Abstract:
A microphone expander attenuates background noise in a digitized microphone signal from a wireless telephone by using a loss function that exhibits hysteresis when the signal is passing through a transition range. This allows the microphone expander to apply loss more effectively because it allows for decreases in speech levels without attenuation. An attenuation level is determined using a first loss function when an averaged signal, derived from the digitized microphone signal using an algorithm that causes the averaged signal to clearly exhibit speech components in the digitized microphone signal, increases from below a lower noise threshold. However, if the averaged signal increases to above an upper speech threshold and then decreases to below that threshold, the attenuation level is determined using a different loss function, which delays introducing loss into the microphone signal as compared to the first loss function. The thresholds are increased, the amount of hysteresis is decreased, and the minimum amount of loss is increased as the noise level increases. The noise level is derived from the averaged signal using an algorithm that isolates noise components. Thus, the exact loss applied to any given sample of the microphone signal can be tailored to account for phenomena such as the tendency of people to speak louder when there is more background noise.

Description:
BACKGROUND OF THE INVENTION 
     1. Field of the Invention 
     The present invention relates to a microphone expander and, more particularly, to a microphone expander that effectively reduces background noise. 
     2. Description of Related Art 
     Microphone-expanders have been used to electronically alter the linear sensitivity of electret microphones so that they mimic the response of carbon microphones, which are less sensitive to low-level signals. This lack of sensitivity to low level signals has the incidental effect of reducing the transmission of background noise, which is typically at a lower level than speech signals. That is, in carbon microphones gain can be applied to the microphone signal to transmit the user&#39;s speech more clearly, while still keeping the noise level in the signal at an acceptable level. 
     FIG. 3 depicts a conventional microphone expander circuit that attempts to mimic that effect in conventional telephones. The signal from a telephone microphone  1  is introduced to an expander logic circuit  2  and an amplifier  3 . The expander logic circuit  2  generates a loss that is applied to the amplifier  3  to adjust the signal from the microphone  1 . 
     The expander logic circuit  2  makes a short-term measurement of the microphone signal and generates a loss signal according to the function shown in FIG.  4 . The loss function is characterized by an upper threshold T U , above which all loss is removed. In FIG. 4 the upper threshold T U  is 80 dB SPL, at which the loss becomes zero and the speech is passed unattenuated by the amplifier  3 . There is a lower threshold T L  below which the maximum loss is applied. In FIG. 4 the lower threshold T L  is 71 dB SPL, at which the loss is 15 dB (a gain of −15 dB). In between the two thresholds, the attenuation represented by the loss varies according to the level of the microphone signal. In the example shown in FIG. 4, the loss varies linearly between the two thresholds. 
     Conventional microphone expanders such as that shown in FIG. 3 can also be used to enhance the signal at the telephone receiver by using the expanded microphone signal output by the expander logic circuit. As shown in FIG. 3, a side tone path  4  introduces the expanded microphone signal to an amplifier  5 , which attenuates the signal by 12 dB. This attenuated signal is added to the signal at the telephone receiver  6 , thus resulting in a corresponding noise reduction in the received signal. 
     However, a difficulty with conventional microphone expanders in a noisy environment is that the difference between speech and noise cannot be defined simply in terms of signal level. As a result, using the microphone signal level to distinguish between the voice signal and unwanted noise is inherently unreliable. In other words, since the conventional approach simply attenuates signals depending on their level, noise at the same level as speech will be passed on as if it were speech. 
     U.S. Pat. No. 4,847,897 discloses another type of microphone expander, which attempts to measure noise utilizing signal properties that typically distinguish it from speech. The measured noise level is used to determine the amount by which to attenuate the microphone signal in the absence of a strong speech signal. This approach uses the noise level to determine the amount of loss to insert in the microphone signal and as an indication of the presence of speech. However, it does not address the problem of how to adjust the attenuation applied to the microphone signal across the normal range of speech and noise levels. 
     U.S. Pat. Nos. 3,889,059 and 3,963,868 describe speakerphones in which noise and voice levels are detected and used to alter a microphone signal. However, they too fail to address the problem of how to adjust the attenuation applied to a microphone signal across a range of speech and noise levels, particularly in a cellular telephone environment where noise levels can be especially severe. 
