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FIELD OF THE INVENTION The invention disclosed herein is an audio amplification system
characterized by protecting human hearing that may be exposed to audio levels beyond
acceptable levels from damage.
BACKGROUND OF THE INVENTION It is common practice for a performer to drive a speaker at
an audio output level which may damage the listener's hearing in an indoor or outdoor music
concert. Government health officials have established noise level restrictions in order to protect
the health of workers' hearing. The United States Government Occupational Safety and Health
Administration (OSHA), the American National Standards Institute (ANSI) and the International
Standards Institute (ISO) have established standards to protect employees from the effects of
exposure to noise. The OSHA standard is 29 C.I. F. R. It is published in 1919.95. Basically, the
standard shows that there is an inverse relationship between the number of hours exposed to
noise per day and the level of noise or speech expressed in decibels. For example, the safety limit
for employees exposed to an acoustic level of 95 decibels is 4 hours, while the limit of exposure
for an acoustic level of 110 decibels is 1/2 hour.
The standard also recognizes that not only the overall sound energy or level but also the
frequency components of sound or noise must be taken into account. In general, the sound in the
higher part of the audio frequency range is more energetic and can cause more damage at some
decibel level than the lower frequency sound. For example, very high decibel levels are
acceptable for sounds close to the low frequency part of the audible range of around 20 Hertz,
while for sounds with the same energy content at high frequencies above 4000 Hertz, It can not
be accepted unless it is a considerably lower decibel level. It is known that the sound levels in the
high frequency part of the spectrum, which often occur in so-called rock music concerts, cause
permanent damage to the listener's hearing. It is also well known that the cumulative value of the
loudness in many concerts exceeds the tolerance for the total amount set by the various
Besides taking steps to protect the hearing of each and every one of the audience who come to
listen to the concert paying the entrance fee, like a musician, an operator of an audio amplifier, a
staff member in the hall to crack down on illegal or disorderly behavior The possibility of damage
to the hearing of the people involved in the concert must also be taken into consideration. An
employee-mounted audio dosimeter that is exposed to any sound emanating from the speakers
indicates that the volume during the concert is quite excessive. People who operate amplification
equipment regularly at concerts are unaware but suffer severe hearing loss. Because their
hearing is below normal, their hearing defects tend to increase the gain of the amplifier to a level
they consider acceptable, which results in permanent hearing loss in the listener.
Systems are known that automatically control the output volume from a set of loudspeakers. U.S.
Pat. Nos. 2,338,551, 2,468,205, 26,169,71, 3,009,999, 4,254,303 and 4,306,115 are examples.
These patents adjust the sound output level from the speaker to compensate for the background
noise level. In other words, if the background noise is high, then the gain of the amplifier system
is increased to exceed the noise level so that useful speech can be heard. If the noise level is
reduced, the acoustic output from the speaker is automatically reduced. Another U.S. Pat. No.
4,583,245 discloses a system for protecting an audio output speaker from possible damage by
driving the speaker with an acoustic wave containing a frequency that causes the speaker
diaphragm to carry a larger excursion designed to withstand the diaphragm. ing. Thus, in this
patent, the input acoustic spectrum from the microphone is separated into component frequency
bands. In this way, the loudspeaker is not damaged as the frequency at which a particular
loudspeaker is optimized is sent to it. The listener's ear reconstructs the acoustic spectrum
emitted by the individual speakers.
SUMMARY OF THE INVENTION An automatic acoustic control system that automatically adapts
to existing standards that set limits on impulsive or instantaneous acoustic levels and cumulative
acoustic volume until the invention disclosed herein is made. There was no one to offer.
SUMMARY OF THE INVENTION In accordance with the present invention, a novel audio
amplification system automatically, instantaneously and selectively attenuates frequency bands
at the decibel level above those permitted by the standard, and further It is characterized in that
the entire frequency is attenuated to a predetermined audible range when the cumulative energy
of the sound exceeds a predetermined level.
