Abstract:
An apparatus for discriminating between signals produced by a desired acoustic source and an undesired acoustic source includes: a first microphone disposed at a first distance from the desired acoustic source; at least a second microphone disposed at a second distance from the desired acoustic source; a proximity estimation block configured to utilize signals form the first microphone and second microphone to produce a signal representing an estimate of the proximity of the desired acoustic source; and a variable gain block configured to adjust the gain of an output signal from at least one of the microphones by utilizing the estimate of the proximity of the desired acoustic source. A method of discriminating between signals produced by a desired acoustic source and an undesired acoustic source includes: utilizing signals from a first microphone and second microphone to produce an output signal representing an estimate of the proximity of the desired acoustic source; and adjusting the gain of the signal from at least one of the first and second microphones by utilizing the estimate of the proximity of the desired acoustic source.

Description:
FIELD OF THE INVENTION 
     The present invention relates generally to the field of communications with possible uses in voice recognition, public address, and recording. The present invention relates more particularly to a signal expander system that can distinguish sounds from acoustic sources placed at different locations relative to a transducer. 
     BACKGROUND OF THE INVENTION 
     Conventional voice communication systems typically incorporate some form of voice activated switch or signal expander for suppressing acoustic background noises. FIG. 1A illustrates a functional block diagram of a conventional bi-stable signal expander system  100  comprising a microphone  105 , an expander control stage  110 , and a variable gain block (stage)  115 . The microphone  105  is coupled to the variable gain block  115 , and the expander control stage  110  is coupled to the microphone  105  and the variable gain block  115 . The expander control stage  110  includes a detector  120  coupled between the microphone  105  and a first input  125  of a comparator  130 . The comparator  130  has a second input  135  coupled to a reference (threshold) voltage source  140  for generating a reference voltage level Vref. 
     When an acoustic source  145  becomes active, the emitted sounds from acoustic source  145  will cause changes in the air pressure. The microphone  105  detects the air pressure changes and translates the air pressure changes into corresponding voltage changes (i.e., microphone output signals  150 ) that are detected by the detector  120 . The detector  120  outputs the microphone output signal  150  as a detector output signal  155 . The comparator  130  compares the voltage level of the detector output signal  155  with the reference voltage level Vref from reference voltage source  140 . If the voltage level of the detector output signal  155  is below the reference voltage level Vref, then the comparator  130  generates an output signal  160  with a logical state that does not activate the variable gain block  115 . As a result, the variable gain block  115  does not add gain to the microphone output signal  150 . When the acoustic source  145  activates, the voltage level of the microphone output signal  150  rises. The detector  120  detects the higher-level microphone output signal  150  and will, as a result, output a higher-level detector output signal  155 . If the voltage level of detector output signal  155  rises above the reference voltage level Vref, the comparator  130  outputs a control signal  160  with a logic state that causes the variable gain block  115  to add gain to the microphone output signal  150 . The variable gain block  115  then outputs the amplified microphone output signal as an audio output signal  165 . 
     A disadvantage of the bi-stable signal expander system  100  is that a low-level sound (e.g., soft speech) from the acoustic source  145  may not be amplified if the magnitude of the low level sound does not trigger the comparator  130 . If the threshold of comparator  130  is set too low, then noise signals in the environment will easily trigger the comparator  130 , thereby activating the variable gain block  115  and amplifying the noise signals (“noise pumping”). If the threshold of comparator  130  is set too high, then softer sounds from the acoustic source  145  may not trigger the comparator  130  and not activate the variable gain block  115 , resulting in inadequate gain for the desired signal. 
     In order to reduce the consequence of low level speech failing to activate the comparator  130 , some systems set the minimum gain of the variable gain block  115  to be only 12 dB to 20 dB below the maximum (fully activated) gain. 
     A disadvantage of the above conventional bi-stable signal expander systems is that ambient or undesired noises having magnitudes above the threshold level of comparator  130  are amplified. Additionally, if the bi-stable expander system  100  will be used in a noisy environment, the threshold of the comparator  130  must be set appropriately so that the external noises do not trigger the comparator  130 , increasing the severity of noise modulation and increasing the likelihood that the system will not respond properly to voice. 