     SUMMARY OF THE INVENTION 
     It is an object of the present invention to provide a microphone expander that avoids the shortcomings of the prior art. 
     It is another object of the invention to provide an improved manner to determine how much loss to apply to a microphone signal by using loss functions that exhibit hysteresis, and, in another aspect of the invention, the thresholds defining the loss functions can be changed as the noise level changes so that the final signal is as optimal as possible. 
     In accordance with a first broad aspect of the invention, a microphone expander for attenuating background noise in a microphone signal comprises a signal averaging circuit for generating an averaged signal including speech components in the microphone signal, and a loss adjusting circuit for determining an amount of attenuation to be added to the microphone signal, wherein the loss adjusting circuit determines the attenuation from a first loss function if the averaged signal increases from below a lower noise threshold and from a second loss function if the averaged signal decreases from above an upper speech threshold, the first loss function providing more attenuation than the second loss function for a given averaged signal level. 
     In accordance with another broad aspect of the invention, such a microphone expander further comprises a noise level averaging circuit for generating a noise level signal representing background noise in the microphone signal and a threshold adjusting circuit for changing the first and second loss functions if the noise level signal exceeds a predetermined value. 
     In accordance with a more specific embodiment of the invention, a microphone expander for attenuating background noise in a digitized microphone signal from a wireless telephone comprises: 
     a signal averaging circuit for generating from an input signal an averaged signal including speech components in the microphone signal by sampling the digitized microphone signal in accordance with the following algorithm:                s   t     =       s     t   -   1       +         p   t     -     s     t   -   1         d1                 if                   p   t       ≥     s     t   -   1                     s   t     =       s     t   -   1       +         p   t     -     s     t   -   1         d2                 if                   p   t       &lt;     s     t   -   1                                    
      where: 
     500/sec&lt;sampling rate&lt;1000/sec, 
     P t =the new sample, 
     s t−1 =the old signal average, 
     s t =the new signal average, and 
     d 1 &lt;d 2 ; 
     a noise level averaging circuit for generating a noise level signal representing background noise in the microphone signal by sampling the averaged signal in accordance with the following algorithm:                n   t     =       n     t   -   1       +         s   t     -     n     t   -   1         d3                 if                   s   t       ≥     n     t   -   1                     n   t     =       n     t   -   1       +         s   t     -     n     t   -   1         d4                 if                   s   t       &lt;     n     t   -   1                                    
      where: 
     500/sec&lt;sampling rate&lt;1000/sec, 
     s t =the new averaged signal sample, 
     n t−1 =the old noise average, 
     n t =the new noise average, and 
     d 3 &gt;&gt;d 4 ; 
     a threshold adjusting circuit for determining parameters range, transition and hysteresis, range being an upper speech threshold minus a quantity determined by adding a predetermined margin to the noise level, transition being the lesser of (i) range multiplied by a factor less than one, and (ii) a default value TRANSITION determined by multiplying a predetermined maximum value for range by the factor, and hysteresis being the lesser of (i) range minus transition, and (ii) a default value HYSTERESIS determined by subtracting TRANSITION from the predetermined maximum value for range; 
     a loss adjusting circuit for determining a parameter delta that is the lesser of (i) transition, and (ii) the averaged signal minus a quantity determined by subtracting transition from an upper speech threshold, if a first loss function provides the amount of attenuation, or the averaged signal minus a quantity determined by subtracting transition plus hysteresis from the upper speech threshold, if a second loss function provides the amount of attenuation, or zero if delta is negative, wherein the first loss function provides the attenuation if the averaged signal increases from below a lower noise threshold defined as the upper speech threshold minus the quantity transition plus hysteresis, and the second loss function provides the attenuation if the averaged signal decreases from above the upper speech threshold, the first loss function providing more attenuation than the second loss function for a given averaged signal level when the averaged signal is less than the upper speech threshold and greater than the lower noise threshold; and 
     a loss calculating circuit for determining the attenuation to be applied to the microphone signal by multiplying a loss multiplier by a quantity determined by subtracting delta from TRANSITION. 
     The present invention is particularly adapted to implementation with a microprocessor. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS 
     The present invention will be better understood from the detailed description of its preferred embodiments which follows below, when taken in conjunction with the accompanying drawings, in which like numerals refer to like features throughout. This brief identification of the drawing figures will aid in understanding the detailed description that follows. 
