Speakerphone control circuit having low gain states and method for controlling gain of speakerphone

A speakerphone control circuit is disclosed. The control circuit includes a first electronic switch coupled between the speakerphone microphone and a hybrid circuit coupled to the telephone line and a second electronic switch coupled between the speakerphone loudspeaker and the hybrid circuit coupled to the telephone line. A first sensing circuit for sensing the microphone signal and a second sensing circuit for sensing the received loudspeaker signal are provided. The outputs of the first and second sensing circuits are multiplexed and digitized and fed to a microprocessor controller. In response to the levels of the multiplexed signal, the microprocessor determines the state of the speakerphone from among four possible states, including a first talk state, low gain talk state, first listen state and low gain listen state. The low gain talk and listen states are provided by gating the respective first and second electronic switches on and off to chop the microphone or speaker signal, respectively, thereby attenuating the signals. This allows substantial minimization of spurious sidetone and echo signals and a lessening of harsh clipping effects. Furthermore, the first talk and listen states preferably are only enterable from each other upon passing through the low gain states.

BACKGROUND OF THE INVENTION 
The present invention relates to the field of telephones, and in particular 
to speakerphones. More particularly, the present invention relates to a 
control circuit for a telephone speakerphone wherein the loudspeaker and 
microphone signal channels have low gain states provided for compensating 
for ambient and channel noise levels. 
In a typical speakerphone circuit, both an input transducer or microphone 
and an output transducer or loudspeaker are provided. It is not possible 
for a user to speak and listen to another party at the same time when 
using a speakerphone because both the microphone and loudspeaker cannot be 
enabled at the same time. This is because if both the microphone and 
loudspeaker are enabled at the same time, the speaker output signal would 
be fed back into the microphone, resulting in instability of the system 
due to feedback. Accordingly, circuitry is necessary to disable the 
microphone and enable the loudspeaker during the "listen" mode, and to 
enable the microphone and disable the loudspeaker during the "talk" mode. 
In telephone systems having an optional speakerphone in addition to the 
conventional handset, the use of the speakerphone is often termed "hands 
free" mode. 
In speakerphone applications, it is necessary to be able to compensate for 
ambient noise levels in the environment in which the speakerphone is 
located and noise on the telephone line communication channel. 
Furthermore, it is necessary to be able to distinguish between actual 
voice signals generated by the microphone and "echo" signals present on 
the microphone channel due to the reflection of noise signals on the 
speaker channel onto the microphone channel by system deviations from 
ideal. The same is true of objectionable "sidetone" signals, which are 
reflected noise signals present on the speaker channel which result from 
noise present on the microphone channel. The need to compensate for noise 
is due to the obvious requirement that a speakerphone must be able to 
receive input signals over a wide area, as opposed to the conventional 
handset, wherein only signals spoken directly into the receiver are 
transmitted, and noise is not a problem, and wherein the ear piece is 
closely coupled to the human ear, so that again noise is not a problem. In 
contrast, in speakerphone applications, compensation for noise levels in 
the environment must be made if proper speakerphone operation is to be 
obtained. 
Additionally, prior art speakerphones suffer from objectionable harsh 
clipping of the trailing or leading parts of voice signals when a 
transition is made between "talk" mode and "listen" mode, or vice versa. 
SUMMARY OF THE INVENTION 
Accordingly, it is an object of the present invention to provide an 
improved control circuit for a speakerphone. 
It is furthermore an object of the present invention to provide an improved 
control circuit for a speakerphone having low gain states in the 
microphone and loudspeaker channels to compensate for noise levels in the 
environment in which the speakerphone is located or on the communication 
channel. 
It is furthermore another object of the present invention to provide an 
improved control circuit for a speakerphone which minimizes objectionable 
sidetone and echo signals. 
It is yet still another object of the present invention to provide a 
speakerphone control circuit having low gain "listen" and "talk" states in 
addition to the standard "listen" and "talk" states. 