According to an exemplary embodiment of the present invention, a microphone is placed in the
area of the listener to sample the sound from the speakers reaching the listener's ear. The sound
representing the signal returned from the microphone is supplied to a microcontroller with
resident firmware for performing a fast Fourier transform (FFT), and the amplitude level is the
intensity or level of the frequency within a series of frequency bands within the audible range
Generates a signal representing In other words, the acoustic signal is transformed from the time
domain to the frequency domain. This attenuates any frequency band beyond the limits set and
stored by the standard and allows integration of the sound level over time, so that the entire
spectrum can be attenuated if the stored volume limit is exceeded . In the preferred embodiment,
the frequency converted signals are provided in bursts to a microcomputer programmed to
control the input and output of the latches provided for each frequency band to be controlled. By
way of example and not limitation, 16 or more such latches are provided. The microcomputer
places the various frequencies in order with respect to the latches, the output of each latch being
the input of the notch filter or attenuator. The attenuator performs any amount of attenuation
and does not attenuate at all for the specified specified frequency band, depending on whether
the decibel level is higher or lower than the stored tolerance reference limit. In other words, as
many notch filters or attenuators are provided as there are frequency bands in which the
acoustic spectrum is analyzed or split. In the exemplary embodiment, sixteen notch filters are
provided, each associated with each latch. The notch filter attenuators are essentially connected
in series in the exemplary embodiment, and each is provided with switches for attenuating and
not attenuating a particular frequency step by step in response to the control signal generated by
the microcomputer. Each notch filter has an electronic shunt switch that operates to bypass a
particular notch filter designated for a particular frequency band for which the corresponding
signal level is considered to be below an acceptable level or upper limit. If the full frequency
band in the audio spectrum being generated by the performer does not exceed the stored limit,
then all changeover switches keep the bypassing of all notch filters so that the sound goes
directly to the amplifier, It is then output to a speaker that gives the audience the sound.
Further in accordance with the present invention, an integrator is provided which functions to
integrate and display the accumulated volume. The entire acoustic frequency spectrum is
attenuated when the cumulative amount of sound tends to break the stored reference limit.
When exceeding the allowable cumulative amount, the system protects the unwanted people
from permanent hearing loss by shutting off the amplifier and stopping the output from the
It will be clear in the following description of an exemplary embodiment of the invention, given
with reference to the drawings, that the above-mentioned features of the control system are
typical acoustic amplification system installed at a concert venue played by a musician who is
usually more appealing to young people than to elderly people.
Of course, this novel hearing protection device can also be used in sound systems other than
amplifying music. Music from speakers at concerts is routinely loud enough to cause temporary
and often permanent hearing loss. In the United States, the Occupational Safety and Health
Administration (hereinafter referred to as OSHA), and internationally the ISO, has already
developed standards to control the time at which sound of a particular frequency and decibel
level can be allowed.
These agencies also set standards for broad spectrum sound levels where the time-weighted
average of permitted sound levels is usually inversely proportional to the decibel level of the
In the installation of FIG. 1, there are four loudspeaker towers 20, 21, 22, 23 each supporting
four loudspeakers, the acoustic output of which is directed to a listener not shown.
Sounds begin at the microphones labeled 24 and 25 which intercept the sounds emanating from
the instrument 26 and the drum 27. Musicians playing these instruments are exposed to strong
sound with the audience. Each microphone intercepting the voice of one or more instruments or
singers outputs an analog signal to a preamplifier. In this particular example, eight identical
preamplifiers are used. The illustrated preamplifier is labeled 28. The analog signal from the preamplifier is delivered to mixers 29, which output on each of coaxial lines 30-33 a broad spectrum
analog signal that is a composite of the sounds produced by the various instruments and singers.
According to the invention, a control device is provided for each loudspeaker tower. In the
particular example with four speaker towers, four controllers 34-37 are used. Two microphones,
such as 38 and 39, are placed in the audience to detect or sample the sound level emitted by the
speakers and heard by the audience.