     FIG. 1 shows a preferred embodiment of a microphone expander circuit according to the present invention. 
     FIG. 2 illustrates the application of loss to a microphone signal according to the present invention. 
     FIG. 3 is a schematic circuit diagram of a conventional microphone expander circuit. 
     FIG. 4 illustrates a typical loss function used by the conventional expander logic circuit shown in FIG.  6 . 
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
     FIG. 1 illustrates a microphone expander circuit  10  in accordance with a preferred embodiment of the present invention. An input line  20  carries a signal from a microphone M. Typically, the microphone signal will be converted by an analog-to-digital converter (not shown) into a 13-bit pulse code modulated signal generated at a sampling rate of 8 KHz. However, the invention is neither limited to use with such a signal, nor even, in its broadest aspects, to use with digital signals. The voice signal line  20  is introduced to an attenuator/amplifier  22 , which provides an amount of attenuation determined by the amount of loss input from a control circuit  100  according to the present invention. The signal output from the attenuator/amplifier  22  is input to the side tone path and is also transmitted to a remote location in the conventional manner. 
     The control circuit  100  includes a band-pass filter  102  that accepts the digitized microphone signal from the input line  20 . The filter  102  passes signals above a frequency of 500 Hz and below a frequency of 2000 Hz. It is known that when there is a significant amount of noise, the noise level is greater than the speech level below 500 Hz. It is also known that almost all speech energy is below 2000 Hz. Accordingly, the band-pass filter  102  assures that only that part of the signal likely to contain more speech than noise is processed by the remainder of the control circuit. Those skilled in the art will understand that the filter  102  in effect provides a signal that reflects an A-weighted sound level measurement. 
     The filtered signal is then introduced to a peak detector  104 , which, like the band-pass filter  102 , typically operates at the signal sampling rate (here 8 Hz). The peak detector  104  is a type of sample-and-hold circuit that samples the filtered signal and holds the absolute value thereof. In the peak detector  104 , however, the held value is replaced by a subsequent sampled value if the absolute value of that subsequent sampled value exceeds the currently held value. The value being held by the peak detector is reset to zero by a signal averaging circuit  106  as discussed below. The signal averager  106  therefore averages what corresponds to an “envelope” of the digitized microphone signal. 
     The signal averager  106  is adapted from conventional components used for generating information about voice signals in speakerphones. It samples the output of the peak detector  104  at a predetermined rate less than the digital sampling rate (here 8 KHz) and resets the peak detector to zero after it takes each sample. It develops an averaged signal according to the following equations:                s   t     =         s     t   -   1       +           p   t     -     s     t   -   1         4                   if                   p   t         ≥     s     t   -   1                 (   1   )                 s   t     =         s     t   -   1       +           p   t     -     s     t   -   1         32                   if                   p   t         &lt;     s     t   -   1                 (   2   )                                
     where: 
     the sampling rate is 667/second 
     P t =the new peak sample 
     s t−1 =the old signal average 
     s t =the new signal average. 
     Equations (1) and (2) produce a signal that contains a representation of the speech components in the signal. Those skilled in the art will appreciate this from the denominators in the equations. The denominator in equation (1) is small (here four), so that the output signal produced by the averager  106  will tend to “attack” relatively quickly, meaning that it will respond quickly to increases in the envelope signal from the peak detector  104 . In contrast, the denominator in equation (2) is large (here 32), so that the output of the signal averager  106  will tend to “decay” much more slowly, meaning that it will respond slowly to decreases in the envelope signal. Accordingly, the averaged signal contains the peaks in the filtered microphone signal, which, when speech is present, represent the speech level. 
     The sampling rate used in the signal averager  106  is important, since it will affect the extent to which the resulting signal can be deemed to represent the speech components in the microphone signal. Typically, the sampling rate is chosen to approximate the rate at which a speaker pronounces separate syllables, the so-called “syllabic rate.” For a typical speaker, a syllable lasts about 100 milliseconds (msec), and it has been found that a signal sampling rate of about 667/second (a sample period of 1.5 msec) provides sufficient response to produce an averaged signal that accurately reflects the speech level. It is believed that a sample period of 2 msec provides the slowest sampling rate consistent with the goal of capturing every syllable in the speech. A sampling rate that is too fast (above about 1000/second) provides an insufficient number of samples to represent the speech in the signal, and will not accurately measure the envelope of the speech. All components above the phantom line L in FIG. 1 operate at the signal sampling rate, while those below the phantom line operate at the chosen syllabic rate. 
