Apparatus and method for discriminating and suppressing noise within an incoming signal

An apparatus and method for discriminating and suppressing noise within an incoming signal which provide a first signal processing unit for processing the incoming signal to generate a first iteration signal representing average difference signal level of the incoming signal; a second signal processing unit for processing the first iteration signal to generate a second iteration signal representing specified aspects of the first iteration signal; a prediction unit for generating a predicted value for the second iteration signal from earlier samples of the second iteration signal; a logic unit for determining a threshold difference between the second iteration signal and the predicted value, the logic unit generating a logic output having a first value when the threshold difference exceeds a predetermined threshold value and having a second value when the threshold difference does not exceed the predetermined threshold value; and a muting unit for muting signals which is operatively connected to receive the incoming signal and the logic output, the muting unit responds to the logic output to mute the incoming signal when the logic output is at one value and to not mute the incoming signal when the logic output is at the other value.

CROSS REFERENCE TO RELATED APPLICATIONS 
The following applications contain subject matter similar to the subject 
matter of this application: 
U.S. application Ser. No. 07/887,470, filed May 22, 1992, entitled 
"Apparatus and Method for Discriminating and Suppressing Noise within an 
Incoming Signal". 
U.S. Pat. No. 5,299,233, issued Mar. 29, 1994, entitled "Apparatus for 
Attenuating a Received Signal in Response to Presence of Noise". 
BACKGROUND OF THE INVENTION 
In certain types of communications systems, such as systems employed with 
wireless telephone systems, information conveyed via analog signalling is 
received by a receiver in a series of encoded representations of 
information, generally in the form of a series of "1"'s and "0"'s 
established via frequency shift keying (FSK) at high frequency. Such 
transmissions may be subject to reflection such as from buildings and 
other objects in the transmission path, so there are sometimes radio 
frequency (RF) signals arriving at a receiver which are time-delayed with 
respect to other received signals in a manner which may interrupt or 
distort reception. As a result, the demodulator (or decoder) of the 
receiver may erroneously convert received signals, thereby becoming 
unstable and producing interference. Such interference is generally 
perceived by a user of such a system in the form of popping or clicking 
sounds or other distracting noises, 
It is common in the communications industry to compensate for interference 
or other noise by providing a feedback circuit with a delay whereby one 
can estimate the noise component of the received signal, generate a 
duplicate approximation of the noise component, and subtract that 
approximate noise component from the original signal to eliminate the 
noise received in the incoming signal. However, an industry standard 
published for wireless telephones and similar systems requires that no 
delays be introduced in such systems; i.e., the system must be what is 
commonly known as a real-time system. Consequently, a solution for 
eliminating noise in such a system must likewise be a real-time system. 
The present invention provides a real-time noise discriminating and 
suppressing system designed to quickly discriminate and suppress noise in 
an incoming signal and which will effect muting in a manner which does not 
prevent understanding received speech when the invention mistakenly 
triggers on true speech. 
SUMMARY OF THE INVENTION 
The present invention includes an apparatus for discriminating and 
suppressing noise within an incoming signal which comprises a first signal 
processing unit for processing the incoming signal to generate a first 
iteration signal which represents average signal level, or signal 
strength, of the incoming signal; a second signal processing unit for 
processing the first iteration signal to generate a second iteration 
signal which is representative of specified aspects of the first iteration 
signal; a prediction unit for generating a predicted value for the second 
iteration signal from a plurality of earlier samples of the second 
iteration signal; a logic unit for determining a difference between the 
second iteration signal and the predicted value, the logic unit generating 
a logic output having a first value when the difference exceeds a 
predetermined threshold value and having a second value when the 
difference does not exceed the predetermined threshold value; and a muting 
unit for muting signals which is operatively connected to receive the 
incoming signal and the logic output, the muting unit responds to the 
logic output to mute the incoming signal when the logic output is at one 
of the first and second values and responds to the logic output to not 
mute the incoming signal when the logic output is at the other of the 
first value and the second value. 