It is still another object of the present invention to provide a 
speakerphone control circuit which minimizes objectionable clipping of 
voice signals when the speakerphone switches from "talk" to "listen" state 
or vice versa. 
It is yet another object of the present invention to provide a speakerphone 
control circuit wherein the additional low gain states are provided by 
circuitry which chops the microphone or speaker channel signal to thereby 
attenuate the respective signals. 
It is yet still another object of the present invention to provide a 
speakerphone control circuit which is controlled by a dedicated 
microprocessor contained within the speakerphone. 
These and other objects are achieved according to one aspect of the present 
invention by a speakerphone control circuit comprising first electronic 
switch means coupled between an input transducer of the speakerphone and a 
telephone line, second electronic switch means coupled between an output 
transducer of the speakerphone and the telephone line, first means for 
sensing the level of a first signal present at an output from the input 
transducer, second means for sensing the level of a second signal present 
at an input to the output transducer, and control means coupled to the 
first and second sensing means and to the first and second electronic 
switch means for evaluating the first and second signals, the control 
means comprising means for determining whether the speakerphone should be 
in a first talk mode, a first listen mode, a low gain talk mode or a low 
gain listen mode, the first signal being attenuated in the low gain talk 
mode by a first control signal from the control means whereby the first 
electronic switch means is cyclically turned on and off and the second 
signal being attenuated in the low gain listen mode by a second control 
signal from the control means whereby the second electronic switch means 
is cyclically turned on and off. 
According to another aspect of the present invention, these and other 
objects are achieved by a speakerphone control circuit comprising first 
means coupled between an input transducer of the speakerphone and a 
telephone line for selectively providing attenuation to a first signal 
present on an output from the input transducer, second means coupled 
between an output transducer of the speakerphone and the telephone line 
for selectively providing attenuation to a second signal present on an 
input to the output transducer, first means for sensing the level of the 
first signal, second means for sensing the level of the second signal, and 
control means coupled to the first and second sensing means and to the 
first and second selective attenuation means for evaluating the first and 
second signals, the control means comprising means for determining whether 
the speakerphone should be in a first talk mode, a first listen mode, a 
low gain talk mode or a low gain listen mode, the control means selecting 
from among the four modes of operation in dependence on the levels of the 
first and second signals such that the first talk mode and the first 
listen mode are only enterable from each other upon passing through the 
low gain modes. 
Methods for controlling the gain of a speakerphone in accordance with the 
disclosed apparatus are also described. 
Other objects, features and advantages of the present invention will be 
apparent from the description which follows.

DETAILED DESCRIPTION 
With reference now to the drawings, FIG. 1 is a block diagram of the 
speakerphone control circuit according to the invention. The speakerphone 
includes input transducer or microphone 10 and output transducer or 
loudspeaker 20. Microphone 10 is coupled through an electronic switch 30 
and preferably through variable amplifier/attenuator 35 to a hybrid 
circuit 50, known to those skilled in the art, which couples the 
speakerphone to the telephone line. Likewise, speaker 20 is coupled 
through an electronic switch 40 to the hybrid circuit 50. Hybrid circuit 
50 couples signals from microphone 10 to the telephone line for 
transmission and receives signals from the telephone line for coupling to 
loudspeaker 20. A microphone sense signal on line 15 coupled to the 
microphone ahead of the electronic switch 30 is fed to a full wave 
rectifier stage 60, the output of which is fed to a filter stage 70. The 
filter stage 70 has a time constant of approximately 12 milliseconds, and 
provides an approximately d.c. or low frequency output signal 
corresponding to the average value of the microphone output signal. The 
full wave rectifier stage 60 preferably is an ideal rectifier stage 
comprising an operational amplifier, the design of which is well known to 
those skilled in the art. An ideal rectifier stage is necessary due to the 
low voltage levels of the microphone output signals. A simple full wave 
rectifier comprising only semiconductor rectifier diodes would not 
function properly, because the signal levels are often much less than 
typical semiconductor diode thresholds. Similarly, a speaker sense signal 
25 is coupled to the speaker line between the hybrid circuit 50 and 
electronic switch 40. The speaker sense signal 25 is fed to full wave 
rectifier stage 80, which is similar to full wave rectifier stage 60. The 
output of full wave rectifier stage 80 is fed to a filter 90, whose 
characteristics are similar to those of filter 70. The outputs of filters 
70 and 90 are coupled into a 2:1 multiplexer 100. Multiplexer 100 samples 
both filter output signals at 4 millisecond intervals as shown in FIG. 2. 
