon

(heard by) both elements wherever they are placed in a room. In the preferred configuration,

the rear audio input ports of the two microphone elements (54 and 52) are oriented such that

they face each other in the elongated tube (51). The front audio input ports of both

microphone elements (54 and 52) face the opposite ends of the tube (51) or other housing

containing the elements.

The unidirectional microphone apparatus shown in Figure 2A is commercially

available from Shure Brothers Incorporated in their AMS line of microphones. Of necessity, both microphone elements have output terminals (60 and 62) from

which electrical signals are produced, the amplitudes of which represent the relative

amplitude of an audio wave impinging upon and thereby detected by the microphone element

(52 and 54). In the embodiment shown in Figure 2A, the first microphone element (54) has

output terminals identified by reference numeral (62). Reference numeral (60) identifies the

output terminals of the second microphone element (52). In the preferred embodiment, these

two sets of electrical output terminals share a common ground and have a signal level from

each microphone element available on their own output line. Accordingly, there are three wires connected to the microphone (50).

The salient feature of the microphone contemplated by the invention herein is that

when audio signals impinge upon the input port (56) of the front direction sensitive microphone element at an angle substantially greater than 60 degrees, the output from the

front microphone is less than 9.5 decibels greater than the output from the rear (52)

directional microphone element. This 9.5 dB signal differential is determined by subsequent

audio signal processing circuitry to be the ratio at which the microphone's output is turned

OFF. Stated alternatively, front-to-back microphone signal differences of less than 9.5 dB

result in the audio signal not being amplified by the system. As will be seen in the

description hereinafter, the 60 degree directional sensitivity is a design choice that is

determined by the signal processing of the audio output signals from the first and second

microphone elements (54 and 52) respectively. As such, the 60-degree cutoff is a

predetermined amount of front-to-back signal differential. The output signals from the directional microphone elements (54 and 52) appear at

what can be considered front and rear output terminals (62 and 60) of the microphone (50).

Signals from these output terminals are subsequently processed by circuitry to determine the

difference in amplitude detected by the front and rear microphone elements (54 and 52). Figure 3A shows a functional block diagram of an audio signal processor that

receives the front and rear output signals from the direction-sensitive microphone shown in

Figure 1 and depicted in Figure 2A. This audio signal processor produces, as an output,

audio signals detected by the microphone (50) when the audio signal level from the first or front directional microphone element exceeds the audio signal level detected by the rear, or

second, microphone element by approximately 9.5 decibels. As set forth above, it has been

determined, and is disclosed in the Anderson patent, that when audio signals are incident

upon the microphone at an angle of 60 degrees, the front cardioid element will have an

output signal that is approximately 9.5 decibels greater than the output level of the rear

cardioid microphone element The discrimination of the front microphone element against

the rear microphone element is performed by the audio signal processing circuit (70 A) shown

in Figure 3A.

Signal output from the cardioid microphones, front microphone element (54) and rear

microphone element (52) are coupled into the audio signal processor (70A) at two inputs

thereof (72A and 74A). In the embodiment shown in Figure 3 A, input (72A) receives signals

from the front directional microphone (54) through its output terminals (62) (not shown in Figure 3A). Audio signals from the rear directional microphone element (52) from its output

terminals (60) are coupled into input (74 A) of the audio signal processing circuit (70 A).

Signals received at both inputs (72A and 74A) are pre-amplified (76 and 78) by equal

amounts to increase the levels of the signals received from the microphone's front and rear

cardioid elements to levels suitable for the subsequent circuitry. Output from pre-amplifier

(76) is coupled to a gain fader stage (80) for additional signal processing as described further

below.

Outputs from preamplifier stages (76 and 78) are then coupled into gain/bandpass

equalization stages (82 and 84) which emphasize the speech-band frequencies from the microphone elements and further amplify the signals for subsequent circuitry. These

equalized signals are fed to matching half-wave-logarithmic-rectifier and filter stages (86 and

88). The output of the half-wave-logarithmic-rectifier and filter stages(86 and 88) are

substantially DC-level signals which do vary but which fairly represent the signal level

amplitude output from the front and rear (54 and 52) cardioid microphone elements within

microphone (50). The outputs of the half-wave-logarithmic-rectifier and filter stages (86 and

88), are compared (90) to determine whether or not the signal at the front cardioid element

(54) exceeds audio detected at the rear cardioid element (52) by some predetermined amount,

i.e. 9.5 dB in the preferred embodiment and to produce a direction-sensitive microphone

control signal (92).

