DESKTOP MIC ARRAY FOR TELECONFERENCING
`
`Peter L. Chu
`
`PictureTel Corporation, MS 635
`222 Rosewood Dr.
`Danvers, MA 01923, USA
`chu@pictel.com
`
`ABSTRACT
`Reducing the noise and reverberance in sound pickup
`has been a problem ever since the microphone was in-
`vented. Elegant solutions using multiple microphones
`in an array are a current hotbed of research [l] [2]
`[3] [4]. Unfortunately, because of the computational /
`monetary cost of these approaches, they have not been
`widely implemented in products.
`In this paper, an automatically steered mic array,
`which works by taking linear combinations of two dipole
`microphones, is presented whose cost is low enough to
`have been implemented in a videoconferencing prod-
`uct. The array is positioned centrally on the conference
`table and provides very reasonable pickup for people
`speaking within a 7 foot radius, adequate for most con-
`ferencing situations. While simple in structure, the ar-
`ray provides a large increase in convenience and perfor-
`mance compared to the common method of laying out
`multiple cardioid microphones on the table, where each
`participant must be within the pickup angle / range of
`a cardioid microphone.
`
`1. DESKTOP ARRAY STRUCTURE
`
`The desktop array consists of two dipole microphones
`mounted perpendicularly to each other, as close to each
`other and as close to the table as possible. The main
`beams of the microphones are parallel to to the table
`top surface. The two dipole microphone outputs go to
`the left and right channels of a stereo A/D converter,
`whose output, in turn, goes to the DSP chip. A block
`diagram of the structure is shown in figure 1. Assuming
`the source is in the far field, as a function of angle the
`response of the dipole microphone to the source is
`
`shows that the dipole has a response of 1 when 8 =
`0, -1 when 8 = 180 degrees, and 0 when 8 = +/ -
`90 degrees. The dipole decreases isotropic noise and
`reverberance by 4.8 dB compared to an omnidirectional
`microphone, assuming the source is on-axis. The more
`commonly used cardioid or unidirectional microphone
`has response
`
`The cardioid pattern has a response of 1 when 8 =
`0, 0 when 8 = 180 degrees, and .5 when 0 = +/ -
`90 degrees. Both cardioid and dipole directivity pat-
`terns reduce isotropic noise and reverberance by equal
`amounts. However, if the noise source is predominantly
`overhead, as is the case for air conditioning vents, the
`dipole with its main beam parallel to the tabletop sur-
`face will do a better job than the cardioid in attenu-
`ating the vent noise because of its response null in the
`vertical axis. On the other hand, because of strong
`reflections from surfaces directly opposite the person
`speaking, the cardioid, with its null in the opposite
`direction of its main axis, often sounds slightly less re-
`verberant than the dipole. Overall, weighing the ad-
`vantages and disadvantages of both, the two patterns
`are fairly equal choices for microphone pickup.
`Assuming a fixed frame of reference for the two per-
`pendicularly mounted microphones, the response for
`microphone A is
`
`and the response for microphone B is
`
`where A is the sound pressure level of the source at
`the dipole and tJ is the angle between the source and
`the on-axis angle of the dipole. Examination of (1)
`
`Adding D A , the signal of microphone A, to D B , the
`signal from. microphone B, and scaling by fi,
`
`(5)
`
`2999
`
`0-7803-2431 6/95 $4.00 0 1995 IEEE
`
`WAVES607_1009-0001
`
`Petitioner Waves Audio Ltd. 607 - Ex. 1009
`
`

`
`yields
`
`Dc(e) = cos(e - 450)
`(6)
`which is simply a dipole microphone pattern shifted
`45 degrees relative to the main axis of microphone A.
`Similarly, subtracting Dg from DA and scaling by f i
`yields
`DD(6') = cos(6 + 45')
`(7)
`which is a dipole microphone pattern shifted -45 de-
`grees relative to the main axis of microphone A. There-
`fore, by taking the sum and difference of the two dipole
`signals and scaling appropriately, easily done in the
`DSP, it is possible to derive two additional dipole pat-
`terns oriented halfway in angle between the two origi-
`nal patterns. The four pickup patterns defined, DA(8),
`DB(8), Dc(S), and DD(8) (shown in figure 2) ade-
`quately cover a fuIl 360 degrees of arc, since a source
`halfway between two beams is down only .688 dB from
`maximal on-axis response. In fact, any arbitrary an-
`gle of rotation of the dipole pattern can be achieved by
`taking the appropriately weighted linear combination
`of the two dipole micropone signals.