     Additionally, if the microphone  105  is oriented away from the acoustic or speech source  145 , the microphone  105  may not be able to properly detect the desired sound waves. As a result, the microphone output voltage level  150  will be low and may not trigger the comparator  130  to permit amplification of the desired detected sound. 
     The above-mentioned bi-stable signal expander systems have a fast attack and slow decay characteristic that causes the switches for controlling gain to respond quickly to a detected sound of a sufficient voltage level and to maintain the gain for a pre-defined time length (e.g., 150 ms to 200 ms) after the voltage level of the detected sound falls below the comparator  130  threshold. By maintaining the gain for the additional pre-defined time length, the quieter-sounding, trailing ends of the speech envelope are not cut off by the bi-stable signal expander system. These trailing ends are typically below the comparator threshold. However, noise is often amplified during the additional pre-defined time length when gain is maintained. 
     The above-mentioned bi-stable signal expander systems also encounter problems when a burst of background noise occurs. For example, the noise burst might be a typewriter key impact or other types of noises with impulse waveforms. The noise burst will trigger the comparator  130  in the bi-stable signal expander system, thereby adding gain to the undesired noise burst. In addition, since the gain is maintained for the above-mentioned pre-defined time length after the noise burst occurrence, subsequent undesired noises are also amplified. 
     FIG. 1B illustrates a conventional variable gain signal expander system  200  including an expander control stage  205  coupled between the microphone  105  and the variable gain block  115 . The expander control stage  205  includes a detector  210  coupled to an amplifier  215 . The microphone output signal  150  is detected by the detector  210  and amplified by the amplifier  215 . As a result, the amplifier  215  generates a control signal  220  with a magnitude that depends on the initial magnitude of the microphone output signal  150 . The amount of gain provided by the variable gain block  115  to the microphone output signal  150  depends on the magnitude of the control signal  220 . 
     The variable gain signal expander system  200  can be designed with a shorter time constant for reduced audibility of “noise pumping”. The shorter time constant reduces the amount of time that high gain is applied to the noise signal as the desired signal drops in amplitude. The effect of the combination of variable gain with reduced time constants on the desired signal is to modulate the envelope of the speech signal. This is not generally desirable but may be an acceptable compromise in noisy environments. 
     A further disadvantage of the variable gain signal expander system  200  is that both the signal from the desired acoustic source and the ambient acoustic noise are detected and used to increase the gain of the variable gain block  115 . Thus as the ambient noise levels increase, the gain of the system for this noise can also increase, resulting in less overall noise reduction. 
     Accordingly, it is desirable to provide a method and system for signal expansion with improved noise rejection capability. 
     SUMMARY OF THE INVENTION 
     The present invention provides a desirable method and system for discriminating between desired sounds from a near-field acoustic source and sounds (noise) from far-field acoustic sources. The invention advantageously prevents gain from being added to the undesired (far-field) loud noises while allowing gain to be added to low-level sounds from the desired (nearby) acoustic source. As a result, the invention can reduce the “noise pumping” problem that is encountered by conventional systems and methods. 
     The present invention can be used to enhance the noise rejection capability of headsets. Additionally, the invention may be used in other applications, such as handsets, as long as two microphones can be placed at different distances from an acoustic source. The invention may also be used to enhance the noise rejection capability of voice recognition, public address, and recording systems. 
     In one embodiment of the present invention, the signal expander system comprises an input block, a proximity estimation block (e.g., a ratio detector) coupled to the input block and a variable gain block coupled to the proximity estimation block. A speech activity detector may be optionally coupled to the proximity estimation block. 
     The input block has at least two inputs (such as two microphones) and three outputs (such a signal output and the individual outputs of each microphone). In one embodiment, the signal output of the input block is the same as the output of the first microphone. In another embodiment, each microphone output is coupled to a corresponding sensitivity adjustment block for closely matching the sensitivities of the microphones. Thus, the outputs of the microphones are derived from their corresponding sensitivity adjustment block. The signal output of the input block is derived from the difference between the outputs of the first microphone and the second microphone. This second embodiment results in a composite directional microphone from the microphone pair in the input block. This embodiment is essentially independent of the proximity issue between the microphones and the acoustic source, and is a convenient by-product of the above two microphone topology. The delay in the output signal from the second microphone alters the polar pattern from bi-directional to cardioid, supercardioid, or hypercardioid, depending on the amount of delay. 