As a matter of design choice, the half-wave-logarithmic-rectifier and filter stages, (86

and 88), have one of their gain values adjusted. Alternatively the comparator 90, is designed such that its output goes true or active when the signal level input at input (72) exceeds that

to input (74) by approximately 9.5 decibels.

The 9.5 dB differential is a design choice and reflects the signal level detected by the

cardioid elements when an audio source is equal to 60 degrees divergence from a normal to

the front microphone element (54). As set forth in the Anderson patent, this 9.5 dB

differential is a function of the response of the cardioid microphone element and the trigger points selected by design of the audio signal processing circuitry (70A).

In effect the audio signal processing circuit (70A) produces as an output, a signal (92) that goes true, or active, when the amplitude of the output from the first or front cardioid

microphone element (54) exceeds the output from the rear or second cardioid element by a

predetermined amount. In the preferred embodiment, this predetermined amount was

determined to be 9.5 decibels. Alternate embodiments could, of course, contemplate a

greater or smaller differential to render the output of the comparator (90) true.

Figure 3A also shows a second audio signal processing circuit (70B) with inputs

(72B and 74B). In an audio system, such as that shown in Figure 1, each microphone (14,

16 and 18) would, of necessity, be connected to its own audio signal processing circuit. For

the audio system shown in Figure 1, a second audio signal processing circuit (70B) would

be connected to a second direction-sensitive microphone. The functional elements shown

within the broken line of Figure 3 A and identified by reference numeral 70A are repeated

within the signal processing circuit identified by reference numeral (70B). As set forth above, the output of the first preamplifier stage (76) is also processed and is coupled to a gain fader stage (80A) which is a simple gain stage, the output level of which

can be varied by the user to adjust the relative gain applied to the different microphones used

in the sound system shown in Figure 1. The gain stage (80A) is a variable gain stage and

simply provides a familiar fader level control for each microphone.

The output of the gain fader stage (80A) is subsequently processed by a bandpass

equalization stage (94) to emphasize speech-band frequency signals such that the circuitry

responds to speech and not extraneous room noises.. The bandpass equalization stage (94) output is rectified and filtered to produce a near-DC signal. This near-DC signal is then fed

to hysteresis gain stage (101 A). This stage adds 6 dB of gain to this signal to give a 6-dB

advantage to any microphone which is ON. This eliminates any indecision of selecting

between two microphones with similar levels. This circuit is also described in the Julstrom

patent. This scaled near-DC signal is fed to a sensing diode circuit (98). Output signals from

the rectification and filter stages (96 A and 96B) and the hysteresis gain stage (101 A and

101 B) that appear on line (99A and 99B), are a processed version of the audio input signals

detected at the front, or first cardioid microphone element (54).

With respect to audio signal processing circuit (70B), it is receiving signals from

another microphone, processing them identically, and producing corresponding signals on

its output line (99B) which signals are coupled to another sensing diode circuit (100).

Sensing diode circuits (98 and 100) are precision rectifier circuits, to greatly reduce

the .3 to .7 volt drop associated with a simple diode. The "anodes" of these circuits are coupled to ground (104) through a resistance (106). At all times, at least one of the sensing

diode circuits will be conducting. At any given instant the channel with the highest input

level, as represented by the scaled DC levels (99 A, 99B) will conduct.

In the event that signals on output lines (99A and 99B) vary in accordance with each

other, indicating that both channels are "hearing" the same signal, only one of the two

sensing diodes circuits (98) and (100) will become forward biased. The other channel's

signal level will be effectively "shadowed" by the higher signal, and its sensing diode circuit will not conduct. The voltage differential across the forward biased diode is sensed by a comparator stage (102 and 104), the output of which indicates that the audio signal it is

receiving exceeds the audio signal input to the other microphone.

Inasmuch as one diode circuit (98 or 100) will turn on when scaled signals on output

lines (99A and 99B) are greater than the other, the circuitry implemented with sensing diode

circuit (98) and comparator (102) and sensing diode circuit (104) and comparator (104) act

as a comparison circuit that produces an output that identifies which of the microphone

signals is greatest or maximum at any instant.