     The noise level averager  108  samples the averaged signal output by the signal averager  106  at a predetermined rate to determine a noise level average according to the following equations:                n   t     =         n     t   -   1       +           s   t     -     n     t   -   1         4096                   if                   s   t         ≥     n     t   -   1                 (   3   )                 n   t     =         n     t   -   1       +           s   t     -     n     t   -   1         4                   if                   s   t         &lt;     n     t   -   1                 (   4   )                                
     where: 
     the sampling rate is 667/second 
     s t =the new averaged signal sample 
     n t−1 =the old noise average 
     n t =the new noise average. 
     Equations (3) and (4) produce a signal that represents the noise in the signal. The denominator in equation (3) is large (here 4096), so that the signal produced by the noise level averager  108  will tend to “attack” very slowly, meaning that it will respond slowly to increases in the signal from the signal averager  106 . In contrast, the denominator in equation (4) is small (here four), so that the signal from the noise detector will tend to “decay” much more quickly, meaning that it will respond quickly to decreases in the envelope signal. Accordingly, the output signal from the noise level averager  108  represents an average of the valleys in the output of the signal averager  106 . Since the averaged signal output from the signal averager  106  has a certain DC bias that represents noise in the signal, detecting the valleys in the signal in effect detects the noise level in the output of the signal averager  106 . In other words, for any given sample, the output of the signal averager  106  typically will have a non-zero component. The noise level averager  108  produces an output that represents a particular kind of “average” of the minimum values of those non-zero components. That average corresponds to the valleys in the signal and thus detects the non-speech components (or noise) in the signal. 
     As shown in FIG. 1, the sampling rate used in the noise level averager  108  is typically the same as that used in the signal averager  106 . As explained above, it is important that the sampling rate properly reflect the “syllabic rate.” Otherwise, the adjustments to the microphone signal to be made in accordance with the present invention (as explained below) will not accurately reflect the actual properties of the speaker&#39;s voice. 
     Equations (1)-(4) are based on algorithms previously used in speakerphones, as set forth in U.S. Pat. No. 5,007,046, which is incorporated herein as if set forth in full. It should be understood that the present invention is not limited to use of these particular algorithms for detecting the levels of the portions of the signal representing speech and noise, respectively. 
     In accordance with the present invention, the output of the signal averager  106 , which in accordance with the above discussion can be deemed to represent the sound energy level of the speech components in the microphone signal  20 , is introduced to a loss adjuster circuit  110 . This circuit outputs a signal delta using the function depicted in FIG.  2 . 
     In FIG. 2 the ordinate depicts the loss (represented as negative gain) in decibels to be applied to the microphone signal by the attenuator/amplifier  22 , and the abscissa represents the output from the signal averager  106  in dB SPL. The parameter delta, determined from this curve, is used to determine the loss in a manner discussed in detail below. 
     FIG. 2 illustrates conceptually an important principle of operation of the present invention. Consider first a case in which the microphone signal is in a so-called “idle” state, when no speech is present (determined as discussed below). In that case, the loss will be determined by a first loss function curve L 2 −L 1 −H 1 . In other words, the loss is a maximum (here 15 dB) as the averaged signal level from the signal averager  106  increases to a lower threshold L 1 . The loss then decreases linearly to zero as the averaged signal level increases to the higher threshold H 1 , and then remains at zero as the averaged signal level continues to increase above H 1 . 
     According to an important feature of the present invention, when the averaged signal level decreases from above H 1  to below H 1 , the microphone signal is in a so-called “speech” state, and the loss is determined by following a second loss function curve H 1 −H 2 −L 2 . In other words, the loss stays at its minimum value of zero dB until it falls below H 2 , at which point it decreases linearly as the averaged signal level decreases to L 2 , at which the loss remains at 15 dB as the averaged signal level continues to decrease. If the averaged signal level then increases above L 2 , the loss again changes according to the curve L 2 −L 1 −H 1 , as before. 