`
`2. DESKTOP MIC ARRAY BEAM
`SELECTION
`
`The algorithm which chooses which of the four beam
`patterns to use in picking up the source should be in-
`sensitive to constant background noise from air vents
`and reverberant energy. Computational simplicity is
`also a major concern.
`The steps in the algorithm for beam selection will
`now be outlined.
`1. Bandpass Filtering- The left and right channels
`from the stereo A/D converter are fed into two separate
`but identical FIR bandpass filters which let through
`frequencies in the 1-4 kHz region (the sampling rate of
`the system is 16 kHz). The bandpass filtering gets rid
`of much of the lower and higher frequency background
`noise. The speech signal below 1 kHz tends to be more
`reverberant than higher frequencies so is less useful for
`finding the source direction. For the left channel, the
`bandpassed output is
`
`c
`
`k < L
`b ( n ) =
`I(" - k)h(k),
`k=O
`and for the right channel,
`
`(8)
`
`k < L
`T(" - k)h(k).
`q ( n ) =
`k = O
`2. Decimation by Four- To reduce computations
`involved in the FIR bandpass filter in the previous step
`
`(9)
`
`by a factor of four, the outputs of the bandpass filters
`are decimated by four. While aliasing is introduced
`in this process, the aliasing has little effect on later
`calculations in which energy will be measured. For the
`left channel,
`
`Lb(m) = lb(4m)
`and for the right channel,
`
`(10)
`
`&(m) = r44m).
`(11)
`3. Formation of Four Beams- Signals from the four
`dipole patterns are derived by taking the appropriately
`scaled sums and differences of the two bandpass, sub-
`sampled signals. The absolute value is taken of the
`samples, so that
`
`(12)
`
`(13)
`
`(14)
`
`(15)
`
`A1 (m) = I Lb ( m ) I
`A+) = IRb(m)I
`1
`Jz
`A3(m) 1 I-(Lb(m) 4- &(m))l
`1
`A4(m) = I-(h(m) - &(m))l
`fi
`4. Average Level found in 20 msec. Blocks- The
`terms A;(m), i = 1,2,3,4 are averaged in 20 millisec-
`ond blocks.
`5. Background Noise Level Estimate- Over the
`last 2 seconds, the minimum 20 millisecond block level,
`derived in step 4, is found for each of the 4 beam pat-
`terns. This value is averaged against previously found
`minima in previous 2 second intervals of time. The re-
`sult is a somewhat biased estimate of the background
`noise level due to vents, fans, etc. A different back-
`ground noise level estimate results for each of the 4
`beam patterns.
`6. Background Noise Level Subtraction- The
`background noise estimate is subtracted from the terms
`in step 3. If the result is less than zero the term is set
`to zero. For i = 1 , 2 , 3 , 4 and N, defined as the noise
`estimate for dipole pattern i,
`&(m) = A,(m) - N,
`(16)
`under the condition that if B;(m) < 0, then B,(m) =
`0. The purpose of this subtraction is to eliminate the
`influence of background noise on beam selection.
`7. Short Term Integrator- The samples from step 6
`are next fed to a short time integrator, to provide some
`smoothing of isolated peaks. For i = 1,2,3,4,
`Ci(m) = .25B;(m) + .75C,(m)
`(17)
`8. Running Peak - To mitigate the effects of rever-
`berant energy on beam selection, a running peak value
`
`3000
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`WAVES607_1009-0002
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`Petitioner Waves Audio Ltd. 607 - Ex. 1009
`
`

`
`is developed for each beam. The philosophy is that the
`peak value of a signal will be proportional to the direct
`path energy while the decaying tails of the signal will
`have a larger portion due to reverberant energy. For
`i = 1 , 2 , 3 , 4
`
`if Di(m) > Ci(m), Di(m) = Ci(m)
`
`(18)
`
`(19)
`else Di(m) = .996Di(m)
`9. Sum of Running Peak and Beam Selection -
`Over a 20 millisecond frame, the sum of the values of
`Q(m) for i = 1 , 2 , 3 , 4 are found, and the index i that
`produces the largest sum is the dipole pattern which is
`chosen as maximizing the source pickup quality. Mak-
`ing decisions every 20 milliseconds has been found to
`lead to no noticeable degradation in performance. In
`fact, the beam selection algorithm has been found to
`yield high quality sound pickup even for the case of
`multiple people talking simultaneously.