A time domain analog signal representative of the sound intercepted by the microphones 38 and
39 is fed back to the controller 34. As will be described in more detail below, the exemplary
controller 34 has the ability to analyze the frequency components of the feedback analog signal
representing the acoustic output of the adjacent loudspeakers. More specifically, the controller
34-37 digitizes the audio signal and converts it from the time domain to the frequency domain
and then performs fast Fourier transform. In other words, a determination is made as to the
intensity of the frequencies in the series of frequency bands that includes the entire feedback
acoustic spectrum.
The controller has the ability to compare each frequency band to the tolerance levels stored for
each particular band. As will be described in detail later, each control unit is provided with means
for attenuating any frequency band of the analog signal representing the frequency beyond the
tolerance set by the OSHA and stored in the control unit. Only the frequency band exceeds the
limit and is attenuated by the controller. If there is no frequency band beyond tolerance, the
sound is simply sent to the loudspeaker without attenuation. The output from a controller, such
as controller 34, is input to an amplifier 49 which drives a loudspeaker 20 mounted on one
tower. The controllers 35, 36 and 37 function in the same way and do not need to be described
at this point in time.
The new system depicted in FIG. 1 also monitors the cumulative amount of sound to which the
audience is exposed. When the cumulative amount of sound exceeds the stored limit, the entire
spectral band is automatically attenuated. In some cases, the speaker may be turned off
completely to meet the stored OSHA or ISO cumulative mass standards. In order to achieve this
function, the installation of FIG. 1 has a block 40 called central integrator. This will be described
in detail later, but includes a device for measuring the accumulated amount and continuously
displaying the accumulated amount. In this particular example, there are four feedback
microphones 41-44 that pick up the sound from the speakers and feed the corresponding analog
signal back to the central integrator 40.
One of the output lines 45-48 from the central integrator (CIU) controls the control signal to
provide frequency band attenuation in accordance with the requirement to meet the cumulative
volume limits allowed by OSHA or ISO. Supply to
An exemplary controller 34 is described with reference to FIGS. 2A and 2B.
In the installation of FIG. 1, four control devices are used because four sets of different purpose
speakers are used. Depending on the installation, more than four controllers may be required, or
only one controller may be required.
Feedback microphones 38 and 39 for one controller are depicted in the upper left part of FIG.
2A. The microphone 38 detects the entire audio output spectrum from the loudspeakers attached
to one tower and supplies the corresponding analog signal to the signal conditioning network
consisting of the pre-amplifier 50, the 16 Hertz high-pass filter 51 and the musical amplifier 52. .
These components are conventional and commercially available. Audio frequencies below 16
hertz are filtered because they are below the human ear's hearing ability. Musical instrument
amplifier 52 performs adjustable offset and adjustable gain signal adjustment operations to scale
the feedback signal appropriately. An Analog Devices AD-521 amplifier 52 or equivalent is used.
A similar signal conditioning network consisting of a pre-amplifier 53, a filter 54 and an
instrument amplifier 55 and another microphone 39 are in a standby state to be used when the
redundant or other signal conditioning network becomes inoperable .
The full spectrum adjusted analog feedback signal is input via line 56 to an analog multiplexer
57 represented by a rectangle comprising two semiconductor switches 58 and 59. Intersil
IH5208 multiplexer or its equivalent is used. Only one of the switches 58 and 59 normally
operates since only one signal conditioning network operates at a time. The analog multiplexer
switch 58 is operated by an I / O port line 60 leading from the control line or microcontroller 61.
After passing through the analog multiplexer 57, the analog audio signal is input to the sample
and hold amplifier 62. The time constant of this sample and hold amplifier is determined by the
values of capacitor 63 and potentiometer 64. A teledyne filter brick TP4856 sample and hold
amplifier is often used, but equivalent amplifiers can be used. A control signal for timing the
sample and hold circuit 62 is provided by the microcontroller 61 via line 65. The entire analog
speech spectrum is sampled continuously.