A further aspect of the present invention includes a method for 
discriminating and suppressing noise within an incoming signal which 
comprises the steps of (1) generating a first iteration signal which is 
representative of average signal level, or signal strength, of the 
incoming signal; (2) generating a second iteration signal which is 
representative of the first iteration signal normalized with respect to a 
maximum signal level expected of the incoming signal; (3) generating a 
predicted value for the second iteration signal from a plurality of 
earlier samples of the second iteration signal; (4) determining a 
difference between the second iteration signal and the predicted value; 
(5) generating a control signal which has a first value when the 
difference exceeds a predetermined threshold value and has a second value 
when the difference does not exceed the predetermined threshold value; and 
(6) providing a muting unit for muting signals which is operatively 
connected to receive the incoming signal and the control signal and 
responds to the control signal to mute the incoming signal when the 
control signal is at one of the first and second values and to not mute 
the incoming signal when the control signal is at the other of the first 
and second values. 
It is, therefore, an object of the present invention to provide a method 
and apparatus for discriminating and suppressing noise within an incoming 
signal which efficiently and accurately identifies and discriminates noise 
within an incoming signal and mutes that noise. 
Further objects and features of the present invention will be apparent from 
the following specification and claims when considered in connection with 
the accompanying drawings illustrating the preferred embodiment of the 
invention.

DETAILED DESCRIPTION OF THE INVENTION 
FIG. 1 is a schematic block diagram of an alternate embodiment of the 
apparatus of the present invention. In FIG. 1, an apparatus 10 for 
discriminating and suppressing noise within an incoming signal is 
illustrated as including an signal level averaging unit 12, a log 
conversion and normalizing unit 14, a prediction unit 16, a comparator 
unit 18, and a muting unit 20. 
Apparatus 10 may be employed in a wireless phone system (i.e., a phone 
system having no cord connecting the phone receiver and a base unit, the 
base unit being configured to receive telephone signals by wire or other 
means). In such wireless telephone systems, the input speech received at 
an input 22 of signal level averaging unit 12 is already band limited to 
four KHz prior to reception by the receiving system. This input speech can 
be received from other sources in other systems, such as recording 
machines, and may be in linear format or in compressed form having been 
compressed according to any of several compression algorithms. Speech 
which is in compressed form [e.g., pulse code modulated (PCM), adaptive 
differential pulse code modulation (ADPCM), differential pulse code 
modulation (DCPM), or the like] passes through a normal decoder for the 
appropriate respective coding system and is then introduced into the audio 
receiver as linear data. Noise detection preferably occurs after the 
received signal is decoded. No delay need be introduced into the receiving 
system if such delay is critical to performance (or if industry standards 
preclude any delay). However, if a small amount of delay is used, the 
early part of a noisy passage of signal can be removed which improves 
performance. 
An incoming signal s is received from the output of a signal decoder (not 
shown in FIG. 1) by signal level averaging unit 12 at input 22. Signal 
level averaging unit 12 calculates an average signal level .delta..sub.t, 
according to the relationship: 
##EQU1## 
where .delta..sub.t =average signal level, 
.vertline.s.sub.i .vertline.=absolute value of the signal level of the 
i.sup.th sample of the incoming signal s, and 
n=number of samples. 
Signal level averaging unit 12 averages the absolute value of incoming 
signal s over n samples at a sampling rate of 8 KHz. 
During normal speech, the output of signal level averaging unit 12 varies 
slowly so that its output can be predicted accurately. However, when 
incoming signal s is noisy, large errors occur in the prediction of 
average signal level .delta.hd t. Such presence of noise in incoming 
signal s can thus be detected whenever the inverse of prediction error is 
less than a predetermined threshold level (or, conversely, whenever 
prediction error is greater than a predetermined threshold level). The 
choice of threshold level can affect the performance of apparatus 10 
because, if the threshold is set too high, apparatus 10 triggers on true 
speech, and if the threshold is set too low, noise can be missed. 
