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Automated measuring system for sound power measurement
Applied Acoustics15(1982)445~157
AUTOMATED MEASURING SYSTEM FOR SOUND POWER MEASUREMENT
T. YANAGISAWAand W. TSUJITA
Faculty of Engineering, Shinshu University, 500 Wakasato, Nagano 380 (Japan) (Received: 26 January, 1982)
SUMMARY
It is sometimes important to know the sound power radiated from a sound source so that,for example, the solution noise abatement can be carried out on the basis of the sound power radiated. Measurement of the sound power in general is carried out in a reverberant enclosure. This measuring method is classedas an indirect one in that the soundpower is obtained via the soundpressure level. Consequently, in order to determine the sound power radiated as precisely as possible, some measurement procedures for obtaining the space-average sound pressure have been devised. However, the procedure, which invoh,es moving the measuring microphone from point to point, is extremely tedious. Accordingly, if the system from data acquisition to data processing is automated, the work required for obtaining the space-average sound pressure will be considerably reduced, as will the time required for processing the data. The development of an automatic measuring system for sound power is discussed and it is shown that the sound power obtained with the system devised agrees well with that obtained by more familiar means.
INTRODUCTION
As is well known, the methods for measuring sound power are grouped into direct and indirect. The direct method is based on measuring both the sound pressure and the particle velocity but to carry out a simultaneous measurement is difficult, so this method is not generally available yet. In contrast to this, the indirect method based on measuring only the sound pressure is well-established. Both the reverberant and an anechoic r o o m method are grouped under the 445 Applied Acoustics 0003-682X/82/0015-0445/$02.75 © Applied Science Publishers Ltd, England, 1982 Printed in Great Britain
446
T. YANAGISAWA, W. TSUJITA
heading of 'indirect'. The anechoic chamber method is more troublesome than the reverberant room method, Measuring the sound pressure in an anechoic chamber is more complicated because of the directionality on the source measured. In contrast to this the effect of the directionality can, in principle, be neglected in the reverberant room method, a situation familiar in the measurement of sound power. However, it is known that there is a problem ill the reverberant room method; the measuring procedure to obtain the space-average sound pressure is tedious because the measuring microphone has to be moved from point to point at all measuring frequencies. In order to electrically improve such a procedure the use of a microphone multiplexing system has already been proposed. 1'2 However, automation of the total system for data acquisition including computation has probably not been devised yet. Accordingly, if the total system is automated by making use of the microprocessor, etc., the total time required for not only the data acquisition but also the computation will be considerably reduced. The automated measuring system devised here consists of two parts, one for acquiring data and the other for software calculation of the sound power. All the hardware is synchronously controlled by timing pulses produced by the software, and sequentially the results obtained are printed on recording paper after computation on the acquired data. These sequential processes carried out at every measuring frequency are automatically repeated to the chosen last stage. BASIS OF THE AUTOMATED MEASURING SYSTEM
Discussion relating to reverberant enclosure A requirement in measuring sound power by the reverberant room method is that the sound field in the room is sufficiently diffuse. The reverberant room used has the dimensions 68.7 m 3 in volume and 101-2 m 2 in surface area and the sound diffusion in the room is satisfied at 400 Hz as shown in previous results, 3 The room constants shown in Fig. 1 are used in known values in the computation of sound power as described below. 2.0, 1.8 LEE
~ / ~
'~ 1.6!
J
o
E 1.,i o o
,
I 200
I
I I i ,,,I
500 1½ Frequency (Hz)
2K
Fig. 1. Response of room constant versus frequency.
AUTOMATED MEASURING SYSTEM FOR SOUND POWER MEASUREMENT
447
Discussion on not only the position of the sound source but also that of the measuring microphone is just as important as that on the sound diffusion. The sound source to be measured has to be placed in such a position that the effect of the walls on the source can be neglected. The sound source position is determined on the basis of ISO recommendation (ISO 3741), i.e. the source is placed 1.5 m away from the walls. The microphone positions and the size of the group of measurements has first to be determined. The size can be determined by considering both the spatial variation and the allowable error in the sound pressure. According to Lubman 4 the number of the measuring microphone numbers required is more than 8 when within the one third octave band width centred on 100 Hz, the reverberation time is 10.5 s and the variation in sound pressure level is _ 0-5 dB. In this instance 16 measuring microphones are used. It also has to be noted that the spacing between microphones is more than the one half wavelength at the lowest measuring centre frequency. The lowest centre frequency recommended by ISO for the reverberant room in the present study is 200 Hz. Consequently, the spacing between microphones corresponding to 200 Hz can be determined here. Finally, it is important to discuss whether or not the effect of the direct sound on the microphone signal can be neglected. Consider the smallest distance between the sound source and the microphone to be determined by the aid of the reverberation distance. If the effect of the direct sound on the reverberant sound is postulated to be less than 0-1 dB, their separation, r, can be given by r > 0-4w/~
(m)
(1)
where V is the volume of the reverberant r o o m and T the reverberation time. F r o m eqn. (1) it can be seen that the smallest separation between sound source and the microphone should be given for the upper frequency because the reverberation time is, in general, reduced inversely with increase in frequency. Consequently, the separation in this case is 2.1 m as T = 2.56s at 4 k H z and V = 68.7m 3. On the basis of the discussions described above, the sound source and the measuring microphones in the reverberant room were finally arranged as shown in Fig. 2. The sound source was placed on the floor and the microphones fixed at 1.7 m above the floor. Discussion relating to hardware The general system for sound power measurement consists of a sound level meter, one third octave band filter and level recorder. The system to which this paper refers was newly constructed with integrated circuit devices making possible the automation of data acquisition. The system constructed is shown in the Fig. 3(a) and the level diagram in Fig. 3(c). Each device in the system is sequentially controlled by timing pulses from the microcomputer shown in Fig. 3(b). The response or the characteristics of each device are described below. The microphone amplifiers are composed of field effect transistors, each of them
448
Fig. 2.