The output signals from the sample and hold amplifier are: microcontroller 61, microcomputer
70, attenuation data buffer 71, attenuator address decoder / demultiplexer 72, inversion buffers
73, 74, 75, 16 latches LT1-LT16, and 16 Are processed in a signal processing circuit consisting
of notch filter attenuators AF1-AF16 connected in series and each having an electronic shunt or
bypass switch. These elements constitute one of the control units such as any one of the control
units 34-37 of FIG.
The decoder / demultiplexer 72 is a Signetics 74L 5154 or equivalent. Attenuator data buffer 71
is a 74 LS 241 buffer, and inverting buffers 73-75 are 74 LS 240. The microcomputer 70 is
68705 U3 manufactured by Motorola. The microcontroller 61 used in the practical embodiment
is an Intel 8096, but an equivalent controller could be used.
The microcontroller performs a fast Fourier transform (FFT) analysis on the conditioned
microphone feedback signal to generate a time domain analog signal in a discrete frequency
domain spectrum, in other words, a discrete band of frequency domain spectrum. It has resident
firmware that converts it to frequency amplitude. In the illustrated embodiment, the conversion
is performed on 16 frequency domain bands. The spectrum is periodically communicated to the
attenuator control network via serial lines C1, C2 and C3. The FFT analysis firmware may be
instructed to split the frequency spectrum into more than 16 eg 32 if necessary. Several light
emitting diodes 67 are provided in conjunction with the microcontroller 61 to indicate the state
of the program as is conventional. The microcontroller 61 with FFT firmware mounted on the
substrate and analog-to-digital (A / D) conversion means 66 on the substrate is a commercially
available integrated circuit. A practical example of an audio dosage control system is described in
Intel's Application Note AP-275, "Embedded Control Application Handbook", with a copyright
date of 1988, in Intel's 8096 Integration A circuit microcontroller is used.
The frequency component output from the microcontroller 61 in the signal processing means is
input to the microcomputer 70 which is the first stage of the attenuator network. The input to
microcomputer 70 is a series of sixteen digital values corresponding to the amplitudes of the
sixteen half octave FFT spectra. When the FFT analysis microcontroller 61 calculates frequency
spectrum values, the microcontroller 61 transmits high bursts of spectrum values via lines C1C3, which are then inactive until the next burst. The calculation of the FFT and the transmission
of the output occur at the same time as the data is input to the microcontroller 61. Real-time
analysis is performed and no data is lost. All data input from the active microphone 38 or the
standby microphone 39 in use are subjected to the FFT.
The microcomputer 70 stores in itself 16 values which represent the allowed decibel levels or
intensities of the 16 different frequency bands. Preferentially stored data is based on tolerance
levels set by OSHA or ISO. There are different frequency intensity values for each of the
frequency bands. This is because some of the frequencies have been found to cause greater
damage to the human ear than others. Therefore, the tolerance level limit for the most harmful
frequencies is set lower than the limits of the other frequencies. The microcomputer 70 receives
spectral data from the FFT analysis microcontroller 61 in serial format. Firmware resident in
microcomputer 70 compares the individual spectral band levels to the allowable shock levels
stored in memory. As will be shown later, all spectral bands with amplitudes exceeding
acceptable levels are automatically attenuated or filtered.
In FIG. 2A, the microcomputer 70 makes a comparison with the stored allowed frequency
intensity levels in milliseconds or less. Digital data that corresponds to the degree to which the
exceeded band is to be attenuated in strength to fall below the limit according to the OSHA
standard if the instantaneous frequency band level exceeds the stored limit for each band
Represented by Attenuation control data is coupled to sixteen latches LT1-LT16 through data
buffer 71. Also, three inverting buffers 73, 74 and 75 are provided. The microcomputer 70
outputs an address to four lines 76. These addresses are input to the decoder / demultiplexer
(Demux) 72. When Demux 72 is enabled by a signal on line 77, latches LT1-LT16 of FIG. 2B are
addressed. Each latch handles control data for one frequency band. Sixteen address lines are
drawn from the inversion buffers 73, 74 and 75. These sixteen address lines are connected to the
sixteen latches to allow attenuator control data to be addressed to the individual frequency band
Five bits of data are obtained. This data is transmitted via the 5-bit parallel bus to any of the
attenuators AF1-AF16 addressed. The data coupled through buffer 71 represents the amount
that any particular frequency band must be attenuated to reduce its strength or level below the
stored tolerance limit for that frequency.