Both filter output signals preferably are sampled in quick succession at 
100 microsecond spacing as shown in FIG. 2. The multiplexed output signal 
C is then coupled to an analog to digital converter 110, which converts 
the analog samples of the microphone and speaker signals into digital 
form, for processing by a microprocessor 120, which may be dedicated to 
control of the speakerphone circuit. Microprocessor 120, which may be a 
type 8048, receives signals from a central service unit, such as a key 
service unit (KSU) of a key telephone system to which the speakerphone is 
attached via bus 130. Microprocessor 120 provides speaker and microphone 
control signals 140 and 150, for controlling the respective electronic 
switches 40 and 30. Signals 140 and 150 are provided not only to enable 
and disable the loudspeaker and microphone at the appropriate times, but 
also to provide attenuation of the speaker and microphone signals, as will 
be described below. Furthermore, microprocessor 120 provides a multiplexer 
control signal 160 for controlling the sampling rate of multiplexer 100. 
Additionally, a mute/disable handset signal 165 is fed to the handset of 
the telephone station set when the speakerphone option is used to mute the 
handset earpiece to prevent coupling into the speakerphone and to disable 
the handset microphone. An additional signal may be provided to variable 
amplifier/attenuation stage 35, in order further to control the gain or 
attenuation levels of the microphone signal. The gain or attenuation 
levels provided by amplifier 35 may be discrete or continuous, although 
preferably they are discrete levels. 
In addition to enabling or disabling the microphone or loudspeaker 
dependent upon the mode of the speakerphone, speaker control 140 and 
microphone control 150 signals are utilized to attenuate the microphone 
and speaker signals. This is accomplished by providing an oscillating 
control signal on control lines 140 and 150, preferably at 12 kHz or 
higher to avoid audio interference, with a duty cycle of 1 to 4 (1/4 on to 
3/4 off). This will provide approximately 12 db of attenuation of the 
microphone or speaker signals. Accordingly, in order to provide the 
described low gain modes of operation, electronic switches 30 and 40 are 
turned on and off by the oscillating control signal on lines 150 and 140, 
respectively, thus chopping the microphone or speaker signal and providing 
the necessary attenuation. 
FIG. 2 shows the way microprocessor 120 controls the sampling of signals A 
and B provided to multiplexer 100. As shown, the signal C, which is the 
output from the multiplexer, comprises successive samples of both speaker 
sense and microphone sense signals A and B, taken at 4 millisecond 
intervals. Both signals A and B are sampled successively and spaced at 
approximately 100 microseconds apart. 
FIG. 4 is a state diagram illustrating the operation of microprocessor 120 
in controlling the gain of the speakerphone control circuit. As shown in 
FIG. 4, the system has 4 different states, 0, 1, 2 and 4. State 1 is the 
"normal talk" state, i.e., a user is speaking into the microphone of the 
speakerphone and the microphone signal is not undergoing attenuation via 
control signal 150. State 2 is the "normal listen" state, i.e., a user is 
listening at the speakerphone and the speaker signal is not undergoing 
attenuation via control signal 140. States 0 and 4 are low gain states, 
wherein the gain of the microphone or speaker signals are attenuated by 12 
db by gating electronic switches 30 or 40 on and off, at approximately 12 
kHz or higher, with a duty cycle of 1:4 (1/4 on to 3/4 off). State 0 is 
known as the "idle talk" state, as it is a low gain talk mode state and 
the system is "idling," i.e., neither sufficient speaker or microphone 
signals are detected to switch the system to a normal listen or talk 
state. State 4 is known as the "idle listen" state because the 
speakerphone circuit is in a low gain listen mode when in this state. 