T. YANAGISAWA,
W. TSUJITA
Configuration of sound source and measuring microphones are in metres. 0, Sound source; 0, measuring
Mic.Amp
Amp.
FI lter
Fe.45
A/D
in reverberant microphones.
room.
Input
(cl
Fig. 3. Block diagram for automatic sound power measuring system and its level diagram. (a) Measuring instruments in system; (b) microcomputer; (c) level diagram of measuring instruments.
AUTOMATED MEASURING SYSTEM FOR SOUND POWER MEASUREMENT
449
having a constant gain of 33 dB and each having a flat frequency response over 100 Hz. In order to electrically move the measuring microphones from point to point, a microphone multiplexing system is used. The IC device for the multiplexing system is type IH-6116 made by Intersil Co. Ltd, and its chanhel capacity equals the number of measuring microphones as determined in the preceeding section. Each measuring microphone is connected to the multiplexing system device through 5 bits in parallel (4 bits for channel selection and 1 bit for chip enable) from the microcomputer. The amplifier following the multiplexing system is made from an operational amplifier. The gain of the amplifier is fixed at a constant 30 dB. This is so that in the case of the microphone with the lowest sensitivity of - 83 dB, the power term in eqn. (7) described in the following section can be cancelled by the total gain of the amplifiers concerned. The frequency response of the system devised is fiat. An active filter device is adopted for the one third octave band filter and its centre frequency and band width can be freely changed. The IC device for the active filter is type 300 VT-4 made by the Frequency Devices Co. The centre frequency can be varied in the range 100 Hz to 2 kHz by controlled voltages of 0.5 V to 10 V, and the band width corresponding to 23 ~o of the centre frequency can be adjusted with the Q value (Q = 4.3 corresponds to 23 ~o band width). The step-wise output voltage through the digital analog converter set in the computer is used for the controlling voltage. The appearance of the step-wise voltage is synchronised with the operation on the multiplexing system as described above. The IC device is also used to obtain the root mean square value of the signal filtered by the active filter. The IC device is type 440J made by the Analog Devices Co. and the operation is in accordance with the formula for obtaining the root mean square value. The integration time for this device can be controlled by the condenser capacity disconnected from the element block, but the time should be determined by the experiments in practice. D i s c u s s i o n r e l a t i n g to s o f t w a r e
The foundation of the program processing the data acquired by the hardware described in the above section is now discussed. It is based on a well-known formula to be found in the textbooks on room acoustics.5 The sound energy density, E, at any point in an enclosure is given by WQ 4w E = E d + E r - 4~r2 c + R c
(W/m3)
(2)
where E a is the sound energy density of the direct sound, E r the sound energy density of the reverberant sound, W the sound power radiated from the sound source, c the speed of the sound in air, Q the directivity factor of the sound source, r the distance between the sound source and the receiving point, and R the room constant as described before.
450
T. Y A N A G I S A W A , W. TSUJITA
If the first term on the right-hand side in eqn. (2) is less than the second term, eqn. (2) can be approximated to 4W E=~ (W/m 3) (3) On the other hand, the sound energy density can be derived from sound intensity and the sound intensity from the sound pressure. Accordingly, if these relations are connected by eqn. (3), the sound power required can be expressed as a function of the sound pressure as W=
p2R 4pc
(W)
(4)
where P is the rms sound pressure and p the density of air. Furthermore, the sound pressure described in eqn.(4) can be related to the microphone signal as follows:
EI=PIO s/2° (V)
P=EllO -s/2° (/~bar)
or
(5)
The microphone signal, E 1, is amplified by each microphone amplifier and further by another amplifier common to all the microphones as shown in Fig. 3. If the total gain of the microphone channel just described is G dB, the sound pressure given by eqn. (5) can be re-written as a function of the output voltage on the amplifier in the last stage as
P=E210 I-s+~)/2° (/~bar)
or
P=E210 ~-s+a+2°)'2° (Pa)
(6)
where E 2 is the output voltage in the last stage. Substituting P given by eqn. (6) into P given by eqn. (4), the sound power required for the single microphone channel is
W = E210 t-s+a+2°):l°. R/4pc (W)
(7)
However, in order to obtain the sound power as precisely as possible, the sound powers obtained with each microphone channel have to be averaged corresponding to the space-average of sound pressure. At the first measuring frequency required, the output voltage on the last stage in the appropriate channel is squared and then it is multiplied by a constant including only the microphone sensitivity and the total gain. Second, in order to get the mean sound power, the values computed at every channel are summed and then they are averaged multiplying by another constant including the number of the channel. Those computation routes are automatically repeated up to and including the last measuring frequency required. 16
Wf= 2(EiZ)(lO(-S+C+2°'l|°,1 ( 4~ -. ~ ) i=1
(W)
(8)
AUTOMATED MEASURING SYSTEM FOR SOUND POWER MEASUREMENT
451
where the subscript, f, shows each centre frequency on the one third octave band filter and i the number of the microphone channels. The computations of eqn. (8) are carried out with the fixed point numeration in the binary number and the results are printed on the record paper after having translated them to the decimal number with ASCII code.