As shown in FIG. 2B, five bits of data can be transmitted to the latch when addressed via buses
79, 80, 81, 82 and 83. The value of the digital data indicates the amount by which a particular
frequency band should be attenuated in order to fall below the tolerance when its intensity is
above the limit. For this purpose, a notch filter attenuator is provided as generally indicated by
the reference AF4. Only notch filter attenuator AF4 is depicted in detail in FIG. 2B, which is
controlled by the data output from latch LT4. The other notch filter attenuators AF1-AF3 and
AF5-AF16 are similar to AF4 and are connected in series and are therefore drawn as a block.
However, each attenuator is provided with bypass or shunt switch means such as two switches
93, 94 in attenuator AF4 and one symbolic switch means labeled 112 on AF4.
The analog audio signal from the mixer 29 of FIG. 1 is input to the input terminals 88 and 89 of
the series first attenuator AF4 arranged halfway to the loudspeaker. The frequency band that
does not require attenuation simply bypasses the filter in that particular band by using electronic
bypass or shunt switch means as conductor means (described below).
Each of the controllers 34-37 depicted in FIG. 1 includes sixteen latches LT1-LT16 and a
corresponding number of attenuators or notch filters AF1-AF16. Each attenuator or filter is tuned
to attenuate a particular frequency band within each of the sixteen. Attenuation of a particular
frequency is initiated by addressing the designated latch to that frequency to control the
individual attenuation filters via decoder / demultiplexer 72, and then control the amount of
attenuation with a filter such as AF4 Attenuation and enable codes are sent to attenuator data
bus 78 to strobe each latch in the attenuator latch network. The attenuator is progressively
removed over a variable time for reset as the excess frequency decreases. The attenuation filter
network consists of programmable active second notch filters AF1-AF16 using conventional dual
operational amplifiers 86 and 87.
As previously indicated, for example, the unconverted composite audio signal generated by the
instrument is supplied from the mixer 29 of FIG. 1 to the input terminals 88 and 89 of each
notch filter. Notch filters remove all frequencies above and below the frequency band to which
they are tuned. In this way, a selected frequency band of the frequency bands supplied to the
input terminals 88 and 89 of each of the sixteen notch filters AF1-AF16 in the controller can be
removed or attenuated.
Attenuation is performed in stages by the analog switch element 90 of the integrated circuit to
avoid noticeable changes in the music when the frequency is to be attenuated. Analog switch
element 90 is shown in a typical frequency band attenuation filter AF4 as four mechanical
switches 91 connected in series with a resistor numbered 92. The damping control resistor 92 is
switched by an intersil DG 211 CJ or equivalent analog switch element 90.
Two single pole double throw solid state analog detours or shunt switches 93 and 94 illustrated
in the attenuation filter AF4 are provided by the signal on line FB4 to switch the audio input
signal to the analog switch 93 from the illustrated position. It is controlled to switch to the line
96 leading to the terminal 95 and also to the terminal 97 of the other analog switch 94. These
two switches are simultaneously operated by the signal on line FB4 whenever operation is
required. When these are closed, the particular frequency to which the notch filter is tuned is at
an intensity or level below the allowable limit if no attenuation is required. In such a case, the
audio signal bypasses the notch filter. If it is necessary to attenuate a particular frequency band,
the signal on lines AS0-AS3 from latch LT4 shown is more likely than a human ear to detect
changes in the intensity or level of the frequency to which each notch filter is tuned. At high
speed, it is forwardly supplied to the analog switch 90 in the notch filter AF4.
The notch filter is essentially conventional and includes a divider circuit consisting of feedback
resistors 100 and 101, coupling capacitors 102 and 103, resistors 104 and 105 and an input
resistor 106, and an amplifier 107 for musical instruments. This amplifier may be an Analog
Devices AD 521 or equivalent. Filter offset and gain adjustments are obtained by potentiometers
108 and 109 respectively.