Signal level averaging unit 12 provides its output 6t to log conversion and 
normalizing unit 14 via a line 24. Average signal level .delta..sub.t is 
provided to log conversion and normalizing unit 14 via a line 24. Incoming 
signal s is preferably received as a series of "1"'s and "0"'s in the form 
of a frequency shift keyed (FSK) signal, converted to linear data by a 
decoder before input to signal level averaging unit 12. 
Log conversion and normalizing unit 14 performs a conversion of average 
signal level .delta..sub.t, according to the known relationship: 
##EQU2## 
where S.sub.norm =average signal level normalized with respect to maximum 
signal level, 
.delta..sub.t =average signal level, and 
.delta..sub.max =maximum expected signal level. 
Thus, log conversion and normalizing unit 14 provides at an output 26 the 
quantity S.sub.norm (expressed in decibels (db)) representing average 
signal level .delta..sub.t normalized with respect to maximum signal level 
.delta..sub.max expected to be received by apparatus 10. Preferably, 
.delta..sub.max 32 8192. 
The implementations of Equations (1) and (2) are preferably simplified for 
use in a binary arithmetic system and it is desirable that the division 
functions be removed for convenience, efficiency, and speed in 
calculation. 
Each incoming signal sample s.sub.i is a 16-bit 2's complement number 
arranged in a format shown below where the arithmetic mask is 19-bit: 
__________________________________________________________________________ 
Bit No. 
18 
17 
16 
15 
14 
13 
12 
11 
10 
9 
8 7 6 5 4 3 2 1 0 
S 15 
14 
13 
12 
11 
10 
9 8 7 6 5 4 3 2 1 
__________________________________________________________________________ 
where "S" is the sign bit and there are 15 magnitude bits, "bits 15-1". 
Equation (1) requires that the "modulus" , or absolute value, of the signal 
level, or signal strength, of each incoming signal sample s.sub.i be 
generated and added to the previous (n-1) samples of .vertline.s.sub.i 
.vertline.. The value of n is preferably chosen to be 8 since it 
represents 2.sup.3 and is therefore easily handled by a binary number 
system. Eight samples can be accumulated in an arithmetic logic unit (ALU) 
in the number format shown above without overflow, and an averaging filter 
which accommodates n=8 has demonstrated good performance in fulfilling the 
requirements of signal level averaging unit 12. 
Thus, when average signal level .delta..sub.t is finally accumulated, it 
resides in an ALU in the following accumulated format: 
##STR1## 
Equation (2) may be simplified to express S.sub.norm in a base 2 logarithm 
format for more straightforward treatment using a binary system in the 
form of Equation (3) as shown below: 
##EQU3## 
.delta..sub.max is preferably chosen as 8192=2.sup.13, since in this 
exemplary system 8192 is the maximum signal level expected in a normal 
decoder output. Any signal in excess of 8192 is, thus, treated as an 
overloaded signal (i.e., saturated speech input or noise). 
Equation (3) simplifies the operation of apparatus 10, especially since it 
avoids division. K is a scaling factor, which can be ignored, and K.sub.1 
is a constant which may be prestored and, therefore, does not require 
calculation. The most significant 16 bits of the ALU in the accumulated 
format described above are applied to log.sub.2 hardware (which is 
available as a known circuit block); the output of the log.sub.2 hardware 
is an 11-bit plus implied positive sign (12-bit) value stored in a 12-bit 
format as shown below: 
##STR2## 
This log.sub.2 value is a signed, 4 -bit exponent, 7 -bit mantissa 
parameter where the 4 -bit exponent represents the position of the most 
significant "1" in the 16 -bit value .delta..sub.t. 
