Beamed Linear Array Microphone System

The invention provides a beamed linear array microphone system for the acoustic pickup of voice and music from substantial distances with a relatively narrow pickup-beam and with the avoidance of acoustical feedback. Response of the system within the pick-up beam is relatively constant over the several normal sound octaves, and response outside of the beam is relatively low, with the acceptance angle of the beam likewise being relatively constant over the several normal sound octaves.

The system of the invention has particular utility for sound enhancement in auditoriums, studios, music halls and other facilities. The linear microphone array of the system may be mounted on or near the ceiling of the auditorium at a remote position from the stage, or other source of sound. An important feature of the system is that it provides a means for sound reinforcement throughout the hall without acoustical feedback.

There are a variety of microphone arrays which are capable of providing a narrow beam response, including, for example, planar or circular arrays. However, a linear "end-fire" array is presently preferred in the system of the invention because of its economy of microphone elements and associated electronics.

FIG. 1 is a somewhat schematic representation of a side view of a typical auditorium with a microphone array installed for inclusion in the system of the invention in one of its embodiments;

FIG. 2A is a schematic representation showing the microphone element spacings along a single linear axis in a partial array in the system of the invention and which responds to various frequency ranges throughout the normal sound octaves to be sensed and amplified by the system;

FIG. 2B is a schematic representation of the microphone element spacings in half an array in the system of the invention, and with the microphones for the various frequency ranges being displaced from one another by predetermined distances, and the microphones in the different sub-arrays or different frequency ranges being shown as displaced from the normal linear axis of the array, only for purposes of illustration and description it being understood that all the microphones in the array are positioned on a single linear axis;

FIG. 3 is a block diagram of a linear microphone array system in accordance with one embodiment of the invention;

FIG. 4 is a graphic representation representing the distances of the various sub-array microphone elements in the system from the sound source;

FIG. 5A is a curve representing the response characteristics of the linear microphone array in the system;

FIG. 5B are further curves representing the linear microphone array response characteristics; and

FIGS. 6, 7A, 7B and 8 are further curves representing the response characteristics of the linear microphone array in the system of the invention.

As shown in FIG. 1, the linear microphone array 10 of the invention is mounted adjacent to the ceiling 12 of an auditorium 14; and the array points toward a stage 16, or other sound source. The array 10, in appearance, will be similar to a long rod, with its sound pick-up beam in line with the longitudinal axis of the rod. No appreciable sound response exists to the rear of the rod, or at any angle to the rod out of the beam. A typical public address loudspeaker 18 is mounted near the ceiling 12, as shown, and it provides sound reinforcement coverage for the entire auditorium.

FIG. 2A shows some of the microphone elements 43-80 which are included in the linear array 10 of FIG. 1. These microphone elements, for example, may be a small commercially available type, such as the Shure Brothers WL83.

As shown in FIG. 2B, and in the following Table A, there are six sub-arrays provided to cover a frequency range of 156 Hz to 10 kHz with each sub-array covering one octave, and with a predetermined microphone spacing in each of the sub-arrays.

In the embodiment under consideration, there are twenty-five microphone elements in each sub-array. Only microphone elements 43-85 are shown in FIG. 2B. Microphone elements 1-42 (not shown) extend to the left of FIG. 2B. An examination of FIG. 2B will reveal that many of the microphone elements may be used in common in the various sub-arrays. This common usage of the microphone elements serves to reduce the number required, for example, from one hundred fifty to eighty-five in the illustrated embodiment. The microphone elements used in each sub-array are listed in Table A. It will be noted that sub-array 6, which covers the lowest frequency octave, and which is at least twice as long as the other sub-arrays, has microphone elements over the entire range from 1-85.

Each of the microphone elements of the array produces an output signal proportional to the instantaneous acoustic sound pressure imposed on each of the microphone elements, which changes in accordance with the frequency or frequencies of the sound signal source. In order to form a sound pick-up beam, the signal phase from each of the microphone elements must be coincident at the center of the beam. This means that a suitable time delay must be provided to each microphone channel to compensate for the difference in distance, and propagation delay, from the sound source at 16 to each microphone element. In the illustrated embodiment with the end-fire microphone array 10 of FIG. 1, and with the beam center extending along the longitudinal axis of the array, the time delays of each microphone element relate directly to the microphone element spacings in the array. Therefore, if microphone element 1 is closest to the sound source at 16, the channel of microphone element 2 requires a lesser time delay equivalent to the spacing between the two microphone elements 1 and 2.