`
`3. DAISY CHAINING DESKTOP ARRAYS
`
`It has been found by experiment that a single mic desk-
`top array picks up people well in a 7 foot radius circle
`about the mic desktop array. Two mic desktop arrays
`may be used by simply adding the left channel of the
`first desktop array to the left channel of the second
`desktop array and adding the right channel of the first
`desktop array to the right channel of the second desk-
`top array and then feeding the resultant summed left
`and right channel signals to the stereo A/D converter.
`Each desktop array will have a beam active. The beam
`selection algorithm for a single mic desktop array works
`well for multiple mic desktop arrays. The addition of
`a second mic desktop array increases the noise and re-
`verberance by 3 dB because the added presence of a
`second beam. The effect of the 3 dB worsening of the
`signal-to-noise is to reduce the radius of coverage of
`each desktop array to 5 feet. Thus, the use of multiple
`desktop arrays does not increase the total area of cov-
`erage but merely serves to alter the shape of the area
`of coverage. In the case of one mic desktop array vs.
`two mic desktop arrays, the pickup area changes from
`a single circle of radius 7 feet to two circles of radius 5
`feet. The area of the two smaller circles equals that of
`the single large circle.
`
`4. ACOUSTIC ECHO CANCELLATION
`
`The acoustic echo canceller duplicates the room trans-
`fer function between loudspeaker and microphone, fil-
`ters the loudspeaker signal with this transfer function,
`and subtracts the result from the microphone signal.Via
`
`this procecedure, the component of the loudspeaker
`signal is eliminated from the microphone signal, with
`no effect on other components of the microphone sig-
`nal. The loudspeaker-to-microphone transfer function
`changes drastically for different desktop array beams,
`so therefore, the loudspeaker-to-mic transfer function
`appropriate to the currently chosen desktop array beam
`must be used for echo cancellation. The acoustic echo
`canceller could store four sets loudspeaker-to-mic fil-
`ter coefficients (which would have to be continually
`updated due to the changing nature of the acoustic
`paths). Alternatively, two echo cancellers, one for the
`left channel signal and one for the right channel sig-
`nal of the stereo A/D converter, could be used, with
`the echo canceller outputs being summed or subtracted
`together to produce the four beam patterns. As yet
`another alternative, one echo canceller could be used
`with only two sets of loudspeaker-to-mic coefficients
`stored, those corresponding to the two base compo-
`nent dipole microphones. When needed, the two miss-
`ing sets of loudspeaker-to-mic filter coefficients could be
`derived from these two stored sets by invoking the same
`operations needed to generate the two shifted dipole
`beams from the two component dipole beams, i.e., ei-
`ther add or subtract filter taps from the two sets of
`stored loudspeaker-to-mic filter coefficients, and then
`scale the resulting sum or difference by .7071 to de-
`rive the tap values for the missing loudspeaker-to-mic
`filter coefficient set. By examining the adaptive tap
`values found for the previous beam choice and the cur-
`rent (different) beam choice, one could easily derive the
`adaptive filter tap values for the two base component
`adaptive filters (two equations in two unknowns).
`
`5. REFERENCES
`
`[l] J . L. Flanagan, J. D. Johnston, R. Zahn, G. W.
`Elko, “Computer-steered Microphone Arrays for
`Sound Transduction in Large Rooms”, J. Acoust.
`Soc. Am. 78(5), November 1985, pp. 1508-1518.
`‘‘ Constant
`[2] M. M. Goodwin, G. W. Elko,
`Beamwidth Beamforming”, ICASSP-93, pp. I-169-
`1-172.
`[3] J . L. Flanagan, D. A. Berkley, G. W. Elko, J. E.
`West, and M. M. Sondhi, “Autodirective micro-
`phone systems”, Acustica 73(2), 1991, pp. 58-71.
`[4] Yves Grenier, “A Microphone Array for Car Envi-
`ronments”, Speech Communication 12( l ) , March,
`1993, pp. 25-39.
`
`300 1
`
`WAVES607_1009-0003
`
`Petitioner Waves Audio Ltd. 607 - Ex. 1009
`
`

`
`Dipole B 4
`
`Left Channel
`of Stereo AID
`
`Right Channel
`of Stereo A/D
`
`I
`
`Conference Table
`
`Figure 1. Schematic of the desktop mic array structure, overhead view,
`looking down onto the conference table.
`
`Dipole Mic A Dipole Mic B Dipole Mic .7071 (A+B) Dipole Mic .7071 (A-B)
`
`Figure 2.4 dipole pickup patterns
`
`3002
`
`WAVES607_1009-0004
`
`Petitioner Waves Audio Ltd. 607 - Ex. 1009