The prestored constant K.sub.i may then be subtracted from the output 
.delta..sub.t of signal level averaging unit 12 by log conversion and 
normalizing unit 14 to generate normalized average signal level 
S.sub.norm. 
Normalized average signal level S.sub.norm is conveyed via a line 28 to a 
first input 29 of comparator unit 18, and is conveyed via a line 30 to an 
input 32 of prediction unit 16. 
The predicted value of the normalized average signal level S.sub.norm is 
determined according to the relationship: 
##EQU4## 
where m=4 and .UPSILON..sub.i are predefined constants. The four past 
scaled values of S.sub.norm are represented by .UPSILON..sub.i 
S.sub.norm.sup.t-i and are accumulated in prediction unit 16. Equation (4) 
may be executed employing an 8 .times.16-bit parallel multiplier where the 
most significant 16 -bits of the S.sub.norm parameter (in the 12 -bit 
format described above) are applied to the 16 -bit input of the multiplier 
and the .UPSILON..sub.i parameters are (prestored as 8-bit signed 
coefficients) applied to the 8-bit input of the multiplier. The output of 
the multiplier is preferably in the 12 -bit format described above (and in 
log.sub.2 format). 
Prediction unit 16 conveys the value of S.sub.norm via a line 34 to a 
second input 36 of comparator unit 18. The apparatus by which prediction 
unit 16 calculates the value S.sub.norm will be discussed in greater 
detail hereinafter in connection with FIG. 3. 
Comparator unit 18 calculates a prediction error parameter E.sub.t, 
preferably in decibels (db), according to the relationship: 
##EQU5## 
This may be simplified to: 
EQU E.sub.t =K.sub.2 (log.sub.2 S.sub.norm -log.sub.2 [S.sub.norm -S.sub.norm 
]) [Eq. 6] 
Execution of Equation (6) requires only two subtractions and two log.sub.2 
functions, thus simplifying execution of Equation (6) by eliminating any 
division operations. K.sub.2 =20log.sub.10 2, and is a constant which can 
be ignored in the preferred exemplary implementation. 
E.sub.t is compared with a predetermined threshold, for example 8 db, so 
that if E.sub.t exceeds the predetermined threshold, it is assumed that 
incoming signal s contains speech information, and if E.sub.t is less than 
the predetermined threshold, it is assumed that incoming signal s contains 
noise. Thus, muting unit 20 receives a logic output representing a 
comparison of E.sub.t with the predetermined threshold via a line 38 from 
comparator unit 18. Muting unit 20 also receives incoming signal s via a 
line 40. Muting unit 20 responds to the logic output signal received on 
line 38 to effect muting on incoming signal s received via line 40 when 
the logic output signal received via line 38 indicates that incoming 
signal s contains noise; i.e., when E.sub.t is less than the predetermined 
threshold. Similarly, muting unit 20 effects no muting of incoming signal 
s when the logic output signal received via line 38 indicates that 
incoming signal s contains speech information; i.e., when E.sub.t exceeds 
the predetermined threshold. Muting unit 20 provides an output signal on a 
line 42, which output signal is either a muted or non-muted signal, 
depending upon the value of the logic output signal received by muting 
unit 20 via line 38 from comparator unit 18. 
Thus, apparatus 10 performs a noise discriminating function and a noise 
suppressing function. When apparatus 10 determines that incoming signal s 
contains noise, it effects muting of incoming signal s until it is later 
determined that incoming signal s contains speech information, at which 
time muting of incoming signal s may be discontinued. Alternatively, 
muting may be effected for a predetermined muting period (e.g., 256 clock 
cycles) on each occasion of noise detection, and a recheck for presence of 
noise may be conducted after each such muting period expires to determine 
anew whether to impose muting for a succeeding muting period. 
Apparatus 10 is based upon a signal level prediction and upon an assumption 
that during normal speech the average signal level varies relatively 
slowly but channel noise often causes the decoder (from which incoming 
signal s is received by apparatus 10) to become unstable resulting in a 
large increase in decoded signal level in a much shorter time than is 
experienced when incoming signal s contains speech information. It is this 
rapid and large increase in level which is perceived as unpleasant clicks 
or pops by a user. 