     That is, unlike known prior art microphone expanders, the amount of loss added to the microphone signal depends on the direction in which the averaged signal level enters the transition zone between L 2  and H 1 . This is accomplished by including a state machine in the loss adjuster circuit  110 . This state machine determines for each sample of the average signal whether the average signal is above H 1  or below L 2 . If so, it then determines if the next sample is below H 1  or above L 2 , respectively. If the signal has changed state from above H 1  to below H 1 , the circuit  110  follows the curve H 1 −H 2 −L 2 ; if the signal has changed state from below L 2  to above L 2 , the circuit follows curve L 2 −L 1 −H 1 . If the circuit detects no state change the loss is determined from the curve used for the preceding sample. 
     Thus, the present invention introduces the concept of hysteresis into the loss determination, since the amount of loss introduced into the microphone signal depends on whether the averaged signal (representing speech) is increasing or decreasing. The implications of using hysteresis in determining loss are significant and are discussed in detail further below. 
     In accordance with another important aspect of the invention, the output of the noise level averager  108  is used to alter the amount of loss added by the attenuator/amplifier  22 . This is accomplished in a threshold adjuster circuit  112 . 
     Beginning with the upper speech threshold H 1 , it is increased from a constant-speech threshold by a predetermined amount as the output from the noise level averager  108  increases above a given level. The constant-speech threshold is slightly less than the expected energy level in a microphone signal under “normal” conditions, that is, when noise is not present and when a speaker is talking at a comfortable level. It has been shown that a speaker will not talk any louder than this threshold if the background noise level is below 50 dBA. In the prior art loss function depicted in FIG. 4 the constant-speech threshold is chosen as 80 dB SPL. In a preferred embodiment of the present invention, it is chosen as 84 dB SPL. 
     The threshold adjuster circuit  112  begins with the chosen value for H 1  and “constructs” loss functions as in FIG.  2 . Initially, it will be appreciated that for the expander circuit  100  to operate properly the lower noise threshold L 2  must always be higher than the noise level. Otherwise, the state machine discussed above would be “trapped” on the H 1 −H 2 −L 2  curve, since the average signal would never fall below an L 2  above the noise level. Accordingly, a parameter range is defined as range=H 1 −noise level−margin. In the present embodiment, margin is set to 1 dB, the effect being to maintain the lower noise threshold L 2  at least 1 dB above the noise level represented by the output of the noise level averager  108 . 
     Next, the threshold adjusting circuit  112  uses the current values for range and H 1  to determine the remaining parameters needed to define the hysteresis curve depicted in FIG.  2  and enable calculation of the loss for a given signal sample. 
     Two parameters, transition and hysteresis, are determined using those values. The parameters transition and hysteresis in effect define the upper noise threshold L 1  and the lower speech threshold H 2 , respectively. (In the present embodiment, H 2 −L 2  =H 1 −L 1  and H 2 −H 1 =L 2 −L 1 , although the invention is not limited to such a relationship.) These parameters are first calculated as follows: 
     
       
         transition= k* range  (5) 
       
     
     
       
         hysteresis=range−transition  (6) 
       
     
     In the present embodiment, k=0.5, so hysteresis=transition. 
     The threshold adjusting circuit  112  then compares the values of transition and hysteresis to default values, TRANSITION and HYSTERESIS, respectively. The default values are determined by choosing a maximum value for range; then TRANSITION=k*range max  and HYSTERESIS=range max −TRANSITION. In the present embodiment, range max  is chosen as twice the normal 9 dB range over which loss is applied in a typical prior art loss function such as that shown in FIG.  4 . Accordingly, since k=0.5, TRANSITION=HYSTERESIS=9 dB. The circuit  112  outputs to the loss adjusting circuit  110  the upper speech threshold value H 1 , represented in FIG. 1 as a parameter threshold, along with the values for transition and hysteresis. Regarding the latter, if k*range&gt;TRANSITION, the parameter transition is set equal to TRANSITION, and if range*(1−k)&gt;HYSTERESIS, the parameter hysteresis is set equal to HYSTERESIS. The parameters threshold, transition and hysteresis completely define the curves depicted in FIG.  2 . 