OPERATION OF AUTOMATED MEASURING SYSTEM
Both the operation of the hardware and the computation of the sound power included in the automated measuring system are carried out through the microcomputer, TMS990/101M made by Texas Instruments Co. Ltd. The flow chart shown in Fig. 4 is used for the operation and the computation. The operation for the measuring system can be simply explained according to the flow chart. At the start of the operation, the lowest measuring centre frequency is set at 100 Hz by the step-wise voltage produced with the digital analog converter in the computer and at the same time, the measuring microphone No. 1 is connected to the
START) lSET CF I JMPXCH ON TON
I ,
I NO
J AID
JMPXCHOFF J, I
Ton=
I
Fig. 4. Flow chart for sound power measuring system. CF, centre frequency of active filter; M P X CH, multiplexer channel; ToN , time holding on multiplexer; Tovr, time holding off multiplexer; *, calculation of ~F '6 E 2 " 10 (s +G+2o)/,o given in eqn.(8); P O W E R CAL, power calculation given by eqn. (8). L..~ i = 1
452
T. YANAGISAWA, W. TSUJITA
relevant channel on the multiplexing system by the 5 bits signal through the parallel I/O port. While the multiplexing system is set in, following the program, the analog signal on the relevant channel through the root mean square device is converted to the digital signal and squared. The squared value is multiplied by the constant including the microphone sensitivity and the total gain of the relevant channel. On the other hand, while the multiplexing system is set off, the corrected digital signal just described is stored in the register in the computer. Those operations are sequentially repeated to the last channel. The values stored are summed and averaged, multiplying by another constant including the reciprocal of the channel number. The mean sound power obtained at 100 Hz is printed. After printing the mean sound power obtained at 100Hz, the next centre frequency is automatically sent through the program, and the same process as described at 100 Hz is carried out. After these processes have been repeated up to and including 2 kHz, given as the last stage, all the processes necessary to obtain sound power are complete.
EXPERIMENTS AND RESULTS FOR AUTOMATED MEASURING SYSTEM
Proper characteristics for the system devised In order to test whether the measuring system itself is correctly operated as planned, experiments are carried out using equivalent microphone signals. First, experiments and results are reported on the dynamic range of the system. The equivalent microphone signal at 1 kHz was applied to the terminal on the active 110
~
1o0
E 90 '- 80
~ 6c •
50
/ lOmV lOOmY lV Input voltage to A I D
u 40
I
I
I
I
/
50
60
70
80
90
10V I
100 110
Theoretical p o w e r spectrum level at l k H z (dB)
Fig. 5.
Correlation diagram between values measured when applying equivalent microphone signals to active filter in system and those calculated separately.
AUTOMATED MEASURING SYSTEM FOR SOUND POWER MEASUREMENT
453
filter. The equivalent input voltages were varied in the range 10 mV to 10 V. The values shown on the ordinate in Fig. 5 are obtained by the designed measuring system and the values on the abscissa in the same figure are calculated separately. From the correlation diagram between the measured and the calculated values shown in Fig. 5, it can be seen that the agreement is good for more than 100 mV and that the difference between them below 100 mV is within 1 dB. The difference may be the result of fixed point numeration in the calculation and it may be improved by adopting a floating point numeration. However, as the computation method by the computer concerned is limited to fixed point numeration, the difference between the values obtained in the range of the lower input level is not really improved. Secondly, the experiments and results are reported for the frequency response of the designed system. The experiments were carried out applying the equivalent microphone signal at 1 mV constant to the input terminal of the appropriate microphone amplifier. The frequency was varied in the range 100 Hz to 2 kHz. The frequency response shown in Fig. 6 is shown as a function of the difference between the values measured and the values calculated separately. From the results shown in the same figure, it can be seen that there are no differences between the measured and theoretical values except for the lower frequency ranges. A very small difference in the lower frequency ranges may result in setting the cut-off frequency of the microphone amplifier to 80 Hz. Hence, from the results described above, it can be said that the measuring system can be used in practice.
cG 0
J ' - ~
;-7, 100
I 2O0
f
I
I ~ ~1~1 500 1k Frequency (Hz)
I
2k
Fig. 6. Frequency response described by difference between values measured when applying equivalent microphone signal to one appropriate microphone amplifier in system and response calculated separately.
Speed-up of operation The operation speed for the measuring system is mainly affected by the integration time in the root mean square circuit and the time holding on and off the multiplexing system. Thus in order to increase the data acquisition speed keeping within the range of the quoted allowable error, both times described above have to be properly determined by experiments on the microphone signals picked up in practice in the field. Experiments for the time combinations given in Table 1 were carried out. In these experiments, the white noise radiated from the
454
X. YANAGISAWA, W. TSUJITA TABLE 1 COMBINATIONS OF INTEGRATION TIME FOR RMS C I R C U I T AND TIMES H O L D I N G ON A N D OFF M U L T I P L E X E R
Integration time (ms)
On-time (ms)
Off-time (ms)
83.5
333 1000
20 900
133-5
333 1000
20 900
loudspeaker with a radius of 5.6cm was used for the sound source, and the 16 electret condenser microphones were used for the measuring ones. Their arrangements in the reverberant room were as in Fig. 2. Moreover, the measurements for each time combination were carried out three times. The experimental results are shown in Figs 7(a) to (d). From the combination of Figs 7(a) and (b) showing the results obtained for the integration time of 83-5 ms, the variation of the values measured on each of the three occasions was found to be small in the frequency ranges less than 250 Hz. However, from Figs 7(c) and (d)
~
75
~
65
1
~55
I
I
(c)
O Q. "1~75
rn
// 100
I
I
I
200
500
11~,
ss 2k, 100
I
I
200
500
I lk
2k
Frequency (Hz) Fig. 7. Band power levels measured with test device. (a) Integration time for R M S circuit is 83.5 ms; times holding on and off multiplexer are 333 ms and 20ms, respectively. (b) Integration time for RMS circuit is 83.5ms; times holding on and off multiplexer are 1000ms and 900ms, respectively. (c) Integration time for RMS circuit is 133.5 ms; times holding on and off multiplexer are 333 ms and 20 ms, respectively. (d) Integration time for R MS circuit is 133.5 ms; times holding on and off multiplexer are 1000ms and 900ms, respectively.