The output lines 98 and 99 from each notch filter are routed to the output terminals 110 and
111 connected to the inputs of the amplifier as labeled 49 in FIG. 1, which amplifier 49 is from
the controller 34 of FIG. It receives a signal and feeds it to four speakers attached to one of the
speaker towers.
Heretofore, it has been described how the individual frequency bands are attenuated in each
control unit.
Gradual decay occurs in microseconds as required and is not noticed by the audience. Similarly, if
the strength or amplitude of a particular frequency falls below the stored tolerance limit, the
attenuation will be phased out. The computer 70 shown in FIG. 2A configures several software
switches whenever one of the bands exceeds tolerance.
The software initiates a special scanning routine that monitors the effect of the attenuation on
feedback from the pickup microphone 38 or 39 during operation. First, the computer applies a
filter attenuation of around 25%. If not enough to drop below an acceptable level, the computer
continues to make more attenuation. For example, assume that the instantaneous level of a
frequency band is 113 dB, which is much higher than the 95 dB allowed by the OSHA for this
frequency. In this case, the computer operates the first four analog switches 91 in the switching
circuit 90 and a time delay for feedback of about 109 decibels for that frequency is detected.
The computer does not stop the operation of the attenuator since it is obvious that the attenuator
will rise again if the inputs are the same. Thus, the computer requires that the level be reduced
to, for example, 10 @ 7 decibels before removing the attenuation. In the illustrated embodiment,
the step reduction is at a rate of 25% and 5% in a few seconds. The attenuation is gradually
reduced. However, if the signal at that frequency again reaches the limit at which the attenuation
is reduced, the computer also increases the attenuation again.
The loop responds faster than the ear detects. Even if the engineers operating the mixers and
amplifiers of the sound system suddenly gain, the audience will not hear any disturbing levels of
sound. The first part of the waveform originates from the loudspeaker tower, but attenuation of
that frequency takes place in microseconds before the waveform is complete at that frequency.
In FIG. 2B, another switching and resistor network, indicated generally by the reference numeral
115, is shown. This network is used to control and limit the allowable cumulative volume over a
given time, which will now be described with reference to FIG.
FIG. 3 shows a central integrator 40 representing a block 40 which is likewise displayed in the
system diagram of FIG. In FIG. 3, the pick-up microphones 41-44 depicted in FIG. 1 are repeated.
In this embodiment, there is one microphone for each of the speaker groups 20-23. Each
microphone supplies an analog signal representing the sound picked up from the speaker to each
signal conditioning system consisting of a pre-amplifier 126, a high pass filter 127 and an
instrument amplifier 128.
These four systems are identical and basically identical to the signal conditioning system at the
top of FIG. 2A. For example, a typical microphone 41 supplies an analog input signal to a
preamplifier 126, the output of which is fed to a 16 Hertz high pass filter 127, and the output of
the high pass filter 127 is fed to an instrument amplifier 128. It becomes an input.
All four conditioning circuits receive audio signals from the acoustic sampling feedback
microphones 41-44. Four substantially similar analog signals from the conditioning circuit are
provided via lines V1-V4 to the input of an analog multiplexer (MUX) 129 consisting of Intersil
IH 5208 or equivalent. MUX 129 includes four solid state switches represented by the numeral
130. The microcontroller 131 addresses the individual switches 130 of MUX 129 via address
select lines A0 and A1. The analog signals generated by the microphones are arranged in order
by the MUX 129 and input to a sample and hold amplifier 132 consisting of a TP4 856 or
equivalent from Teledyne Filbrick. The time constant or sampling rate of sample and hold
amplifier 132 is controlled by capacitor 133 and potentiometer 134.