     The proximity estimation block includes two inputs (from the microphones) and one output for generating the proximity estimation block output signal. As described below, differing degrees of signal conditioning may be applied in series with the input and output paths of the proximity estimation block. The proximity estimation (i.e., the distance between each microphone in the input block and the acoustic source) is made, for example, based on the ratio between the two inputs of the proximity estimation block. However, other methods may also be used to make the proximity estimation. 
     In one embodiment of the proximity estimation block, the first input permits the output signal of the first microphone to be received by one input of the divider, while the second input permits the output signal of the second microphone to be received by the other input of the divider. The output signal of the proximity estimation block is derived from the divider output signal. 
     Two types of signal conditioning blocks may be interposed on the input side of the divider. For example, filters may coupled to the divider input side to band-limit the signals received by the divider. Rectifying detectors may also be coupled to the divider input side to simplify the operation of the divider. 
     In one implementation, the divider is configured to take the difference of the logarithm of the divider input signals. This implementation is particularly suited to an analog implementation. The output of this implementation is the logarithm of the ratio. In some applications, the antilog of this ratio value may be taken. However, direct use of the log output frequently will result in better noise margins and permits use of simpler circuitry. 
     On the output side of the divider, a clipping and smoothing function may be inserted to reduce the effects of phase induced transients. This function may be implemented in either analog or digital form. 
     The variable gain block includes two inputs and one output and may be implemented in three basic forms, as described below. The first input of the variable gain block is the output signal from the input block. The second input of the variable gain block is from either the direct output of the proximity estimation block or from a speech activity detector having an output derived from the proximity estimation. The output of the variable gain block is the system audio output. All implementations of the variable gain block typically includes a variable gain amplifier or an attenuator, either of which is controlled directly or indirectly by the inputs to the proximity estimation block and may include components for controlling the timing of the gain changes. All implementations of the variable gain block may include a delay element on the block input to compensate for any delays introduced by the proximity estimation process. 
     One implementation of the variable gain block includes a conventional amplitude based expander where the gain of the variable gain block is determined by an input level when the proximity estimation indicates that the distance to the acoustic source is within a proscribed distance. Thus, the gain applied to the output of the first microphone is determined by both the level of the output signal itself and by the estimate of proximity (as determined by the proximity estimation block) exceeding a threshold value. The proximity estimate input to this implementation is binary regardless of the gain versus input level characteristics of the expander when activated. 
     A second implementation of the variable gain block includes a form of bi-stable expander that assumes a high or low gain state depending on the binary value of the proximity based speech activity detector. Thus, the output signal of the first microphone is amplified in response to the proximity estimate exceeding a threshold value. 
     A third implementation of the variable gain block includes a variable gain element providing a gain that is a function of the direct output of the proximity estimation block. The function may be: (i) linear, (ii) non-linear, (iii) logarithmic, or (iv) arbitrary. Thus, the gain provided to the output signal of the first microphone is a function of the direct output of the proximity estimation block. The second input of the variable gain block in this implementation is directly coupled to the output of the proximity estimation block. 
     The speech activity detector includes one input and one output. The detector input is connected to the output of the proximity estimation block and permits transmission of the proximity estimation block output signal to one input or a comparator. The other input of the comparator is a proximity reference value that establishes the limit of the distance within which the desired acoustic source is expected to be located. All acoustic sources more distant than this value will be considered as noise. The output of the comparator is the output of the speech activity detector and provides a binary indication of whether the desired acoustic source is active. 
     All embodiments of the present invention typically incorporate all four blocks (input block, proximity estimation block, variable gain block, and speech activity detector) in one of two general configurations depending on whether the variable gain block accepts the proximity estimate directly or uses the speech activity detector output. Any of the variations of the input block or the proximity estimation block can be optionally used for any embodiment disclosed herein. The speech activity detector is typically the same form in all embodiments. The variable gain block can be any of the three implementations mentioned above. 