A U T O M A T E D M E A S U R I N G SYSTEM F O R S O U N D P O W E R M E A S U R E M E N T
455
obtained for the integration time of 133.5 ms, the value of each measurement on the three occasions agreed well with each other in the range of the measuring frequency. Consequently, in order to increase the operation speed, keeping the allowable error within 1 dB, the integration time of 133.5 ms should be adopted for the root mean square circuit and the combination of 333 ms and 20 ms is appropriate for holding on and off the multiplexing system. The total time required from data acquisition to processing it is 1 min 40 s with the time combination as determined above.
Compar&on with more usual methods It has already been confirmed that the measuring system itself satisfies the fundamental requirements and is capable of being applied to field measurements. Results of measurements with the system are compared with those obtained with other measurement techniques. First, the results of the frequency analysis obtained with each measurement are compared with white noise radiated from the same loudspeaker as described in the preceding section and with the noise from an electric cleaner. In the measurements with the loudspeaker as sound source, two typical reverberant room and anechoic chamber methods were added to the method developed here. In each case the overall sound pressure level of the white noise was kept at a constant level of 86 dB. The experiments with the described system were carried out with the same arrangements of sound source and measuring microphones as shown in Fig. 2. Measurements with the reverberant room method were also carried out with the same arrangements as in the system described but the measuring instruments were ones generally used and the sound power calculated by the well-known formula. On the other hand, in the measurements with the anechoic chamber method, the sound powers were calculated on the basis of 10 sound pressure levels measured on an hemisphere with a radius of 1 m around the source. The frequency analysis results obtained from the three different measuring methods are shown in Fig. 8. It can be seen that the results obtained are all similar to
rn v -~ 70
n
m•5G100
~ I 200
~ ~ I , t,ll 500 Frequency
, lk
2k
(Hz)
Fig. 8. One third octave band power levels for white noise radiated from loudspeaker, analysed by various methods. - - , Anechoicroom method; , normal reverberationmethod; - - - --, automatic method with test device.
456
T. YANAGISAWA, W. TSUJITA TABLE 2 MEAN
VALUES OF DIFFERENCE OF MEAN V A L U E S MEASURED BY VARIOUS M E T H O D S
Comparison of different methods
Mean L,alues of difference (dB)
Automatic method versus normal reverberation method Automatic method versus anechoic method Normal reverberation method versus anechoic method
0.7 1.3 0.6
one another. In order to review in more detail the differences in the measurements, information on differences between the values obtained is tabulated in Table 2. From the results shown in Table 2 the mean value of the differences obtained by the automated and generally used reverberation techniques is 0-7dB, between the automated and anechoic chamber methods 1.3 dB and between the reverberant and anechoic chamber methods 0.6 dB. Hence it can be said that the reliability of the values obtained by the new system is the same as obtained by other methods of measurement. In the measurements on the electric cleaner the usual reverberation room method was added to the method of the new system. Measurements by the usual method were carried out using only one microphone and the general instrumentation and the sound power calculated on the basis of 16 sound pressure levels. The experimental results are shown in Fig. 9 together with the values calculated with the formula recommended by ISO. From the figure it can be seen that the values obtained agree well in the measuring frequency ranges except for the values at 100Hz and 125 Hz.
80 an v
~> 70 o~ 6o I Q. 13
g so m
100
'
I
200
,
,
I
J ,,,I
500 Frequency
lk
2k
(Hz)
Fig. 9. One third octave band power levels for electric cleaner analysed by various methods. - calculated with the equation recommended bu ISO; - - - , normal reverberation method; - - - automatic method with test device.
AUTOMATED MEASURING SYSTEM FOR SOUND POWER MEASUREMENT
457
CONCLUSION In o r d e r to cut d o w n the time r e q u i r e d for m a n u a l w o r k in o b t a i n i n g s o u n d p o w e r by the r e v e r b e r a t i o n r o o m m e t h o d , an a u t o m a t e d system has been developed. The system consists o f b o t h h a r d w a r e for d a t a acquisitions a n d software for the calculation, a n d the o p e r a t i o n on each p a r t is sequentially c o n t r o l l e d by the a l g o r i t h m p r o p e r for the m e a s u r e m e n t . F r o m e x p e r i m e n t a l results o b t a i n e d in s o u n d power m e a s u r e m e n t s it can be seen that the system has e n o u g h precision a n d reliability for use in practice. So it can be said that the a u t h o r s ' initial goal o f a u t o m a t i n g s o u n d power m e a s u r e m e n t has been achieved.
REFERENCES 1. E. J. WOOTTEN,Instrumentation for space averaging sound pressure levels, J. Sound Vib., 16(1) (1971), pp. 59-69. 2. H. HOGNESTADand O. H. BJOR,A microphone multiplexer for the measurement of the space average of sound levels, Applied Acoustics, 8(1) (1975), pp. 13-25. 3. T. YANAGISAWAand T. UEMURA,Reverberation time measuring system with correlation technique, Applied Acoustics, 14(5) (1981), pp. 377-85. 4. D. LUBMAN,Spatial averaging in sound power measurements, J. Sound Vib., 16(1) (1971), pp. 43-58, 5. For example, see L. L. BERANEK,Acoustics, McGraw-Hill, New York, 1954, p.318.