With respect to the output of the differential amplifier or comparator 102, its output

will go "true" on output line (106) if sensing diode circuit (98) is forward biased. Sensing

diode circuit (98) will become forward biased only if the voltage on bus 110 is less than the

voltage from the audio signal processing circuit 70 A on line 97 A. The signal on bus 110 can

be considered a max signal corresponding to the greater amplitude signal of the front

electrical signals output from each direction-sensitive microphone. Conversely, sensing diode circuit (100) will become forward biased only if the signal on line (97B) is greater than

the voltage level on the bus 110, hereafter the "max bus."

Outputs from the comparators (102 and 104) are used to gate audio switches (112 and

114) via the AND gates (122A and 122B) and the hold-up circuits (123 A and 123B). The

audio signals from the audio signal processing circuit (70A) and the max bus (110) and its

associated circuitry (80A, 94A, 96A, 98 and 102) effectively act to gate audio signals to an

output (120) only if two conditions are satisfied: the audio must originate from in front of

the microphone, as indicated by a ratio of front-element level to rear-element level, and

determined by the audio signal processing circuitry (70A) AND the signal from the same

microphone must be the largest audio signal detected by all of the microphones, as

determined by the amplitude processing circuitry (80 A, 94A, 96A and 98 and 102).

Audio signals on line (77A and 77B), which are output from the channel fader stages

(80A and 80B) are substantially the audio signals detected at the front cardioid microphone

element of microphone (50). The switches (112 and 114) are prevented from going to an ON

state unless the outputs from the audio signal processing circuits (70A and 70B) are

themselves true. Output signals (92A and 92B) are logically "AND"ed (122A and 122B)

with the outputs from the comparative circuits (102 and 104) to provide the gate or enable

signal for the switches (112 and 114) through the hold-up circuits (123A and 123B). As the

"AND"ed output signals (122A and 122B) are very impulsive, due to the impulsive nature

of speech, hold-up circuits extend the signals at lines ( 122A and 122B) to approximately .5

seconds, for two reasons: First the hold-up circuit bridges gaps in speech so that the microphone stays ON, and second, the hold-up circuit allows several microphones to turn

ON simultaneously for several talkers. This is discussed in the Julstrom patent.

Those skilled in the art will recognize that the signal processing shown in the

apparatus of Figure 3 A could be accomplished using digital signal processing techniques.

Referring to Figure 3B, there is shown a functional block diagram of digital signal processor implementing the aforementioned processes, albeit in a digital domain.

Figure 3B could be implemented using a digital signal processor, a microcontroller, a microprocessor, or other digital technology.

With respect to Figure 3B, input signals to a digital signal processor (310 A) are

received at input port (72 A and 74 A). Both of these signals are preamplified and converted

to digital signals by the preamplifier and analog-to-digital (A D) converter stages (76 and 78)

and then fed into a digital signal processor (DSP) for subsequent processing. The output of

the A/D converters can be either serial or parallel streams of data.

The digital representations of the signals from the front microphone element (54) and

the rear element (52) are then both bandpass equalized (82 and 84), rectified, converted to

logarithmic signals, and then digitally filtered (86 and 88) to produce two numbers in two

registers (301 and 302), each representing the envelope of the signals picked up from each

cardioid microphone element at any point in time. These two numbers are compared (90)

to each other on a sample-by-sample, or on a sub-sampled basis if the amplitude from the

front element (54) exceeds that from the rear element (52) by some predetermined amount.

If the amplitude from the front element (54) exceeds that from the rear element (52) by this amount, a decision is made that a talker is within the acceptance angle of the microphone and

a flag is set in register (92) indicating that this criterion has been met.

The audio signal received from the front microphone element (54) is also processed

by a gain setting routine (80A), which increases or decreases the effective data amplitude based on input from a user-adjustable control. This scaled signal is then digitally bandpass

filtered (94) as in the preferred embodiment, and then it is rectified and filtered (96), to

formulate what is a near-DC representation of the audio signals detected by the front

microphone element (54); this representation is stored in a register (97 A). This register is

then tested against all of the other channels' registers (97B) as set forth above, to compare

the output of the first microphone elements, first or front directional element to that output

from other microphones. The channel's register that is highest for a given sampling cycle

"wins" the max bus comparison, and a comparison flag (307) is set to true for that channel.

The comparison flag (307) and the register (92) are then logically "AND"ed (308) together.