     Additionally, the present invention provides a signal expander using digital processing to implement the functions mentioned above. In this embodiment of the invention, the output signals from the microphones in the input block are first digitized. Analog processes are replaced by digital arithmetic blocks and algorithms. This digital implementation also enables more sophisticated processes to be utilized without requiring additional hardware. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS 
     FIG. 1A is a functional block diagram of a conventional bi-stable signal expander system. 
     FIG. 1B is a functional block diagram of a conventional variable gain signal expander system. 
     FIG. 2A is a functional block diagram of a signal expander system in accordance with an embodiment of the present invention. 
     FIG. 2B is an illustration of a telephone headset that can implement the present invention. 
     FIG. 2C is an illustration of a telephone handset that can implement the present invention. 
     FIG. 3 is a functional block diagram of an analog divider in accordance with the present invention. 
     FIG. 4A illustrates the simulated source waveform when the source is close to the first microphone. 
     FIG. 4B illustrates the output waveform of each microphone when the source is close to the first microphone. 
     FIG. 4C illustrates the detected output waveform of the signal from each microphone and the linear ratio, when the source is close to the first microphone. 
     FIG. 4D illustrates waveforms of the logarithm of the signal from each microphone and the difference between the logarithms, when the source is close to the first microphone. 
     FIG. 4E illustrates waveforms of the clipped and smooth logarithmic output which in some embodiments would represent the output of the proximity estimation block, when the source is close to the first microphone. 
     FIG. 4F illustrates the output waveform of each microphone when the source is distant from the first microphone. 
     FIG. 4G illustrates the detected output waveform of the signal from each microphone and the linear ratio, when the source is distant from the first microphone. 
     FIG. 4H illustrates waveforms of the logarithm of the signal from each microphone and the difference between the logarithms, when the source is distant from the first microphone. 
     FIG. 4I illustrates waveforms of the clipped and smooth logarithmic output which in some embodiments would represent the output of the proximity estimation block. 
     FIG. 5 is a functional block diagram of a signal expander system in accordance with another embodiment of the present invention. 
     FIG. 6 is a functional block diagram of a signal expander system in accordance with another embodiment of the present invention. 
     FIG. 7 is a functional block diagram of a signal expander system in accordance with another embodiment of the present invention. 
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
     Those of ordinary skill in the art will realize that the following description of the preferred embodiments is illustrative only and not in any way limiting. Other embodiments of the invention will readily suggest themselves to those skilled in the art. 
     FIG. 2A illustrates a functional block diagram of a signal expander system  300  in accordance with a first embodiment of the present invention. The system  300  is advantageously not activated by far-field noises and allows the use of a lower threshold within the expander control section  320  to better pick up low level speech. The signal expander system  300  is capable of detecting the signals generated by an acoustic source  305  and includes a first microphone  310  coupled to a variable gain block (stage)  315  and to an expander control stage  320 . The expander control stage  320  is also coupled to the variable gain block  315 . The expander control stage  320  may be implemented by, for example, the conventional expander control stage  205  used in the conventional variable gain signal expander system  200 , as shown in FIG.  1 B. The variable gain block  315  produces the audio output signals  322 . 
     The signal expander system  300  further includes a proximity estimation block  325  that is coupled to the first microphone  310  and to a second microphone  330 . A speech activity detector (comprising a comparator  335 ) has a first input  340  coupled to the proximity estimation block  325  output and a second input  345  coupled to a reference (proximity) voltage source  350 . Thus, the second input  345  receives the reference voltage level Vref from the reference voltage source  350 . The reference source  350  is a proximity threshold, which determines whether the acoustic source that is producing a signal is a near-field or far-field source. The reference source  350  then disables the expander system  300  if a near-field source is not active. The comparator  335  may enable or disable the expander control stage  320  by use of, for example, a conventional transistor switch (not shown). The comparator  335  may also be coupled to a subsequent stage (not shown) for receiving the speech activity detector output and for processing detected speech activity. 