AUTOMATED MEASURING SYSTEM FOR SOUND POWER MEASUREMENT
T. YANAGISAWAand W. TSUJITA
Faculty of Engineering, Shinshu University, 500 Wakasato, Nagano 380 (Japan) (Received: 26 January, 1982)
SUMMARY
It is sometimes important to know the sound power radiated from a sound source so that,for example, the solution noise abatement can be carried out on the basis of the sound power radiated. Measurement of the sound power in general is carried out in a reverberant enclosure. This measuring method is classedas an indirect one in that the soundpower is obtained via the soundpressure level. Consequently, in order to determine the sound power radiated as precisely as possible, some measurement procedures for obtaining the space-average sound pressure have been devised. However, the procedure, which invoh,es moving the measuring microphone from point to point, is extremely tedious. Accordingly, if the system from data acquisition to data processing is automated, the work required for obtaining the space-average sound pressure will be considerably reduced, as will the time required for processing the data. The development of an automatic measuring system for sound power is discussed and it is shown that the sound power obtained with the system devised agrees well with that obtained by more familiar means.
INTRODUCTION
As is well known, the methods for measuring sound power are grouped into direct and indirect. The direct method is based on measuring both the sound pressure and the particle velocity but to carry out a simultaneous measurement is difficult, so this method is not generally available yet. In contrast to this, the indirect method based on measuring only the sound pressure is well-established. Both the reverberant and an anechoic r o o m method are grouped under the 445 Applied Acoustics 0003-682X/82/0015-0445/$02.75 © Applied Science Publishers Ltd, England, 1982 Printed in Great Britain
446
T. YANAGISAWA, W. TSUJITA
heading of 'indirect'. The anechoic chamber method is more troublesome than the reverberant room method, Measuring the sound pressure in an anechoic chamber is more complicated because of the directionality on the source measured. In contrast to this the effect of the directionality can, in principle, be neglected in the reverberant room method, a situation familiar in the measurement of sound power. However, it is known that there is a problem ill the reverberant room method; the measuring procedure to obtain the space-average sound pressure is tedious because the measuring microphone has to be moved from point to point at all measuring frequencies. In order to electrically improve such a procedure the use of a microphone multiplexing system has already been proposed. 1'2 However, automation of the total system for data acquisition including computation has probably not been devised yet. Accordingly, if the total system is automated by making use of the microprocessor, etc., the total time required for not only the data acquisition but also the computation will be considerably reduced. The automated measuring system devised here consists of two parts, one for acquiring data and the other for software calculation of the sound power. All the hardware is synchronously controlled by timing pulses produced by the software, and sequentially the results obtained are printed on recording paper after computation on the acquired data. These sequential processes carried out at every measuring frequency are automatically repeated to the chosen last stage. BASIS OF THE AUTOMATED MEASURING SYSTEM
Discussion relating to reverberant enclosure A requirement in measuring sound power by the reverberant room method is that the sound field in the room is sufficiently diffuse. The reverberant room used has the dimensions 68.7 m 3 in volume and 101-2 m 2 in surface area and the sound diffusion in the room is satisfied at 400 Hz as shown in previous results, 3 The room constants shown in Fig. 1 are used in known values in the computation of sound power as described below. 2.0, 1.8 LEE
~ / ~
'~ 1.6!
J
o
E 1.,i o o
,
I 200
I
I I i ,,,I
500 1½ Frequency (Hz)
2K
Fig. 1. Response of room constant versus frequency.
AUTOMATED MEASURING SYSTEM FOR SOUND POWER MEASUREMENT
447
Discussion on not only the position of the sound source but also that of the measuring microphone is just as important as that on the sound diffusion. The sound source to be measured has to be placed in such a position that the effect of the walls on the source can be neglected. The sound source position is determined on the basis of ISO recommendation (ISO 3741), i.e. the source is placed 1.5 m away from the walls. The microphone positions and the size of the group of measurements has first to be determined. The size can be determined by considering both the spatial variation and the allowable error in the sound pressure. According to Lubman 4 the number of the measuring microphone numbers required is more than 8 when within the one third octave band width centred on 100 Hz, the reverberation time is 10.5 s and the variation in sound pressure level is _ 0-5 dB. In this instance 16 measuring microphones are used. It also has to be noted that the spacing between microphones is more than the one half wavelength at the lowest measuring centre frequency. The lowest centre frequency recommended by ISO for the reverberant room in the present study is 200 Hz. Consequently, the spacing between microphones corresponding to 200 Hz can be determined here. Finally, it is important to discuss whether or not the effect of the direct sound on the microphone signal can be neglected. Consider the smallest distance between the sound source and the microphone to be determined by the aid of the reverberation distance. If the effect of the direct sound on the reverberant sound is postulated to be less than 0-1 dB, their separation, r, can be given by r > 0-4w/~
(m)
(1)
where V is the volume of the reverberant r o o m and T the reverberation time. F r o m eqn. (1) it can be seen that the smallest separation between sound source and the microphone should be given for the upper frequency because the reverberation time is, in general, reduced inversely with increase in frequency. Consequently, the separation in this case is 2.1 m as T = 2.56s at 4 k H z and V = 68.7m 3. On the basis of the discussions described above, the sound source and the measuring microphones in the reverberant room were finally arranged as shown in Fig. 2. The sound source was placed on the floor and the microphones fixed at 1.7 m above the floor. Discussion relating to hardware The general system for sound power measurement consists of a sound level meter, one third octave band filter and level recorder. The system to which this paper refers was newly constructed with integrated circuit devices making possible the automation of data acquisition. The system constructed is shown in the Fig. 3(a) and the level diagram in Fig. 3(c). Each device in the system is sequentially controlled by timing pulses from the microcomputer shown in Fig. 3(b). The response or the characteristics of each device are described below. The microphone amplifiers are composed of field effect transistors, each of them
448
Fig. 2.