FIG. 3 is a block diagram of the system of the invention in one of its embodiments and illustrates the basic sequential switching functions exerted on the output signals from the microphone elements 1-85 prior to their application to to an output amplifier 30. Output amplifier 30 drives the loudspeaker 18 of FIG. 1. A microprocessor 36 controls all the functions of the system under the control of a conventional clock circuit 38. In FIG. 3. The output of each of the microphone elements 1-85 is amplified in a corresponding preamplifier 32 and fed to a corresponding sample-and-hold circuit 34. Sampling is initiated by microprocessor 36, and it occurs simultaneously for all of the microphone elements 1-85. The sampling rate, must be at least two times the maximum frequency, for example 40 kHz.

The samples stored in the sample-and-hold circuits 34 are selected sequentially by microprocessor 36 and fed to an analog/digital converter 40 in which they are converted to corresponding digital signals. The digitized samples from converter 40 are stored in a random access memory (RAM) 42. This memory also serves to provide the variable delay required by each microphone element for beam forming, as discussed above. Such delay is accomplished by reading the samples of the different microphone elements from the memory at different subsequent sampling periods in accordance with the desired delays.

The twenty-five samples of sub-array 1, for example, are read from memory with the desired delays. An amplitude distributor circuit 44 receives the outputs of sub-array 1 and modifies their digitized amplitudes in accordance with a Taylor, or similar distribution factor to substantially reduce the beam sidelobe levels. The digital signals of sub-array 1 are then fed to a summing network designated "Add 1" in which they are summed to provide a beam output for the 5 kHz to 10 kHz frequency range of sub-array 1. The digital output is then processed in a digital band-pass filter designated "BP Filter 1" to attenuate any frequency components below and above the 5 kHz to 10 kHz range of sub-array 1 in order to suppress acoustic feedback. The filtering occurs over a number of sampling periods, depending on the signal frequency. The output is then applied to a digital-analog converter 46 in which it is converted to analog form to provide the analog sub-array 1 output, which is applied to amplifier 30.

The same procedure as described in the preceding paragraph is utilized for the samples of sub-arrays 2-6. All six sub-array outputs are combined and amplified for the array system output from amplifier 30. Microprocessor 36 (FIG. 3) performs all the control and processing operations of the signals from microphone elements 1-85, in the manner described above. It should be noted that the time delays for each of the microphone channels in a sub-array include any difference in delay of the sub-array bandpass filters, so that the total delays following the filters are equal. For example, the lowest frequency, and narrowest bandpass filter (156-312 Hz) has a longer dela than the highest frequency bandpass filter (5-10 kHz), and as a consequence all microphone channels of this latter sub-array 1 will have a longer delay than for sub-array 6.

It should be pointed out that all of the components of the system of FIG. 3 shown in block form are standard commercial elements, which are readily available. For that reason, it is believed unnecessary for the clear understanding of the present invention to show and describe the various components in circuit detail.

FIG. 4 illustrates the distance relations between the microphone elements of the array and a sound source of a selected sub-array in a simulated system. The sub-array elements are evenly spaced along the array length. Vectors "a" and "b" vary as a function of the angle with respect to the array for a given distance D, between the array center d13 and the sound source. It is noted that, for convenience, the angle .theta. is referenced from the perpendicular to the array axis, the array beam center (.phi. of FIG. 3) is then at 90 degrees. The term "s" represents the element-to-element spacing of a sub-array. The parameters for the distances d1-d25 of a sub-array can be determined from FIG. 4.