     In a preferred embodiment, the proximity estimation block  325  includes a divider  360  coupled to the first input  340  of the comparator  335 . The divider  360  computes the ratio of both microphone output signals  400  and  405 . Since the microphone output signals  400  and  405  are each determined by air pressure, the divider  360  is also effectively comparing the ratio of the air pressures at first microphone  310  and second microphone  330 . An analog implementation of the divider is discussed below with reference to FIG.  3 . The divider  360  has a first input  365  coupled to a first detector  370  and a second input  375  coupled to a second detector  380 . A first band-pass filter  385  is coupled between the first microphone  310  and the first detector  370 . A second band-pass filter  390  is coupled between the second microphone  330  and the second detector  380 . In an alternative embodiment, the band-pass filters  385  and  390  may be replaced with high-pass filters. In yet another alternative embodiment, the band-pass filters  385  and  390  may be omitted so that the first microphone  310  is directly coupled to the first detector  370  and the second microphone  330  is directly coupled to the second detector  380 . The functionality of the divider  360 , detectors  370  and  380 , and band-pass filters  385  and  390  are discussed below in additional detail. 
     In the various possible implementations of the present invention, the first microphone  310  is closer than the second microphone  330  in distance to the desired acoustic source  305 . For example, the present invention may be implemented in a headset  392  (FIG. 2B) or handset  394  (FIG. 2C) wherein the first microphone  310  is placed near the user&#39;s mouth and the second microphone  330  is placed near the user&#39;s ear. Alternatively, the first microphone  310  may be placed on a headset boom, while the second microphone  330  is placed on, for example, a desk near the headset. As another alternative, the first microphone  310  may be placed near an acoustic source  305  in a room, while the second microphone  330  may be placed at a slightly further distance from the acoustic source  305 . Other configurations are possible so that the first microphone  310  is closer than the second microphone  330  in distance from the acoustic source  305 . By placing the first microphone  310  and the second microphone  330  at different distances from the acoustic source  305 , the principle of the “inverse square law” is utilized. According to the inverse square law, the sound power detected at a given location from an acoustic source is inversely proportional to the square root of the distance between the given location and the acoustic source. Since sound pressure is proportional to the square root of sound power, sound pressure at the given location is inversely proportional to the distance between the given location and the acoustic source. 
     As further shown in FIG. 2A, the detector  370  detects the microphone output signal  400 , while the detector  380  detects the microphone output signal  405 . While it is possible to determine the average ratio of instantaneous acoustic pressures and use this information to estimate source proximity, in practice it is easier to smooth the waveform with rectifying detectors  370  and  380 . The detectors  370  and  380  must be well matched in gain and time response. The attack and release times should also be fast enough to follow the envelopes of the respective signals  400  and  405  but slow enough to smooth the waveform in the frequency band of interest. 
     The difference in path length from the desired acoustic source  305  to microphones  310  and  330  results in the peaks in the detected response in detectors  370  and  380  to be somewhat displaced in time. This, in turn, causes the calculated ratio to fluctuate. Smoothing the peaks results in a reduction in the amount of ripple in the calculated ratio. 
     It is desirable to further smooth the output of the divider  360  to prevent false triggering on the onset or end of a signal produced by a distant sound source. This occurs because during the time between the arrival of a sound at either microphone and its arrival at the other microphone, the ratio is calculated using the response of the first microphone  310  to the received signal and the noise floor of the other microphone  330 . This can result in very large ratios, either greater or less than one (1), depending on which microphone is stimulated first. (This is determined by the direction of the acoustic source  305 ). Note that this is true even if the time-aligned ratio is equal to unity. Once the signal ends, a similar situation occurs since one microphone continues to be stimulated for a brief interval after stimulation of the other microphone has ceased. This transient is in the direction of the reciprocal of the initial transient. Any known suitable component for inhibiting response to these transients and limiting abrupt changes in gain will improve the performance of the signal expander system of the present invention. 
     Referring now to FIG. 3, an analog implementation of the divider  360  is shown. A first logarithmic amplifier  450  is coupled between the positive (+) input of a differential amplifier  455  and the detector  370  (FIG.  2 A). A second logarithmic amplifier  460  is coupled between the negative (−) input of differential amplifier  455  and the detector  380  (FIG.  2 A). An anti-logarithmic amplifier  465  may be optionally coupled to the differential amplifier  455  output. The use of the logarithmic amplifiers  450  and  460  enables the differential amplifier  455  to produce a signal  470  proportional to the ratio of the magnitude of the input signals and improves the resolution of the proximity threshold estimation. The output signal  470  is generated either directly from the differential amplifier  455  output or the anti-logarithmic amplifier.  465  output, depending on whether or not the anti-logarithmic amplifier  465  is implemented. The anti-log amplifier  465  changes the characteristic of the output signal  470  for processing by a subsequent stage such as comparator  335  (FIG.  2 A). The output signal  470  indicates the ratio of the magnitudes of the microphone output signals  400  and  405  (FIG.  2 A). 