During normal speech, the relatively slowly-varying signal level of 
incoming signal s can be predicted relatively accurately. However, when 
the decoder becomes unstable (as when noise is present) this signal is not 
predictable and results in a large error in the predictor output. 
Apparatus 10 detects the onset of a noise burst whenever the prediction 
error parameter E.sub.t is less than a predetermined threshold. The choice 
of the threshold directly affects performance and responsiveness of the 
system, and is generally determined empirically. 
Instead of relying upon a predetermined muting period to effect noise 
reduction, it may be advantageous to detect the termination of a noise 
burst, a burst end, to determine when to cease muting. To detect a burst 
end, apparatus 10 may track the slope of a quantizer step size which is 
calculated simply as the average difference of signal level (or another 
parameter related to signal level) over some interval. This slope is 
compared to a threshold and, if the slope is more negative than the 
threshold, then the end of an error burst is determined to have occurred. 
Once such a burst end is detected, muting may be terminated or may be 
continued for a short interval in order to ensure full recovery of the 
decoder before discontinuing muting. The optimum interval for such 
extended muting is preferably determined empirically. 
FIG. 2 is a flow diagram illustrating an alternate embodiment of the method 
of the present invention. In FIG. 2, the method starts at a block 60 with 
reception of incoming signal s (see FIG. 1), and average signal level 
.delta..sub.t is calculated (Block 62) as a function of incoming signal 
level s, according to the relationship: 
##EQU6## 
where .delta..sub.t =average signal level, 
.vertline.s.sub.i .vertline.=absolute value of the signal level of the 
i.sup.th sample of the incoming signal s, and 
n=number of samples. 
Then there is calculated (Block 64) a normalized average signal level 
S.sub.norm, according to the relationship: 
##EQU7## 
where S.sub.norm =average signal level normalized with respect to maximum 
signal level, 
.delta..sub.t =average signal level, 
.delta..sub.max =maximum expected signal level, 
K=10log.sub.10 2, and 
K.sub.1 =log.sub.2 .delta..sub.max. 
A predicted value of the normalized average signal level S.sub.norm is 
calculated (Block 66), according to the following relationship: 
##EQU8## 
where S.sub.norm =predicted average signal level normalized to maximum 
expected signal level of said incoming signal, 
.UPSILON..sub.i =a scaling factor, 
S.sub.norm.sup.t-i =i.sup.th past sample of normalized average signal 
level, and 
m=number of samples. 
In the preferred embodiment, it has been found sufficient for the needs of 
a wireless telephone application to employ a value of 4 for the number of 
samples m, and to employ predetermined constants for the values of scaling 
factors .UPSILON..sub.1. 
A prediction error parameter E.sub.t is calculated, preferably according to 
the expression: 
##EQU9## 
where K.sub.2 =20log.sub.10 2. 
E.sub.t is compared with a predetermined threshold value (Block 70 ) to 
generate a logic output signal reflecting that: comparison; the logic 
output value has a first value when E.sub.t exceeds the predetermined 
threshold value, thereby indicating the presence of speech in incoming 
signal s; and the logic output signal has a second value when E.sub.t is 
less than the predetermined threshold, thereby indicating the presence of 
noise in incoming signal s (Block 72). Thus, incoming signal s is muted by 
a muting device when the logic output signal indicates the presence of 
noise in incoming signal s (Block 74). When the logic output signal 
indicates the presence of speech in incoming signal s, the muting device 
effects no muting of incoming signal s (Block 76). 