T. YANAGISAWA,
W. TSUJITA
Configuration of sound source and measuring microphones are in metres. 0, Sound source; 0, measuring
Mic.Amp
Amp.
FI lter
Fe.45
A/D
in reverberant microphones.
room.
Input
(cl
Fig. 3. Block diagram for automatic sound power measuring system and its level diagram. (a) Measuring instruments in system; (b) microcomputer; (c) level diagram of measuring instruments.
AUTOMATED MEASURING SYSTEM FOR SOUND POWER MEASUREMENT
449
having a constant gain of 33 dB and each having a flat frequency response over 100 Hz. In order to electrically move the measuring microphones from point to point, a microphone multiplexing system is used. The IC device for the multiplexing system is type IH-6116 made by Intersil Co. Ltd, and its chanhel capacity equals the number of measuring microphones as determined in the preceeding section. Each measuring microphone is connected to the multiplexing system device through 5 bits in parallel (4 bits for channel selection and 1 bit for chip enable) from the microcomputer. The amplifier following the multiplexing system is made from an operational amplifier. The gain of the amplifier is fixed at a constant 30 dB. This is so that in the case of the microphone with the lowest sensitivity of - 83 dB, the power term in eqn. (7) described in the following section can be cancelled by the total gain of the amplifiers concerned. The frequency response of the system devised is fiat. An active filter device is adopted for the one third octave band filter and its centre frequency and band width can be freely changed. The IC device for the active filter is type 300 VT-4 made by the Frequency Devices Co. The centre frequency can be varied in the range 100 Hz to 2 kHz by controlled voltages of 0.5 V to 10 V, and the band width corresponding to 23 ~o of the centre frequency can be adjusted with the Q value (Q = 4.3 corresponds to 23 ~o band width). The step-wise output voltage through the digital analog converter set in the computer is used for the controlling voltage. The appearance of the step-wise voltage is synchronised with the operation on the multiplexing system as described above. The IC device is also used to obtain the root mean square value of the signal filtered by the active filter. The IC device is type 440J made by the Analog Devices Co. and the operation is in accordance with the formula for obtaining the root mean square value. The integration time for this device can be controlled by the condenser capacity disconnected from the element block, but the time should be determined by the experiments in practice. D i s c u s s i o n r e l a t i n g to s o f t w a r e
The foundation of the program processing the data acquired by the hardware described in the above section is now discussed. It is based on a well-known formula to be found in the textbooks on room acoustics.5 The sound energy density, E, at any point in an enclosure is given by WQ 4w E = E d + E r - 4~r2 c + R c
(W/m3)
(2)
where E a is the sound energy density of the direct sound, E r the sound energy density of the reverberant sound, W the sound power radiated from the sound source, c the speed of the sound in air, Q the directivity factor of the sound source, r the distance between the sound source and the receiving point, and R the room constant as described before.
450
T. Y A N A G I S A W A , W. TSUJITA
If the first term on the right-hand side in eqn. (2) is less than the second term, eqn. (2) can be approximated to 4W E=~ (W/m 3) (3) On the other hand, the sound energy density can be derived from sound intensity and the sound intensity from the sound pressure. Accordingly, if these relations are connected by eqn. (3), the sound power required can be expressed as a function of the sound pressure as W=
p2R 4pc
(W)
(4)
where P is the rms sound pressure and p the density of air. Furthermore, the sound pressure described in eqn.(4) can be related to the microphone signal as follows:
EI=PIO s/2° (V)
P=EllO -s/2° (/~bar)
or
(5)
The microphone signal, E 1, is amplified by each microphone amplifier and further by another amplifier common to all the microphones as shown in Fig. 3. If the total gain of the microphone channel just described is G dB, the sound pressure given by eqn. (5) can be re-written as a function of the output voltage on the amplifier in the last stage as
P=E210 I-s+~)/2° (/~bar)
or
P=E210 ~-s+a+2°)'2° (Pa)
(6)
where E 2 is the output voltage in the last stage. Substituting P given by eqn. (6) into P given by eqn. (4), the sound power required for the single microphone channel is
W = E210 t-s+a+2°):l°. R/4pc (W)
(7)
However, in order to obtain the sound power as precisely as possible, the sound powers obtained with each microphone channel have to be averaged corresponding to the space-average of sound pressure. At the first measuring frequency required, the output voltage on the last stage in the appropriate channel is squared and then it is multiplied by a constant including only the microphone sensitivity and the total gain. Second, in order to get the mean sound power, the values computed at every channel are summed and then they are averaged multiplying by another constant including the number of the channel. Those computation routes are automatically repeated up to and including the last measuring frequency required. 16
Wf= 2(EiZ)(lO(-S+C+2°'l|°,1 ( 4~ -. ~ ) i=1
(W)
(8)
AUTOMATED MEASURING SYSTEM FOR SOUND POWER MEASUREMENT
451
where the subscript, f, shows each centre frequency on the one third octave band filter and i the number of the microphone channels. The computations of eqn. (8) are carried out with the fixed point numeration in the binary number and the results are printed on the record paper after having translated them to the decimal number with ASCII code.