     In accordance with this embodiment of the invention, the default values TRANSITION and HYSTERESIS are used until the measured noise level increases to above 50 dB SPL. In that case, the lower speech threshold is 66 dB SPL, which can be appreciated by considering that range=84−50−1=33, whereby L 2  will be 66 dB SPL (H 1 −range max ). In other words, transition and hysteresis will both be set to their default values TRANSITION and HYSTERESIS when the noise level is less than 50 dB SPL and H 1 =84 dB SPL. Accordingly, as long as the measured noise level from the noise level averager  108  is below 50 dB SPL, H 1  is 84 dB SPL and L 2  is 66 dB SPL. 
     However, when the measured noise level increases above 50 dB SPL, H 1  is increased by 2.5 dB for every 10 dB increase in the measured noise level above 50 dB SPL. This is based on the known phenomenon that a speaker talks between 3 dB to 6 dB louder for every 10 dB increase in the measured noise level above 50 dBA. The amount by which H 1  is increased is less than the observed increase in speech level to prevent the introduction of excessive attenuation as the noise level increases. In any event, it will be appreciated that L 2  will also shift the same amount up to a noise level of 70 dB SPL, since range&gt;range max  until that noise level is reached, as explained below. This has the effect of shifting the hysteresis curve in FIG. 2 upward along the abscissa. That is, until the noise level reaches 70 dB SPL, transition and hysteresis both remain at their default values of 9 dB, but loss will be added to the microphone signal at higher levels of the averaged signal than in environments in which the measured noise level is under 50 dB SPL. 
     This feature of the invention reflects the fact that people talk louder in the presence of more background noise. Therefore, it can be expected that the averaged signal from the signal averaging circuit  106  will increase. Adding loss to the microphone signal under such circumstances will attenuate the increased background noise, but not result in a loss of the speech component of the signal since it is also at a higher level under such conditions. 
     Yet another aspect of the invention relates to threshold adjustment when the measured noise level exceeds 70 dB SPL. In that case, range&lt;range max , having the effect of raising the lower noise threshold L 2  one dB for every dB increase in the measured noise level. However, the upper speech threshold continues to increase 0.25 dB for every one dB increase in the measured noise level. This has the effect of compressing the hysteresis curve in FIG.  2 . This will be better understood by considering an example of this feature of the invention. 
     For a measured noise level of 70 dB SPL, range=H 1  (89)−noise level (70)−margin (1)=18. Then, 0.5*range=(1−0.5)*range=9, so that transition=hysteresis=9. However, if the measured noise level is, say 78 dB SPL, then range=H 1  (91)−noise level (78)−margin (1)=12. Then, 0.5*range=(1−0.5)*range=6, so that transition =hysteresis=6, because this is less than the default value of 9. This means that as the measured noise level increases to these excessive levels, the maximum amount of loss is inserted at a higher signal level than would otherwise be the case. It also has the effect of increasing (from zero) the minimum amount of loss inserted into the signal, as a result of the manner in which the actual loss inserted into the microphone signal is determined. 
     To determine the exact loss to be inserted into the microphone signal, the loss adjuster circuit  110  calculates a parameter delta. For the “idle” state delta is the lesser of transition or the averaged signal−(H 1 −transition). For the “speech” state delta is the lesser of transition or the averaged signal−[(H 1 −(transition+hysteresis)]. If the value either of the averaged signal−(H 1 −transition) or of the averaged signal−[H 1 −(transition+hysteresis)] is negative, then delta=0. In effect, delta corresponds to the “location” of the averaged signal on the relevant loss function curve. 
     A loss calculation circuit  114  calculates from delta the loss inserted by the attenuator/amplifier  22 . The circuit  114  subtracts delta from TRANSITION and then multiplies the result by a loss multiplier. The loss multiplier generally corresponds to the slope of the transition portion of the relevant loss function curve in FIG. 2 in the low noise condition (below 50 dB SPL). In the present embodiment, that value would be 1.67 (15 dB maximum loss over a signal transition of 9 dB). However, one of the advantages of the manner in which the present invention determines the actual loss is that the loss multiplier can be any value that produces the most advantageous results. In another preferred embodiment, it is chosen as 2.0. 