FIG. 3 is a schematic block diagram of a representative embodiment of the 
prediction unit of the present invention. In FIG. 3, a prediction unit 16 
is illustrated as a multi-stage digital filter comprising a plurality of 
delay units 50.sub.1, 50.sub.2, 50.sub.3, 50.sub.4, . . . 50.sub.n. Each 
delay unit 50.sub.i delays the received signal S.sub.norm (normalized 
average signal level) by an additional time period so that an output line 
52.sub.1 conveys the received signal delayed one time period, 
S.sub.norm.sup.t-4 ; output line 52.sub.2 conveys the received signal 
delayed by two time periods, s.sub.norm.sup.t-2 ; output line 52.sub.3 
conveys the received signal delayed three time units, S.sub.norm.sup.t-3 ; 
output line 52.sub.4 conveys the received signal delayed by four time 
units, S.sub.norm.sup.t-4 ; and output line 52.sub.n conveys the received 
signal delayed by n time units, S.sub.norm.sup.t-n. 
Each of the respective output signals S.sub.norm.sup.t-i conveyed by output 
lines 52.sub.i are respectively multiplied by a scaling factor 
.UPSILON..sub.i so that output signal S.sub.norm.sup.t-1 is multiplied by 
scaling factor .UPSILON..sub.1, output signal S.sub.norm.sup.t-1 is 
multiplied by scaling factor .UPSILON..sub.2, output signal 
S.sub.norm.sup.t-3 is multiplied by scaling factor .UPSILON..sub.3, output 
signal S.sub.norm.sup.t-4 is multiplied by scaling factor .UPSILON..sub.4, 
and output signal S.sub.norm.sup.t-n is multiplied by scaling factor 
.UPSILON..sub.n. The resultant scaled output signals .UPSILON..sub.i 
S.sub.norm.sup.t-i are summed in a summer 54 which provides at an output 
56 a predicted value S.sub.norm for normalized average signal level 
S.sub.norm based upon a plurality of earlier samples S.sub.norm.sup.t-i of 
S.sub.norm. 
For purposes of clarity and ease in understanding the present invention, 
like elements will be identified by like reference numerals in the various 
drawings. 
FIG. 4 is a schematic block diagram of the preferred embodiment of the 
present invention. In FIG. 4, an incoming signal s is received by an 
apparatus 100 from the output of a signal decoder (not shown in FIG. 4) by 
a signal difference sampling unit 11. 
Signal difference sampling unit 11 provides an output which represents the 
difference between a current sample of incoming signal s with a previous 
sample of incoming signal s. When incoming signal s is sampled at regular 
intervals (i.e., at a constant sampling rate), then the difference output 
generated by difference sampling unit 11 is directly related to the rate 
of change, or slope, of the signal level of incoming signal s at each 
sampling. This rate of change of signal level is particularly useful in 
discriminating noise from communications signals, especially speech 
signals, since the slope of noise signals is significantly different from 
the slope of speech signals. Noise signals are very easily and, more 
important, very quickly distinguishable from speech signals by their 
slope. The difference signal output may, for example, be determined by an 
arrangement such as the device illustrated in FIG. 5. 
FIG. 5 is a schematic block diagram of details of a representative device 
suitable for use as the signal difference sampling unit of the preferred 
embodiment of the present invention illustrated in FIG. 4. In FIG. 5, an 
incoming signal sample s.sub.i is received at a juncture 101. Signal 
sample s.sub.i is applied to a delay unit 102 and to the positive node 104 
of a summer 106. Delay unit 102 generates a delayed (preferably a 1-sample 
delay) signal sample s.sub.i-1 , and delayed signal sample s.sub.i-1 is 
applied to a negative node 108 of summer 106. Summer 106 generates at an 
output 110 a difference signal s.sub.i -S.sub.i-l. Difference signal 
s.sub.i -S.sub.i-1 is received by signal difference level averaging unit 
13 at input 22 (FIG. 4). Signal difference level averaging unit 13 
calculates an average difference signal level .DELTA., preferably 
according to the relationship: 
##EQU10## 
where .DELTA..sub.t =average difference signal level, 
.vertline.s.sub.i -s.sub.i-1 .vertline.=absolute value of the difference 
between the signal level of the i.sup.th sample of the incoming signal s 
and the (i-1).sup.th sample of the incoming signals, and 
n=number of samples. 