Again, in state 4, the system is idling. State 0, the idle talk state, 
however, is only an intermediate state. The speakerphone cannot remain in 
state 0 indefinitely. State 4, on the other hand, is the "quiescent" 
state, and provided sufficient microphone or speaker signals are not 
sensed, the speakerphone can remain indefinitely in state 4. This will be 
explained in more detail below. 
The operation of the system will now be explained with reference to FIG. 4 
using a hypothetical starting situation. Suppose a user is talking into 
the speaker 20 of the speakerphone circuit at sufficient volume. The 
system will then be in state 1, the talk state. Microprocessor 120 
receives a sample of the microphone sense signal via multiplexer 100 and 
A/D converter 110. The microphone sense signal is denoted in FIG. 4 by M. 
If the microphone sense signal M is greater than some minimum value MMIN 
of the microphone sense signal multiplied by some constant, as shown in 
FIG. 4, 1.25, plus a threshold value KMLO as required by the system 
hardware, the speakerphone control circuit will remain in state 1, as 
shown by loop 200. This is indicated by the following equation: 
EQU M&gt;1.25 MMIN+KMLO. (1) 
The minimum value of the microphone sense signal MMIN is determined by the 
minimum value sampled by microprocessor 120 over time. The minimum value 
of the microphone sense signal, for example, during the silent intervals 
between spoken speech, provides a reliable indication of the background 
noise in the environment. This minimum value MMIN is thus the approximate 
value of the noise. 
By determining the minimum value of the microphone sense signal over time, 
if the noise level should increase, the value of MMIN will also increase. 
If the noise level should decrease, a new, lower value of MMIN will be 
obtained. Accordingly, the value MMIN tracks the noise level. This can be 
accomplished in the system software by periodically incrementing the 
minimum value if the noise level increases, and providing a new, lower 
value when it decreases. 
As long as the microphone sense signal is greater than 1.25 MMIN+KMLO, the 
system stays in state 1, thus indicating that a user is talking into the 
microphone. Once the value of M drops below 1.25 MMIN+KMLO, for a period 
of time t0, which is set continuously in state 1 every time a microphone 
sample satisfies the inequality given by (1), the system will move to 
state 0 via line 205. Timeout t0 may be set, for example, at a value equal 
to 204 milliseconds, as shown in FIG. 4. When the system moves from state 
1 to state 0, due to insufficient microphone signal for the time period 
t0, a new timeout t4 is set, preferably to 552 milliseconds. Accordingly, 
a transition from state 0 to state 4 will be made automatically if 
insufficient microphone signal is present to allow a transition back to 
state 1 after timeout t4. Additionally, a blocking delay time b4 is set, 
approximately to a value of 36 milliseconds. The blocking delay time b4, 
set when entering state 0 from state 1, is the amount of time before a 
decision whether a transition will be allowed to state 4 from the 
originating state, in this case state 0, in the absence of a timeout, 
i.e., if a voice interrupt occurs due to a sensed speaker signal. 
Accordingly, once state 0 is reached from state 1, a decision to enter 
state 4 cannot be made until at least 36 milliseconds have elapsed. State 
0 is known as the idle talk state, and is a low gain state generated by 
chopping the microphone signal via the oscillating signal on line 150. An 
attenuation of approximately 12 db is obtained by chopping the signal at 
12 kHz or higher with a duty cycle 1:4 (1/4 on to 3/4 off). As discussed, 
state 0 is an intermediate state. The system cannot remain indefinitely in 
state 0. There are two ways to leave state 0: via a timeout (in this case 
t4), or via a voice interrupt generated by receipt of a speaker or 
microphone signal. 
In state 0, both the microphone and speaker sense signals are evaluated. If 
the microphone sense signal M meets the inquality: 
EQU M.gtoreq.1.25 MMIN+KMHI (2) 
where KMHI is a second threshold value used in determining whether the 
system should go from state 0 to 1, the system will move from state 0 to 
1. KMHI is set at a somewhat higher level than the threshold KMLO in order 
to provide hystersis, i.e., in order to enter a particular state, a higher 
threshold is required than to leave that state. By moving from state 0 to 
state 1, the gain of the system is increased because the user is speaking. 