     Referring again to FIG. 2A, the divider  360  generates the output signal  470  that depends on the ratio of the magnitudes of microphone output signals  400  and  405 . The magnitude of the output signal  470  increases when the pressure at microphone  310  increases relative to microphone  330  but is independent of the absolute magnitude of the signals as long as the ratio between them remains constant. The comparator  335  compares the magnitude of the output signal  470  with the reference voltage level Vref from the reference voltage source  350 . The precise voltage of Vref produced by reference source  350  is dependent upon specific circuit values but would typically be set to equal the voltage produced by the output signal  470  when the desired acoustic source  305  is at the maximum distance (under conditions of normal use) from the microphones  310  and  330 . However, the reference voltage source  350  may be set to other values, based upon the expected noise level in the environment, the type of sounds generated by the acoustic source  305 , the relative distance between the microphones  310  and  330  from the acoustic source  305 , and/or other factors. If, for example, the microphones  310  and  330  are implemented in the headset  392  (FIG. 2B) or handset  394  (FIG.  2 C), then the relative distance between the microphones  310  and  330  from the acoustic source  305  (e.g., the speaker&#39;s mouth) can be determined. 
     When the acoustic source  305  is inactive, the microphone output signals  400  and  405  will be approximately equal in magnitude and will be voltage representations of ambient (far-field) noise that are detected by the microphones  310  and  330 , respectively. Thus, the output signal  470  will be approximately unit one in value. Assume that the reference voltage level Vref is set at a magnitude slightly above unit one. Since the magnitude of the output signal  470  is below the reference voltage level Vref, the comparator  335  will generate an output signal  490  with a low level, thereby not enabling the expander control stage  320 . Since the expander control stage  320  is not enabled, the variable gain block  315  does not add gain to the microphone output signal  400 . 
     Assume that a noise burst occurs in the environment in which the signal expander system  300  is located. The noise burst is detected by the second microphone  330  and, as a result, the magnitude of microphone output signal  405  will increase. The noise burst is also detected by the first microphone  310 . However, the air pressure due to the noise burst will be approximately equal at the locations of first microphone  310  and second microphone  330 , based on the inverse square law. The above approximation is based on the assumption that both microphones  310  and  330  can be treated as being approximately equal in distance from the source of the far field noise. As a result, the magnitudes of the microphone output signal  400  and microphone output signal  405  will be approximately equal. Therefore, the proximity estimation block  325  will generate an output signal  470  that is approximately unity or less than unity in value. Since the magnitude of output signal  470  will be less than the reference voltage level Vref, the comparator output signal  490  will have a zero or low value, thereby disabling the expander control stage  320 . Since the expander control stage  320  is not enabled, the variable gain block stage  315  does not add gain to noise burst detected by the first microphone  310 . 
     Assume that the acoustic source  305  becomes active. For example, the acoustic source  305  may be the speaker&#39;s mouth. The acoustic source  305  becomes active when the speaker starts to speak. Since the acoustic source  305  is active, the microphone output signal  400  of first microphone  310  will increase in magnitude where the first microphone  310  is disposed near the acoustic source  305 . The amount of difference between the sound pressures is dependent on the relative distances between the desired source and each microphone. The sound pressure from the acoustic source  305  will not be as high at the second microphone  330  due to the greater distance of that microphone from the desired acoustic source  305 . As a result, the magnitude of output signal  470  will increase and rise above the reference voltage level Vref. Since the magnitude of the output signal  470  is greater than Vref, the comparator  335  will generate a comparator output signal  490  with a high or non-zero value, thereby enabling the expander control stage  320 . In some implementations with closely spaced microphones, the difference between the microphone output levels will be relatively small but the difference is still measurable. 