OPERATION OF AUTOMATED MEASURING SYSTEM
Both the operation of the hardware and the computation of the sound power included in the automated measuring system are carried out through the microcomputer, TMS990/101M made by Texas Instruments Co. Ltd. The flow chart shown in Fig. 4 is used for the operation and the computation. The operation for the measuring system can be simply explained according to the flow chart. At the start of the operation, the lowest measuring centre frequency is set at 100 Hz by the step-wise voltage produced with the digital analog converter in the computer and at the same time, the measuring microphone No. 1 is connected to the
START) lSET CF I JMPXCH ON TON
I ,
I NO
J AID
JMPXCHOFF J, I
Ton=
I
Fig. 4. Flow chart for sound power measuring system. CF, centre frequency of active filter; M P X CH, multiplexer channel; ToN , time holding on multiplexer; Tovr, time holding off multiplexer; *, calculation of ~F '6 E 2 " 10 (s +G+2o)/,o given in eqn.(8); P O W E R CAL, power calculation given by eqn. (8). L..~ i = 1
452
T. YANAGISAWA, W. TSUJITA
relevant channel on the multiplexing system by the 5 bits signal through the parallel I/O port. While the multiplexing system is set in, following the program, the analog signal on the relevant channel through the root mean square device is converted to the digital signal and squared. The squared value is multiplied by the constant including the microphone sensitivity and the total gain of the relevant channel. On the other hand, while the multiplexing system is set off, the corrected digital signal just described is stored in the register in the computer. Those operations are sequentially repeated to the last channel. The values stored are summed and averaged, multiplying by another constant including the reciprocal of the channel number. The mean sound power obtained at 100 Hz is printed. After printing the mean sound power obtained at 100Hz, the next centre frequency is automatically sent through the program, and the same process as described at 100 Hz is carried out. After these processes have been repeated up to and including 2 kHz, given as the last stage, all the processes necessary to obtain sound power are complete.
EXPERIMENTS AND RESULTS FOR AUTOMATED MEASURING SYSTEM
Proper characteristics for the system devised In order to test whether the measuring system itself is correctly operated as planned, experiments are carried out using equivalent microphone signals. First, experiments and results are reported on the dynamic range of the system. The equivalent microphone signal at 1 kHz was applied to the terminal on the active 110
~
1o0
E 90 '- 80
~ 6c •
50
/ lOmV lOOmY lV Input voltage to A I D
u 40
I
I
I
I
/
50
60
70
80
90
10V I
100 110
Theoretical p o w e r spectrum level at l k H z (dB)
Fig. 5.
Correlation diagram between values measured when applying equivalent microphone signals to active filter in system and those calculated separately.
AUTOMATED MEASURING SYSTEM FOR SOUND POWER MEASUREMENT
453
filter. The equivalent input voltages were varied in the range 10 mV to 10 V. The values shown on the ordinate in Fig. 5 are obtained by the designed measuring system and the values on the abscissa in the same figure are calculated separately. From the correlation diagram between the measured and the calculated values shown in Fig. 5, it can be seen that the agreement is good for more than 100 mV and that the difference between them below 100 mV is within 1 dB. The difference may be the result of fixed point numeration in the calculation and it may be improved by adopting a floating point numeration. However, as the computation method by the computer concerned is limited to fixed point numeration, the difference between the values obtained in the range of the lower input level is not really improved. Secondly, the experiments and results are reported for the frequency response of the designed system. The experiments were carried out applying the equivalent microphone signal at 1 mV constant to the input terminal of the appropriate microphone amplifier. The frequency was varied in the range 100 Hz to 2 kHz. The frequency response shown in Fig. 6 is shown as a function of the difference between the values measured and the values calculated separately. From the results shown in the same figure, it can be seen that there are no differences between the measured and theoretical values except for the lower frequency ranges. A very small difference in the lower frequency ranges may result in setting the cut-off frequency of the microphone amplifier to 80 Hz. Hence, from the results described above, it can be said that the measuring system can be used in practice.
cG 0
J ' - ~
;-7, 100
I 2O0
f
I
I ~ ~1~1 500 1k Frequency (Hz)
I
2k
Fig. 6. Frequency response described by difference between values measured when applying equivalent microphone signal to one appropriate microphone amplifier in system and response calculated separately.
Speed-up of operation The operation speed for the measuring system is mainly affected by the integration time in the root mean square circuit and the time holding on and off the multiplexing system. Thus in order to increase the data acquisition speed keeping within the range of the quoted allowable error, both times described above have to be properly determined by experiments on the microphone signals picked up in practice in the field. Experiments for the time combinations given in Table 1 were carried out. In these experiments, the white noise radiated from the
454
X. YANAGISAWA, W. TSUJITA TABLE 1 COMBINATIONS OF INTEGRATION TIME FOR RMS C I R C U I T AND TIMES H O L D I N G ON A N D OFF M U L T I P L E X E R
Integration time (ms)
On-time (ms)
Off-time (ms)
83.5
333 1000
20 900
133-5
333 1000
20 900
loudspeaker with a radius of 5.6cm was used for the sound source, and the 16 electret condenser microphones were used for the measuring ones. Their arrangements in the reverberant room were as in Fig. 2. Moreover, the measurements for each time combination were carried out three times. The experimental results are shown in Figs 7(a) to (d). From the combination of Figs 7(a) and (b) showing the results obtained for the integration time of 83-5 ms, the variation of the values measured on each of the three occasions was found to be small in the frequency ranges less than 250 Hz. However, from Figs 7(c) and (d)
~
75
~
65
1
~55
I
I
(c)
O Q. "1~75
rn
// 100
I
I
I
200
500
11~,
ss 2k, 100
I
I
200
500
I lk
2k
Frequency (Hz) Fig. 7. Band power levels measured with test device. (a) Integration time for R M S circuit is 83.5 ms; times holding on and off multiplexer are 333 ms and 20ms, respectively. (b) Integration time for RMS circuit is 83.5ms; times holding on and off multiplexer are 1000ms and 900ms, respectively. (c) Integration time for RMS circuit is 133.5 ms; times holding on and off multiplexer are 333 ms and 20 ms, respectively. (d) Integration time for R MS circuit is 133.5 ms; times holding on and off multiplexer are 1000ms and 900ms, respectively.