If the background noise level increases, a larger microphone sample level 
will be required in order to move to state 1, since the minimum microphone 
sample will have increased. Accordingly, the speakerphone adapts to the 
noise level. The user will be required to speak in a louder voice in order 
to switch the speakerphone to the talk state, and the system will not 
erroneously switch over simply because noise levels have increased. 
When the system moves from state 0 to 1, as shown by line 210, a new 
blocking delay b0 is set, preferably at a value of 8 milliseconds. 
Accordingly, when the system reaches state 1 from state 0, a decision to 
return again to state 0 will not be made until a time period of at least 8 
milliseconds has elapsed. Of course, if the microphone signal M satisfies 
the inequality (1) when in state 1, at least the time period given by 
timeout t0 (204 msecs.) must elapse before a return to state 0 may be 
made. The blocking delay b0, however, allows a return to be made to state 
0 after a short time (8 msecs) in the event a noise signal having a rapid 
rate of rise momentarily is sensed, but which is thereafter insufficient 
to maintain the speakerphone in state 1. Accordingly, the 204 msec. timer 
t0 is not set unless "double confirmation" that the sensed signal is a 
voice signal, as determined by inequalities 1 and 2, is made. If 
inequality (2) is satisfied, but inequality (1) is not, the system will 
return to state 0 after the blocking delay time b0 of 8 msecs. Otherwise, 
the 204 msec. timer t0 is set and a timeout must occur to return to state 
0. 
If the microphone signal is below 1.25 MMIN+KMHI when in state 0, for the 
period of time given by timeout t4, which was set when moving from state 1 
to state 0 (552 msecs.), or if the sensed speaker signal S satisfies the 
following inequality: 
EQU S&gt;2M+SMIN+KSSW (3) 
where KSSW is a speaker signal threshold value for the state 0 to 4 
transition, the system will move from state 0 to state 4 via line 220. 
When moving from state 0 to state 4, a new blocking delay b0 is set at 
approximately 16 milliseconds. Accordingly, the system cannot make a 
decision to return to state 0 from state 4 until a 16 millisecond time 
interval has elapsed. The state 0 to state 4 transition inequality (3), 
above, is used because the system should only move from state 0 to state 4 
if the speaker signal is sufficiently greater than the microphone signal, 
indicating that the user is listening and not talking. This is necessary 
because a sidetone signal generated by reflection in the hybrid circuit 50 
of noise signals from the microphone 10 may reach the speaker sense line 
25 from the microphone due to variation of the hybrid circuit 50 from the 
ideal. Accordingly, the system must be able to distinguish between voice 
signals received on the telephone line from the far end and nuisance 
sidetone signals generated by the microphone. The system therefore 
compares the sampled speaker signal with twice the sampled microphone 
signal plus the signal SMIN and the constant KSSW according to inequality 
(3). The signal SMIN is the minimum value of the speaker signal sensed by 
the microprocessor 120, and accordingly, represents the noise level. Since 
the sidetone generated as a result of microphone noise can never be 
greater than twice the sampled microphone signal, if the sensed speaker 
signal is greater than twice the microphone signal plus the noise signal 
and an arbitrary constant, microprocessor 120 determines that the user is 
no longer talking and thus listening and that therefore a transition 
should be made from the idle talk state 0 to idle listen state 4. As shown 
by the state diagram in FIG. 4, the idle listen state 4 or the listen 
state 2 can only be reached from talk state 1 by moving through idle talk 
state 0. Accordingly, if a transition is to be made from talk mode state 1 
to listen mode state 2, or vice versa, both low gain states 0 and 4 must 
be traversed. 
The use of low gain states 0 and 4 is important for two purposes. As 
discussed, the provision of these states allows spurious sidetone and echo 
(the latter being noise signals present on the microphone sense line due 
to reflection of noise signals on the speaker line) signals to be 
distinguished from voice signals. Accordingly, the sidetone signals can be 
reduced by attenuating signals on the microphone channel via switch 30. 
Similarly, as will be described below, echo signals from the speaker 
channel can also be attenuated via switch 40. This provides significant 
advantages as far as further processing of the sampled microphone and 
speaker signals is concerned, since it allows more accurate analog to 
digital conversion because signals of lower dynamic range can be handled, 
for example. 
A second advantage of low gain states 0 and 4 resides in the fact that 
harsh clipping effects are minimized, since the system can only change 
from normal talk to normal listen mode, or vice versa, by traversing the 
two low gain states. This provides a more gradual transition between 
states, thus eliminating harsh, sudden transitions and clipping because 
the respective microphone or speaker signals are allowed to fade between 
talk-listen or listen-talk switching. 
Idle listen state 4 is provided as the quiescent or rest state of the 
system. That is, if a user is not talking into the speakerphone and the 
signal is not in high gain listen state 2 due to the sensed value of the 
speaker signal, the system will remain in idle listen state 4 
indefinitely. Idle listen state 4 is a low gain listen state, wherein the 
speaker signal is rapidly chopped by electronic switch 40 controlled by 
speaker control signal 140. This provides approximately 12 db of 
attenuation due to the gating of electronic switch 40 by a 12 kHz 
oscillating signal having a duty cycle of 1:4. A transition from idle 
listen state 4 to listen state 2 will be made if a sufficient speaker 
signal from the telephone line is sensed. Accordingly, as shown by state 
line 230, if the sensed speaker signal signal S satisfies the following 
inequality: 
EQU S&gt;SMIN+KSHI (4) 
where SMIN is the minimum value of the speaker signal or the noise signal 
and KSHI is a constant threshold for the state 4 to state 0 transition, a 
transition will be made from state 4 to state 2. At the same time that the 
transition is made to state 2, a blocking delay b4 is set at approximately 
8 milliseconds. This prevents a decision to return to state 4 from state 2 
from being made for 8 msecs. Once in state 2, the inequality which applies 
is: 
EQU S&gt;SMIN+KSLO (5) 
The speaker signal must be greater than SMIN, the minimum speaker level or 
noise level, plus a threshold value KSLO in order to stay in state 2 in 
the loop designated 235. KSLO is set at a value below KSHI to provide 
hysteresis. A timeout period t4 of 124 msecs. is set when entering state 2 
once inequality (5) is satisfied. Once the speaker signal decreases below 
the value given by (5), for a time period t4 of approximately 124 
milliseconds, a transition via line 240 will be made from state 2 to state 
4. The timeout t4 set in order to move to state 4 from state 2 is less 
than the timeout set in order to move to state 0 from state 1 (204 
msecs.), because it is considered preferable for it to be easier for a 
person at the speakerphone to interrupt. Accordingly, only a 124 msec. 
timer is provided to leave the listen state, whereas a 204 msec. timeout 
is provided in order to leave talk state 1. Upon the return to state 4, a 
new blocking delay b0, set at approximately 24 milliseconds, will be 
implemented. The blocking delay b0 prevents a decision from being made to 
move to state 0 from state 4 unless 24 milliseconds have elapsed. 
Once in state 4, a transition can be made to state 0 via line 225 if the 
following inequality is satisfied: 
EQU M&gt;2S+MMIN+KMSW (6) 
Accordingly, in order to switch to the low gain talk state 0, the 
microphone sample must be greater than twice the speaker signal plus MMIN, 
the noise level, plus a constant threshold KMSW. As in the state 0 to 4 
transition, the microphone sample must be at least twice as large as the 
speaker sample in order to eliminate confusion with echo signals from the 
speaker line reflected onto the microphone sense line by the hybrid 
circuit 50. Such echo signals might be caused by noise signals present on 
the telephone line, for example. As shown in FIG. 4, a timeout t4 of 100 
msecs. and a blocking delay b4 of 16 msecs. are set when moving from state 
4 to state 0. Accordingly, a timeout to state 4 from state 0 will occur 
after 100 msecs. unless a decision is made to move to state 1 via line 210 
due to satisfaction of inequality (2), and a decision to return to state 4 
due to a voice interrupt can only be made after at least 16 msecs. have 
elapsed (satisfaction of inequality (3)). 