Signal difference level averaging unit 13 preferably averages the absolute 
value of difference signal level (s.sub.i -s.sub.i-1) over n samples at a 
sampling rate of 8 KHz. 
During normal speech, the output of signal difference level averaging unit 
13 varies slowly so that its output can be predicted accurately. However, 
when incoming signal s is noisy, large errors occur in the prediction of 
average difference signal level .DELTA.. Such presence of noise in 
incoming signal s can thus be detected whenever the inverse of prediction 
error is less than a predetermined threshold level (or, conversely, 
whenever prediction error is greater than a predetermined threshold 
level). 
Returning to FIG. 4, signal difference level averaging unit 13 provides its 
output .DELTA..sub.t to log conversion and normalizing unit 14 via a line 
24. Average signal level .DELTA..sub.t is provided to log conversion and 
normalizing unit 14 via a line 24. Incoming signal s is preferably 
received as a series of "1"'s and "0"'s in the form of a frequency shift 
keyed (FSK) signal, converted to linear data by a decoder before input to 
signal difference sampling unit 11. 
Log conversion and normalizing unit 14 performs a conversion of average 
difference signal level .DELTA..sub.t, preferably according to the known 
relationship: 
##EQU11## 
where .DELTA.S.sub.norm =average difference signal level normalized 
difference with respect to maximum signal level, 
.DELTA..sub.t =average difference signal level, and 
.DELTA..sub.max =maximum expected difference signal level. 
Thus, log conversion and normalizing unit 14 provides at an output 26 the 
quantity .DELTA.S.sub.norm (expressed in decibels (db)) representing 
average difference signal level .DELTA..sub.max normalized with respect to 
maximum difference signal level .DELTA..sub. expected to be received by 
apparatus 100. 
The implementations of Equations (11) and (12) are preferably simplified 
for use in a binary arithmetic system and it is desirable that the 
division functions be removed for convenience, efficiency, and speed in 
calculation. Equation (12) may be simplified to express .DELTA.S.sub.norm 
in a base 2 logarithm format for more straightforward treatment using a 
binary system in the form of Equation (13) as shown below: 
##EQU12## 
Equation (13) simplifies the operation of apparatus 100, especially since 
it avoids division. K is a scaling factor, which can be ignored, and 
K.sub.1 is a constant which may De prestored and, therefore, does not 
require calculation. 
The prestored constant K.sub.1 may be subtracted from the output 
.DELTA..sub.t of signal difference level averaging unit 13 binary log 
conversion and normalizing unit 14 to generate normalized average 
difference signal level .DELTA.S.sub.norm. 
Normalized average difference signal level .DELTA.S.sub.norm is conveyed 
via a line 28 to a first input 29 of comparator unit 18, and is conveyed 
via a line 30 to an input 32 of prediction unit 16. 
The predicted value of the normalized average difference signal level 
.DELTA.S.sub.norm is preferably determined according to the relationship: 
##EQU13## 
where m=4 and .UPSILON..sub.i are predefined constants. The four past 
scaled values of .DELTA.S.sub.norm are represented by .UPSILON..sub.i 
.DELTA..sub.norm.sup.t-i and are accumulated in prediction unit 16. 
Equation (14) may be executed employing an 8.times.16 -bit parallel 
multiplier where the most significant 16 -bits of the .DELTA.S.sub.norm 
parameter (in the 12-bit format described above) are applied to the 16 
-bit input of the multiplier and the .UPSILON..sub.i parameters are 
(prestored as 8-bit signed coefficients) applied to the 8-bit input of the 
multiplier. The output of the multiplier is preferably in the 12 -bit 
format described above (and in log.sub.2 format). 