Table B lists the various parameters and equations for determining the plot of the beam characteristics. This table, along with the various parameters and their definitions, lists the distance equations d1-d25, using the relations of FIG. 4. Each of the distance equations represents that of a sub-array microphone. The term P is a fixed constant (0.4 in this case) used to determine the microphone spacing S. The value of P provides a balance between sidelobe level and acceptance beamwidth for a given array length. The term "v" represents the vertical offset distance of the array axis and the source. This term is normally set to zero if the array is pointing at the sound source, as is the case in FIG. 1. The first square root term in each of the equations determines distance over the range of angle .theta., and the second term provides for the distance related time delay for each element at a fixed beam angle .phi.. Each of the distance equations (d1 to d25) represents that of a sub-array microphone. Each microphone output voltage and the phase of its output voltage are represented by the quatrature "x" and "y" terms in their "Taylor" distribution multipliers (0.057-1.00). These are summed as "xs" and "ys" terms, and the sub-array output voltage "e" is determined by the square root of the sum of the squares of xs and ys. The output voltage e is for a defined angle theta, and for a particular sub-array with its element spacing, s, (i.e., spacing between adjacent microphone acoustic phase centers) and frequency. Response at any other frequency can be determined with the corresponding sub-array and element spacing.

Table C lists the x and y phase coordinate terms for each element as a function of the distance and wavelength. There is also a multiplier term for each element, a Taylor amplitude distribution term, (0.057 to 1.00) for the case shown will suppress the sidelobes some 46 dB or greater. The x and y terms are separately summed for each value of .theta. in Table C. The sub-array output e.sub..theta. is determined by taking the square root of the sum of x.sub..theta. and y.sub..theta. squares. The relative output in dB, edB.sub..theta., is also shown on the bottom of Table C.

FIG. 5A shows the resulting array angular response plot at 1250 Hz.+-.90 degrees from the beam center. The 3 dB beam width is approximately .+-.21 degrees, and the 40 dB beam width is .+-.43 degrees. FIG. 5B shows the very low response of the array in the rear 180 degrees. FIG. 6 illustrates how the beam may be broadened to a degree by setting the beam angle, with associated element delays to 75 degrees.

The beam width across each sub-array frequency band increases toward the low end of the band. This is because the element spacing, determined at the top end of the band, remains constant, leaving fewer wave lengths of aperture at the lower end of the band. However, there is a response overlap of the sub-array bandpass filters, and when the outputs of the adjacent sub-arrays are added the effect is to narrow the beam at the low end of the band and broaden the beam at the upper end of the band. This causes the beam widths to become equal. Depending on the sharpness of the filter cut-offs, the beam widths may be fairly constant over each sub-array band, and over the whole array band. FIGS. 7A and 7B illustrate that effect. FIG. 7A shows the one-half beam width at 1250 Hz, as in FIG. 5A; and FIG. 7B shows the broader beam width at 625 Hz at the low end of the band of the same sub-array.

Table D shows further calculations with respect to those shown in Table B with modified calculation of e.sub..theta.. Data for xs.sub..theta. and ys.sub..theta. at 625 Hz is stored as xs1.sub..theta. and ys1.sub..theta. and added to the data taken at the top end of the sub-array 5 at the same frequency, both at half value.

FIG. 8 shows the resulting half-beam plot. Note that the beam width lies between those of FIGS. 7A and 7B. It can also be shown that the beam width at 1250 Hz, when added to the low end of the sub-array 3, produces the same beam width as at 625 Hz. The beam width at the low end of sub-array 6, 156 Hz will be equivalent to that shown in FIG. 7B.

One of the factors of concern in an auditorium with sound reinforcement is the isolation between the microphone and the loudspeaker which limits the acoustic gain that can be utilized before acoustic feedback occurs. The following Table E shows an analysis which indicates that with the very low array sidelobes it is possible to provide sound levels throughout a typical auditorium equivalent to that at 6 feet from the sound source with approximately 20 dB feedback margin.

The invention provides, therefore, a unique beamed microphone array system which utilizes a linear end-fire array of microphone elements directed at the source of sound to be enhanced, and which represents an economical and practical system for enhancing the sound so that it can be distinctly heard throughout the auditorium or other facility without acoustic feedback. The microphone array system may also be used for recording at relatively large distances from the source.

While a particular embodiment of the invention has been shown and described, modifications may be made. Variation in beamwidth, sidelobe levels and frequency band are possible with changes in the number of elements in each sub-array, changes in the amplitude distribution and number of sub-arrays, as known to those skilled in the field of acoustic physics. It is intended in the claims to cover all such modifications which come within the true spirit and scope of the invention.