If this condition is true, then the audio data from the output of gain routine (80 A) is routed

to the adder stage (112) where it is added to the other channels' signals. From here, the data

is sent to the digital-to-analog (D/A) converter (114) and converted back to an analog output

signal (120). The aforementioned routines describe one channel (310A), and these routines

can be duplicated for the second channel (310B).

Fig. 3C shows yet another alternate embodiment of the invention using a

microprocessor (212) to make gating decisions, but using analog circuitry to pass the audio

signal. In Fig. 3C, the comparison of microphone output levels is after the microphone

I S preamplifiers (76 and 78) via A/D conversion (200 and 202) to the microprocessor. The

signal from the front microphone cartridge (54) is passed through the preamplifier (76) and to the fader stage (204). The output from this fader stage is fed into a third A/D converter

(206), which provides the data for the max bus routines. The microprocessor sends a gating

control signal to audio switch (208) which feeds the audio signal to line (210) for output to

subsequent audio device in the system. All of the routines for filtering and decisions are

done in similar fashion as the DSP implementation as illustrated in fig. 3B..

Those skilled in the art will recognize that the combination of the direction-sensitive microphones, the outputs of which vary with the angle of incidence of audio signals received

by them, are capable of capturing audio signals from sources that are not directly in front of

them. As microphones recede from the talker, the talker's voice produces an increasingly

weak signal, which the microphone is not able to detect and discriminate against background

noise. An adjacent microphone, another second microphone adjacent to a talker, might pick up that talker's voice albeit with less intensity.

The audio signal processing circuits described herein, analyze the output of the

direction-sensitive microphones and amplify such outputs only if the output of the

microphone front input exceeds that from the rear input by some predetermined amount If

the directional microphone front input level is substantially greater than the rear input level,

the microphone is detecting audio that originating within some predetermined angle in front

of the microphone. Subsequent processing of the outputs of all microphones that have, or are detecting,

such audio signals are compared to identify which microphone is detecting the strongest

signal. The microphone that is detecting the strongest audio signal, and that has an audio

signal originating from in front of the direction-sensitive microphone, i.e., greater than 9.5

dB difference between the front and rear inputs, is the microphone most likely to be closest and having the loudest output of the talker.

Accordingly, by this invention, the output of one microphone is identified as having the largest amplitude for a given audio source. The output of the microphone that best hears

a source is transmitted to other audio processing equipment such as a loudspeaker, tapes or other audio distribution equipment.

Claims

CLAIMSWhat is claimed is:

1. A sound system comprising:

a first direction-sensitive microphone means having front and rear microphone

elements respectively coupled to front and rear output terminals, said first direction-sensitive

microphone means for receiving a first acoustic signal at said front microphone element and

at said rear microphone element and for producing a front electrical signal at said front

output terminal representative of the acoustic signal detected by said front microphone

element and producing a rear electrical signal at said rear output terminal representative of the acoustic signal detected by said rear microphone element;

a second direction-sensitive microphone means having front and rear microphone

elements respectively coupled to front and rear output terminals, said second direction-

sensitive microphone means for receiving a second acoustic signal at said front microphone

element and at said rear microphone element and for producing a front electrical signal at

said front output terminal representative of the acoustic signal detected by said front

microphone element and producing a rear electrical signal at said rear output terminal

representative of the acoustic signal detected by said rear microphone element;

a first audio signal processing means coupled to said front and rear output terminals

of said first direction-sensitive microphone means for producing a first direction-sensitive

microphone control signal when said front electrical signal of said first direction-sensitive microphone exceeds said rear electrical signal of said first direction-sensitive microphone by a predetermined amount;

a second audio signal processing means coupled to said front and rear output

terminals of said second direction-sensitive microphone means for producing a second

direction-sensitive microphone control signal when said front electrical signal of said second direction-sensitive microphone exceeds said rear electrical signal of said second direction- sensitive microphone by a predetermined amount;

audio signal level comparison means, coupled to said first direction-sensitive

microphone means to receive said front electrical signal of said first direction-sensitive

microphone and coupled to said second direction-sensitive microphone means to receive said

front electrical signal of said second direction-sensitive microphone, for determining which

of said front electrical signals of said first and second direction-sensitive microphones is

greater in amplitude and for producing a max signal corresponding to the greater amplitude

signal of said front electrical signals of said first direction-sensitive microphone and said

second direction-sensitive microphone and for comparing said max signal to said front

electrical signals of said first and second microphones and producing a microphone selection

signal identifying which of said first and second microphones has the largest amplitude front

electrical signal;

gating means, coupled to said audio signal level comparison means to receive said

microphone selection signal, and coupled to said first and second audio signal processing

means to receive said first and second direction-sensitive microphone control signals, for producing an audio output signal corresponding to the greater amplitude front electrical

signal detected by one of said first and second direction-sensitive microphones, according

to said microphone selection signal and said first and second direction-sensitive microphone

control signals.