Prediction unit 16 conveys the value of .DELTA.S.sub.norm via a line 34 to 
a second input 36 of comparator unit 18. The apparatus by which prediction 
unit 16 calculates the value .DELTA.S.sub.norm was discussed in connection 
with FIG. 3. 
Comparator unit 18 calculates a prediction error parameter E.sub.t, 
preferably in decibels (db), according to the relationship: 
##EQU14## 
This may be simplified to: 
EQU E.sub.t =K.sub.2 (log .sub.2 .DELTA.S.sub.norm -log .sub.2 
[.DELTA.S.sub.norm ]) [Eq. 16] 
Execution of Equation (16) requires only two subtractions and two log.sub.2 
functions, thus simplifying execution of Equation (16) by eliminating any 
division operations. K.sub.2 =20log.sub.10 2, and is a constant which can 
be ignored in the preferred exemplary implementation. 
The remainder of the preferred embodiment of the present invention is 
constructed and operates in substantially the same manner as has been 
previously described in connection with FIGS. 1-3. 
FIG. 6 is a flow diagram illustrating the preferred embodiment of the 
method of the present invention. In FIG. 6, the method starts at a block 
60 with reception of incoming signal s (see FIG. 5), and average 
difference signal level .DELTA..sub.t is calculated (Block 62) as a 
function of incoming signal level s, according to the relationship: 
##EQU15## 
where .DELTA..sub.t =average difference signal level, 
.vertline.s.sub.i -=absolute value of the difference between the signal 
level of the i.sup.th sample of the incoming signal s and the (i-1).sup.th 
sample of the incoming signals, and 
n=number of samples. 
Then there is calculated (Block 64) a normalized average difference signal 
level .DELTA.S.sub.norm, preferably according to the relationship: 
##EQU16## 
where .DELTA.S.sub.norm =average difference signal level normalized with 
respect to maximum signal level, 
.DELTA..sub.t average difference signal level, 
.DELTA..sub.max =maximum expected difference signal level, 
K=10log.sub.10 2, and 
K.sub.1=log.sub.2 .delta..sub.max. 
A predicted value of the normalized average difference signal level 
.DELTA.S.sub.norm is calculated (Block 66), preferably according to the 
following relationship: 
##EQU17## 
where .DELTA.S.sub.norm =predicted average difference signal level 
normalized to maximum expected difference signal level, 
.UPSILON..sub.i =a scaling factor, 
.DELTA.S.sub.norm.sup.t-i =i.sup.th past sample of normalized average 
difference signal level, and 
m=number of samples. 
In the preferred embodiment, it has been found sufficient for the needs of 
a wireless telephone application to employ a value of 4 for the number of 
samples m, and to employ predetermined constants for the values of scaling 
factors .UPSILON..sub.i. 
A prediction error parameter E.sub.t is calculated, preferably according to 
the expression: 
##EQU18## 
where K.sub.2 =20log.sub.10 2. 
E.sub.t is compared with a predetermined threshold value (Block 70) to 
generate a logic output signal reflecting that comparison; the logic 
output value has a first value when E.sub.t exceeds the predetermined 
threshold value, thereby indicating the presence of speech in incoming 
signal s; and the logic output signal has a second value when E.sub.t is 
less than the predetermined threshold, thereby indicating the presence of 
noise in incoming signal s (Block 72). Thus, incoming signal s is muted by 
a muting device when the logic output signal indicates the presence of 
noise in incoming signal s (Block 74). When the logic output signal 
indicates the presence of speech in incoming signal s, the muting device 
effects no muting of incoming signal s (Block 76). 
It is to be understood that, while the detailed drawing and specific 
examples given describe preferred embodiments of the invention, they are 
for the purpose of illustration, that the apparatus of the invention is 
not limited to the precise details and conditions disclosed and that 
various changes may be made therein without departing from the spirit of 
the invention which is defined by the following claims: