Automatic gain control of transponded supervisory audio tone

The process of the present invention encompasses a method for automatically controlling the gain of a transponded modulation signal. The process receives a demodulated signal, detects whether SAT is present, and transmits the SAT. Before transmission, the transmitted SAT is adjusted for the gain of the modulator and for the gain of the receiver.

FIELD OF THE INVENTION 
The present invention relates generally to the field of communications and 
particularly to automatic gain control of a signal. 
BACKGROUND OF THE INVENTION 
Landline telephony uses supervision to detect changes in the switch-hook 
state caused by the telephone user. Mobile telephone supervision also 
performs this process but must also ensure that adequate RF signal 
strength and interference protection is maintained. This is accomplished 
by the supervisory audio tone (SAT), a continuous out-of-band modulated 
signal. 
Three SAT signals are used in the United States cellular system, AMPS. 
These SAT signals are at 5970 Hz, 6000 Hz, and 6030 Hz. Only one of these 
frequencies is employed at a given cell site. 
The SAT operates by the mobile unit receiving the SAT from the base station 
and transponding it back to close the loop. The base station looks for the 
return of the specific SAT it sent out. If another SAT is returned, the 
cell interprets this as the call between the mobile and the cell being 
corrupted by interference. 
When a mobile receives a signal from the base station, it detects whether 
SAT is present. Based on this frequency, the mobile generates its own SAT 
and transmits it back to the base station. This method requires the mobile 
to perform a large number of million instructions per second (MIPS) in 
order to detect the received SAT and generate the transmitted SAT. 
Consumers are demanding smaller cellular telephones for greater 
portability. To reduce the size of the telephones, the number of parts in 
the telephone must be reduced. This can be accomplished by performing many 
of the telephone's functions in a digital signal processor (DSP). This, in 
effect, replaces a number of integrated circuits with a single DSP that 
performs the same function as the replaced ICs. 
Replacing the present SAT detection and generation circuits with a DSP 
would require a very MIPS intensive process. This would take up time the 
DSP could be used for other tasks. Additionally, the received SAT must be 
adjusted for gain differences caused by the mobile's hardware before it is 
transmitted back to the base station. This would take additional time away 
from the DSP. There is a resulting need for a simple process to detect SAT 
and adjust its gain. 
SUMMARY OF THE INVENTION 
The process of the present invention encompasses a method for automatically 
controlling the gain of a transmitted modulation signal. This control is 
based on a received demodulated signal. The process filters the received 
signal to produce a filtered signal. This filtered signal is processed by 
an autocorrelation function to generate a plurality of autocorrelation 
values. A desired energy of the filtered signal is divided by at least one 
of the autocorrelation values to produce a preliminary gain adjustment 
signal. A further gain is derived from the preliminary gain adjustment 
signal by taking the square root of the preliminary gain adjustment 
signal. The derived gain is scaled in response to the gain of the 
modulator that modulates the transmitted signal. This device gain is 
filtered to produce a filtered gain signal. The transmitted signal is 
generated in response to the filtered gain signal and the filtered signal.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT 
The process of the present invention enables a SAT detection and automatic 
gain control (AGC) process to be incorporated into a DSP without requiring 
a large amount of processing time. This process is illustrated in the 
block diagram of FIG. 1. 
The process begins by the received signal being demodulated (102). This 
demodulation process produces a signal whose rms value is affected by both 
the received signal and the gain of the demodulator. This affect is 
removed by multiplying (103) the demodulated signal by the value N. N is 
chosen so that for a known received signal, the signal at the input to the 
bandpass filter (104) is known; i.e., N is adjusted for each 
radiotelephone to remove the effects of demodulator gain variance from 
radiotelephone to radiotelephone. 
To illustrate this adjustment, one embodiment of the present invention 
might have a received signal with an approximate 2.0 kHz deviation at a 
6.0 kHz rate. The actual received signal may be other than exactly 2.0 kHz 
deviation, but the transmitted signal needs to be exactly, or 
substantially close to a 2.0 kHz deviation independent of the exact 
received signal deviation. The transponded received signal, therefore, 
needs to have its gain adjusted as the demodulated received signal varies. 
For a received signal with a 2.0 kHz deviation at a 6.0 kHz rate, a 
demodulated signal, s(t) is produced. This signal is then sampled and gain 
adjusted to produce a discrete signal, x(n), with an rms value D. x(n) is 
then filtered by a narrowband bandpass filter (104) characterized by 
transfer function h(n) whose output, y(n) has rms value E. The BPF (104) 
removes speech and noise so that y(n) is a fairly close representation of 
SAT. 
y(n) is input to an autocorrelation function calculator (105) that 
generates outputs r.sub.x (0), r.sub.x (1), and r.sub.x (2). These output 
values are the energy of each sample of the sampled signal, x(n). r.sub.x 
(0) for an input sine wave of rms value E has value E.sup.2. 
The radiotelephone's transmitter also has a variance from radiotelephone to 
radiotelephone for effective transmit deviation. To produce a known 
deviation, the modulation level must be adjusted to compensate for this 
variance. This is done by adjusting the value of the constant L for each 
individual radiotelephone. 
The process of the present invention performs SAT modulation gain 
adjustment by using the E.sup.2 value to derive a gain adjustment for the 
output signal prior to its passage to the transmitter. This adjustment is 
performed by the process of the present invention by scaling the signal, 
y(n) to produce a signal of amplitude G. An input sinusoidal signal of rms 
value G is required at the input of the transmitter to produce a 2.0 kHz 
deviated signal at the transmitter output. 
G is determined by the following equation: 
##EQU1## 
where the values for 
##EQU2## 
and L are chosen so that for an autocorrelation output of E.sup.2, a 
signal of rms value G is generated if y(n) is at the correct frequency and 
within an amplitude window, as determined by the SAT detector. 
The output of the above operation is input to a low pass filter (LPF) 
(106). The LPF (106) reduces the variance of the gain adjustment signal to 
keep the transmitted signal from experiencing large amplitude changes. 
The output of the LPF (106) is one input to a switching operation (101). If 
SAT is detected by the SAT detector (110), the output of the LPF (106) is 
used in subsequent operations. If SAT is not detected, the gain of the 
transmitted signal is set to zero. 
The SAT detector (110) of the present invention determines the presence of 
the SAT signal by the amplitude and frequency characteristics of the 
received signal. To accomplish this, the process illustrated in FIG. 2 is 
used. 
This process begins by the autocorrelation values from the autocorrelation 
calculator being input to the SAT detector. Using the Levinson-Durbin 
recursive process, direct form coefficients are generated from the 
autocorrelation values. The pole locations of the autocorrelation values 
are then generated by inputting the coefficients into a quadratic 
equation. The frequency of the received signal is then determined by these 
pole locations. The amplitude characteristic is derived via r.sub.x (0) 
and and compared with a possibly variable threshold. If E.sup.2, as 
measured by r.sub.x (0) is above the threshold, the received SAT is 
determined to have sufficient amplitude to be detected, if the frequency, 
as determined by pole locations, is acceptable. 
The SAT frequency detection process can be represented mathematically as 
follows. The z--transform of a sine wave is: 
##EQU3## 
where: 
##EQU4## 
and T=sampling rate. 
This can also be written as: 
##EQU5## 
The equation for a second order linear time-invariant discrete-time system 
is: 
##EQU6## 
Equating the coefficients of the sine function with that of H(z) gives: 
##EQU7## 
Since the frequency of the received signal is wanted, the frequency of a 
sinusoidal signal can be explicitly represented by the denominator of 
H(z). The numerator of H(z) gives only gain information. Using the 
quadratic equation to find the pole locations in the z-domain gives: 
##EQU8## 
The frequency of the received sinusoidal signal is: 
##EQU9## 
The denominator coefficients of the second-order linear time-invariant 
discrete-time system can be easily determined by using the Levinson-Durbin 
recursive process. The Levinson-Durbin recursive process for determining 
the direct form coefficients is as follows, where r.sub.xx (m) is the 
unbiased estimate of the true autocorrelation. 
##EQU10## 
where: X(n)=discrete data sample, 
N=analysis length. 
Now r.sub.xx (0), r.sub.xx (1), and r.sub.xx (2), the autocorrelation 
values from the autocorrelation calculator, are used to calculate the 
direct form coefficients. 
a.sub.2 (0)=1 
a.sub.2 (1)=a.sub.1 (1)+.GAMMA..sub.2 .multidot.a.sub.1 (1) 
a.sub.2 (2)=.GAMMA..sub.2 
##EQU11## 
If the frequency of the signal is determined to be either 5970 Hz.+-.10 Hz, 
6000 Hz.+-.10 Hz, or 6030 Hz.+-.10 Hz, SAT is present in the received 
signal. If SAT is present, the SAT detector (110) enables the switching 
function (101) to allow the gain adjusted signal through. If SAT is not 
present, the SAT detector (110) enables the switching function (101) to 
output a mute control (120) signal.