The microcontroller 131 is similar in configuration and function to the controller of FIG. As in the
previous case, the microcontroller 131 performs a Fast Fourier Transform (FFT) analysis on the
conditioned microphone feedback signal to transform the time domain analog signal into a
discrete frequency domain spectrum band. The microcontroller 131 is an Intel 8096 or
The analog input signal to the microcontroller 131 is converted for processing into a digital
signal, as suggested by the block labeled A / D and labeled A / D. The microcontroller 131
controls the sampling rate of the sample and hold unit 132 by means of the signal transmitted by
the line 137. A group of light emitting diodes and limiting resistors 138 provide an indication of
the programmed state of the microcontroller.
The output serial lines C1-C3 derived from the central integrator transmit the converted
frequency spectrum to the input of the microcomputer 139, as well as occurring in the
controllers already described. A Motorola MC68705U3 or equivalent microcomputer 139 is
used. Feeding the central integrator 40 from the four microphones 41-44 provides an average
acoustic level in the listener area rather than from a single point. The four microphones also
provide redundancy. If one fails or breaks down, the software in microcontroller 131 disconnects
the speaker by holding the appropriate switch in MUX 129 open.
The microcomputer 139 determines the continuous sum or integration of equal amounts for
each frequency band. The weighting factors are stored in a memory implemented in the
computer. The computer accesses the weighting factor for each band from the internal memory.
The computer multiplies the band with the individual weighting factors, as the relative
impairment to hearing at a particular frequency is different. The weighted volumes are summed
and stored as integral volume in a computer implemented register.
Control of event length and initiation is also programmed into microcomputer 139 using
switches 140, 141 and 142. The volume is continuously stored in non-volatile RAM memory
device (NVRAM) 143. This memory is a DS1220 memory or equivalent from Dallas
Semiconductors. A key operated switch 144 is provided to inhibit unauthorized adjustment of the
total amount allowed. Since the memory 143 is non-volatile, the stored accumulated amount is
held for any period of time, so if there is a power failure during the audio event, the memory will
start to be added to the accumulated amount at the time of power failure.
The 8-bit data bus 145 is output from the microcomputer 139. This data bus provides signals to
control the digital display. Display group 150 indicates the predetermined duration of the event
entered for display by using switches 140 and 141. Another group of displays 151 indicates the
amount of time that has elapsed since the start of a concert or other audio event. Yet another
group of displays 152 is continually updated for the duration of the event to display the
cumulative amount to date. Another group of displays 153 displays the cumulative amount of
audio that can be tolerated within a predetermined event duration. The signals on lines 154 and
155 strobe display decoders 156 and 157. Another line 158 strobes a latch element 159 such as
Intersil 74 LS 373 to update the attenuation level indicator 160. The decoder 161 controls bar
graph display.
The program of the computer 139 is essentially a software representation of the OSHA or ISO
volume standard. This standard allows for a certain equivalent number of decibel times, ie, timeweighted average (TWA), for various frequencies and the acoustic spectrum as a whole. As the
allowable time-weighted averages approach their expected limits, increase the amount of decay
of all controllers 34-37 or any of the controllers in the system to limit them within the duration
of the programmed event. It must be limited to If the time-weighted average is exhausted before
the end of the scheduled event, the audio amplification system must be completely shut off and
adapted to the programmed OSHA total sound for the corresponding event duration .
Attenuator control signals which control the reduction of the total acoustic volume or the
interruption of the audio amplification system appear at the lower right terminals A0-A3 of FIG.
A cable such as 45-47 in FIG. 1 directs these signals from central integrator 40 to each controller
34-37, the control signal being the attenuator or volume controller 115 in part B of FIG. 2B. Are
coupled to corresponding terminals A0-A3 of the second damping means in each controller.
Attenuator or volume controller 115 is a solid state switch circuit symbolized by resistors 116119 in series with mechanical switches 120-123, respectively. Intersil DG211 CJ or equivalent 4-
pole solid state switch is used in the practical embodiment. The network 115 selectively switches
the attenuation register coupled to the decrease register 124 according to the attenuation code
signal input to the control signal terminals A1 to A4, thereby causing the audio output signal to
be led to the output terminals 110 and 111. Provides overall full band attenuation.
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