A U T O M A T E D M E A S U R I N G SYSTEM F O R S O U N D P O W E R M E A S U R E M E N T
455
obtained for the integration time of 133.5 ms, the value of each measurement on the three occasions agreed well with each other in the range of the measuring frequency. Consequently, in order to increase the operation speed, keeping the allowable error within 1 dB, the integration time of 133.5 ms should be adopted for the root mean square circuit and the combination of 333 ms and 20 ms is appropriate for holding on and off the multiplexing system. The total time required from data acquisition to processing it is 1 min 40 s with the time combination as determined above.
Compar&on with more usual methods It has already been confirmed that the measuring system itself satisfies the fundamental requirements and is capable of being applied to field measurements. Results of measurements with the system are compared with those obtained with other measurement techniques. First, the results of the frequency analysis obtained with each measurement are compared with white noise radiated from the same loudspeaker as described in the preceding section and with the noise from an electric cleaner. In the measurements with the loudspeaker as sound source, two typical reverberant room and anechoic chamber methods were added to the method developed here. In each case the overall sound pressure level of the white noise was kept at a constant level of 86 dB. The experiments with the described system were carried out with the same arrangements of sound source and measuring microphones as shown in Fig. 2. Measurements with the reverberant room method were also carried out with the same arrangements as in the system described but the measuring instruments were ones generally used and the sound power calculated by the well-known formula. On the other hand, in the measurements with the anechoic chamber method, the sound powers were calculated on the basis of 10 sound pressure levels measured on an hemisphere with a radius of 1 m around the source. The frequency analysis results obtained from the three different measuring methods are shown in Fig. 8. It can be seen that the results obtained are all similar to
rn v -~ 70
n
m•5G100
~ I 200
~ ~ I , t,ll 500 Frequency
, lk
2k
(Hz)
Fig. 8. One third octave band power levels for white noise radiated from loudspeaker, analysed by various methods. - - , Anechoicroom method; , normal reverberationmethod; - - - --, automatic method with test device.
456
T. YANAGISAWA, W. TSUJITA TABLE 2 MEAN
VALUES OF DIFFERENCE OF MEAN V A L U E S MEASURED BY VARIOUS M E T H O D S
Comparison of different methods
Mean L,alues of difference (dB)
Automatic method versus normal reverberation method Automatic method versus anechoic method Normal reverberation method versus anechoic method
0.7 1.3 0.6
one another. In order to review in more detail the differences in the measurements, information on differences between the values obtained is tabulated in Table 2. From the results shown in Table 2 the mean value of the differences obtained by the automated and generally used reverberation techniques is 0-7dB, between the automated and anechoic chamber methods 1.3 dB and between the reverberant and anechoic chamber methods 0.6 dB. Hence it can be said that the reliability of the values obtained by the new system is the same as obtained by other methods of measurement. In the measurements on the electric cleaner the usual reverberation room method was added to the method of the new system. Measurements by the usual method were carried out using only one microphone and the general instrumentation and the sound power calculated on the basis of 16 sound pressure levels. The experimental results are shown in Fig. 9 together with the values calculated with the formula recommended by ISO. From the figure it can be seen that the values obtained agree well in the measuring frequency ranges except for the values at 100Hz and 125 Hz.
80 an v
~> 70 o~ 6o I Q. 13
g so m
100
'
I
200
,
,
I
J ,,,I
500 Frequency
lk
2k
(Hz)
Fig. 9. One third octave band power levels for electric cleaner analysed by various methods. - calculated with the equation recommended bu ISO; - - - , normal reverberation method; - - - automatic method with test device.
AUTOMATED MEASURING SYSTEM FOR SOUND POWER MEASUREMENT
457
CONCLUSION In o r d e r to cut d o w n the time r e q u i r e d for m a n u a l w o r k in o b t a i n i n g s o u n d p o w e r by the r e v e r b e r a t i o n r o o m m e t h o d , an a u t o m a t e d system has been developed. The system consists o f b o t h h a r d w a r e for d a t a acquisitions a n d software for the calculation, a n d the o p e r a t i o n on each p a r t is sequentially c o n t r o l l e d by the a l g o r i t h m p r o p e r for the m e a s u r e m e n t . F r o m e x p e r i m e n t a l results o b t a i n e d in s o u n d power m e a s u r e m e n t s it can be seen that the system has e n o u g h precision a n d reliability for use in practice. So it can be said that the a u t h o r s ' initial goal o f a u t o m a t i n g s o u n d power m e a s u r e m e n t has been achieved.
REFERENCES 1. E. J. WOOTTEN,Instrumentation for space averaging sound pressure levels, J. Sound Vib., 16(1) (1971), pp. 59-69. 2. H. HOGNESTADand O. H. BJOR,A microphone multiplexer for the measurement of the space average of sound levels, Applied Acoustics, 8(1) (1975), pp. 13-25. 3. T. YANAGISAWAand T. UEMURA,Reverberation time measuring system with correlation technique, Applied Acoustics, 14(5) (1981), pp. 377-85. 4. D. LUBMAN,Spatial averaging in sound power measurements, J. Sound Vib., 16(1) (1971), pp. 43-58, 5. For example, see L. L. BERANEK,Acoustics, McGraw-Hill, New York, 1954, p.318.