One exception may be provided to having state 4 as the quiescent system 
state. State 4 is normally the state to which the system returns when the 
operator of the speakerphone is not talking, since it is preferable that 
the person with the speakerphone be always able to listen to the person at 
the far end. In this way, the person with the speakerphone knows that he 
is listening to someone and that communications has not been interrupted. 
One feature, however, is often provided in key and branch exchange 
telephone systems called the "call announce" feature, which allows a 
person to call a speakerphone, announce his call with a distinctive tone 
and listen in to the room where the speakerphone is located to determine 
if someone in the room but temporarily away from the immediate vicinity of 
the speakerphone answers. In this instance, it is perferable that the 
speakerphone be in the low gain talk state 0 so that the person calling 
can determine if someone is present to answer. Accordingly, when the "call 
announce" feature is provided and used, the quiescent state of the system 
is changed from state 4 to state 0, i.e., state 4 becomes an intermediate 
state having a timeout period to state 0 and state 0 becomes a stable low 
gain state. 
Table 1 summarizes the 4 states of the present system and the manner in 
which the microphone (MIC) and speaker (SPK) signals are controlled. Table 
2 defines the various signal samples, threshold constants, blocking delays 
and timeouts described above with respect to the state diagram. The 
various thresholds (K) are arbitrary and depend on the attenuation levels 
provided by the hardware, although, as discussed, low and high thresholds 
are used to provide system hysteresis. 
TABLE 1 
______________________________________ 
State MIC SPK 
______________________________________ 
Talking 1 ON, O dB OFF 
Listening 
2 OFF ON, O dB 
Idle Talk 
0 Chopped, -12 dB 
OFF 
Idle Listen 
4 OFF Chopped, -12 dB 
______________________________________ 
TABLE 2 
______________________________________ 
M -- MIC sample 
S -- Speaker sample 
MMIN -- Minimum MIC sample 
SMIN -- Minimum SPK sample 
KMLO -- state 1 self loop threshold 
KMHI -- state 0 to 1 transition threshold 
KMSW -- state 4 to 0 transition threshold 
KSLO -- state 2 self loop threshold 
KSHI -- state 4 to 2 transition threshold 
KSSW -- state 0 to 4 transition threshold 
(b0) e.g. -- Blocking delay before transition 
allowed to switch to state 0. 
(t0) e.g. -- Timeout to state 0. 
______________________________________ 
FIG. 3 illustrates the manner in which additional, non chopped levels of 
gain/attenuation are provided for the microphone signal via 
gain/attenuation stage 35. As shown in FIG. 3, three lines of bus 130 
which couples signals from the central processing unit of the central key 
service unit (KSU) to the station set are connected to three output lines 
of microprocessor 130. The output lines are wired directly together and 
comprise a wired OR connection. Accordingly, signals for controlling the 
gain of the microphone signal from the KSU CPU or signals from 
microprocessor 130 can control the gain of stage 35. Stage 35 might 
comprise, for example, a multiplexer, such as a CMOS type 4051 
multiplexer, which selects various gains or levels of attenuation from a 
resistor bank, depending upon the address inputs to the multiplexer. These 
gains/attenuations are fixed gains/attenuations, and are provided in 
addition to the attenuation provided by chopping the microphone signal via 
electronic switch 30. 
In the foregoing specification, the invention has been described with 
reference to a specific exemplary embodiment thereof. It will, however, be 
evident that various modifications and changes may be made thereunto 
without departing from the broader spirit and scope of the invention as 
set forth in the appended claims. The specification and drawing are, 
accordingly, to be regarded in an illustrative rather than in a 
restrictive sense.