     The operation of the present invention can be appreciated by considering some working examples of the present embodiment. When the measured noise level is less than 70 dB SPL, transition=TRANSITION. Therefore, if delta is less than TRANSITION, then loss=the quantity (TRANSITION−delta) multiplied by the loss multiplier of 1.67. This is the condition that prevails when the averaged signal is lower than the speech threshold for the relevant loss function. On the other hand, when delta=transition=TRANSITION, meaning that the averaged signal is in the speech region (above the upper speech threshold) of the relevant loss function, loss=0. If the averaged signal is below the noise threshold for the relevant loss function, the maximum loss of 15 dB is inserted. 
     When the measured noise exceeds 70 dB SPL, the present invention exhibits several phenomena. Taking the example above, in which the measured noise level was 78 dB SPL, consider first a signal level of 83 dB SPL while the state machine is in the speech state. In that case transition=hysteresis=6, and delta=averaged signal−(H 1 −[(transition+hysteresis)]=83−79=4. Then, the loss is (9−4)*1.67=8.33. If the state machine is in the idle state under these conditions, the averaged signal−(transition)=83 −84 =−1, whereby delta=0. The loss is (9−0)*1.67=15 dB, which is the maximum loss. 
     If the averaged signal is at the upper speech threshold of 91 dB when the measured noise level is 78 dB SPL, then in the speech state transition=hysteresis=delta=6, because the averaged signal−H 1  [(transition+hysteresis)]=12. The loss will be (TRANSITION−delta)*1.67=5 dB. Thus, in this case loss is inserted even when the averaged signal is at the upper speech threshold. 
     From the above discussion, it will be appreciated that the microphone expander of the present invention reduces the amount of loss introduced to the microphone signal as the noise level increases. This enables the invention to account for the fact that a speaker will talk louder in the presence of noise, so that introducing loss even in the presence of speech will not affect microphone performance. Above a certain noise level, the maximum loss is always taken out regardless of speech level. By doing so, the present invention maintains the signal level at a more reasonable level even in the presence of extremely excessive noise. 
     It will also be appreciated that when the measured noise level increases to a certain value, the microphone expander according to this embodiment of the invention will insert loss regardless of the averaged signal level. This occurs because the lower noise threshold eventually reaches a value that is the same as the upper speech as the noise level continues to increase. In that event, the maximum amount of loss is inserted because transition becomes 0 regardless of the signal level, so that delta=0. This reflects the expected reaction of the speaker to talk so loudly in the presence of high background noise that adding the maximum loss will actually reduce the microphone signal to a more normal level. 
     The loss adjusting circuit  110 , threshold adjusting circuit  112  and the loss calculating circuit  114  will typically be provided as a microprocessor programmed to perform as described above. They are depicted herein as separate circuits for the sake of describing their operation. However, it is within the scope of the invention to embody these circuits in other ways, such as separate logic circuits using TTL logic elements. 
     It will be appreciated that the expanded signal will not only be transmitted to a remote location as a telephone signal, but also will be introduced to the side tone path as well, as in the prior art shown in FIG.  3 . 
     The present invention is particular well adapted for use in a mobile telephone environment. In a quiet environment such as an office, the background noise is not only low but also stable. However, in a mobile setting, such as a car or busy street or sports stadium, noise is not only greater but extremely dynamic as well. The optimum microphone expander should be capable of changing the loss it adds to the microphone signal in a manner that accounts for these properties. The present invention, by using a dynamic loss function that uses hysteresis to account for various phenomena present in actual speech. For example, speech levels trail off at the ends of words, thus mimicking termination of speech and triggering signal attenuation in conventional microphone expanders. However, the loss function with hysteresis according to the present invention accounts for that phenomenon by using a different loss curve once it is established that speech has begun. The present invention also enables dynamic adjustment of the loss function depending on the noise level. Accordingly, the threshold at which the maximum loss is inserted can be adjusted as the noise level increases. Moreover, the manner in which the invention is implemented enables that feature to be used to account for the fact that people speak louder in the presence of background noise. 
     While preferred embodiments of the invention have been depicted and described, it will be understood that various changes and modifications can be made other than those specifically mentioned above without departing from the spirit and scope of the invention, which is defined solely by the claims that follow.