2. The sound system of claim 1 where at least one of said first and second direction-

sensitive microphones are comprised of cardioid microphone elements.

3. The sound system of claim 1 where at least one of said first and second direction-

sensitive microphones are unidirectional microphones.

4. The sound system of claim 1 where at least one of said first and second direction-

sensitive microphones are Shure Brothers Inc. AMS microphones.

5. The sound system of claim 1 where at least one of said first and second said first

audio signal processing means is comprised of an audio preamplifier.

6. The sound system of claim 1 where at least one of said first and second said first

audio signal processing means is comprised of a gain bandpass equalization stage.

7. The sound system of claim 1 where at least one of said first and second said first

audio signal processing means is comprised of a logarithmic rectifier and filter stage.

8. The sound system of claim 1 where at least one of said first and second said first

audio signal processing means is comprised of a half wave logarithmic rectifier and filter

stage.

9. The sound system of claim 1 where at least one of said first and second said first audio signal processing means is comprised of a comparator stage.

10. The sound system of claim 1 wherein said audio signal level comparison means is

comprised of a bandpass equalization stage.

11. The sound system of claim 1 wherein said audio signal level comparison means is

comprised of a rectification and filter stage.

12. The sound system of claim 1 wherein said audio signal level comparison means is

comprised of a sensing diode circuit

13. The sound system of claim 1 wherein said audio signal level comparison means is comprised of a comparator.

14. The sound system of claim 1 wherein said gating means includes an audio switch.

15. The sound system of claim 1 wherein at least one of said first and said second audio

signal processing means is comprised of a digital signal processor.

16. The sound system of claim 1 wherein at least one of said first and said second audio signal processing means is comprised of a microprocessor.

17. The sound system of claim 1 wherein said audio signal level comparison means is

comprised of a digital signal processor.

18. The sound system of claim 1 wherein said audio signal level comparison means is

comprised of a microprocessor.

19. The sound system of claim 1 wherein said gating means is comprised of a digital

signal processor.

20. The sound system of claim 1 wherein said gating means is comprised of a

microprocessor.

21. In an audio system comprised of a plurality of direction-sensitive microphones that detect acoustic signals originating from an audio source, a method of selectively amplifying acoustic signals from a single direction-sensitive microphone comprising the steps of:

a) detecting an acoustic signal with front and rear microphone elements at said

plurality of direction-sensitive microphones to produce front and rear electrical

output signals from each direction-sensitive microphone of said plurality of direction-sensitive microphones;

b) processing said front and rear electric output signals from each direction-

sensitive microphone of said plurality of direction-sensitive microphones to produce

a microphone output control signal, from each of said direction-sensitive

microphones having a front electric output signal exceeding its rear electric output

signal by a predetermined amount;

c) comparing the front electrical output signals of said plurality of direction-

sensitive microphones to produce a microphone selection control signal identifying

which of said microphones has the largest amplitude front electrical signal;

d) gating a front electrical signal of one of said plurality of microphones to an

output according to said microphone selection signal and said direction-sensitive microphone control signals.

22. The method of claim 21 wherein said step d) is comprised of a logical AND of said

microphone selection signal and said direction-sensitive microphone control signals.

23. The method of claim 21 wherein said step b) of processing said front and rear electric

output signals from each direction-sensitive microphone of said plurality of direction-

sensitive microphones to produce a microphone output control signal, from each of said direction-sensitive microphones having a front electric output signal exceeding its rear

electric output signal by a predetermined amount includes the step of: preamplification of said front and rear electric output signals.

24. The method of claim 23 further including the step of:

bandpass equalizing said front and rear electric output signals of said plurality of microphones.

25. The method of claim 24 further including the step of:

logarithmically rectifying and filtering said front and rear electric output signals of

said plurality of microphones.

26. The method of claim 25 further including the step of:

comparing said front and rear electric signals of said plurality of microphones to

produce said microphone output control signal.