     The operation of the expander control stage  320  is previously described above with reference to FIG.  1 B. The expander control stage  320  generates a gain control signal  495  that controls the gain provided by the variable gain block  315  and that has a value dependent on the magnitude of the microphone output signal  400 . Based on the value of gain control signal  495 , the variable gain block  315  will provide a corresponding amount of gain to the microphone output signal  400 . The resulting amplified microphone output signal or audio signal  322  is then produced by the variable gain block  315 . The gain provided by block  315  may be linear, logarithmic, or other non-linear function. 
     The gain curve of the variable gain block  315  is preferably based on psycho-acoustic principles and may, for example, be linear, logarithmic, or non-linear. As an option, the gain increase and/or gain decrease provided by the variable gain block  315  is gradual. 
     Any sound that has a significantly higher pressure at a position near first microphone  310  (when compared to a position near the second microphone  330 ) is presumed to be the desired sound from the acoustic source  305 . The proximity estimation block  325 , therefore, permits this desired sound to receive an increased gain from variable gain block  315 . Any sound that has an equal or lower pressure at a position near the first microphone  310  (when compared to a position near the second microphone  330 ) is presumed to be noise. The proximity estimation block  325 , therefore, will not permit the variable gain block  315  to amplify the noise. 
     It is further noted that an optional delay element  497  may be implemented along the path of signal  400  to compensate for delays in the ratio detection process, particularly in a digital implementation. 
     As stated above, a conventional subsequent stage (not shown) may also receive the comparator output signal  490  for subsequent processing. For example, the subsequent stage may be a conventional device for suppressing side tones. When the comparator output signal  490  is at a high level, the subsequent stage may set a particular volume level so that side tones are suppressed, as the speaker is speaking at the first microphone  310 . If the comparator output signal  490  is at a low level, the subsequent stage may set another volume level that is not required for side tone suppression. 
     The subsequent stage may also be other downstream modules or applications such as a conventional speech recognition circuit. The comparator output signal  490  may be used by a speech recognition circuit as an enable signal for translating speech sounds from first microphone  310  into text on a computer screen (not shown). When the comparator output signal  490  is zero in value, the speech recognition application is prevented from erroneously translating ambient noise detected by first microphone  310  into text on the computer screen. 
     In a preferred configuration, the rectifying detectors  370  and  380  are used to simplify the subsequent divider stage  360 . Without rectification, the divider  360  must operate in all four quadrants (i.e., with numerator, denominator, and output being allowed to be positive or negative). The design of such a divider is unnecessarily complex for this application. Rectification of the divider input signals allows the divider to operate in a single quadrant and results in a simpler circuit. 
     The band-pass filters  385  and  390  may be set so that only frequencies occurring in the speech envelope are passed through the proximity estimation block  325 . Typically, the frequency range of the band-pass filters would be about 200 Hz to 7 kHZ for speech applications. Reducing the upper cut-off frequency to 700 Hz would help reduce some of the ripple in the ratio detection. It is preferable that filters  385  and  390  be well-matched in gain and frequency response. The low frequency noise will be filtered by the band-pass filters  385  and  390 , thereby making the signal expander system  300  more robust for speech detection and noise rejection. 
     Reference is now made to the waveform diagrams in FIGS. 4A to  4 I. FIG. 4A shows a waveform  480  which represents a simulated output of acoustic source  305  (FIG.  2 A). The waveform  480  is shown as a function of pressure versus time (milliseconds). FIGS. 4B to  4 E illustrate various waveforms when the acoustic source  305  is close to the first microphone  310 . FIG. 4B shows the output waveform diagrams of each microphone  310  and  330  as a function of pressure versus time. The waveform  400  is the output of the first microphone  310  (FIG.  2 A), while the waveform  405  is the output of the second microphone  330  (FIG.  2 A). The axis denoted as “Mic 1 Pressure” is the pressure at the first microphone  310 , while the axis denoted as “Mic 2 Pressure” is the pressure at the second microphone  330 . 
     The waveform  405  is delayed approximately 0.284 milliseconds relative to the waveform  400 , for a path length difference of approximately 97.6 millimeters between the first microphone  310  and the second microphone  330 . It is noted that in the examples of FIGS. 4B to  4 E, an attenuation of approximately 13.81 dB of the waveform  400  (from the first microphone  310 ) corresponds to a distance of approximately 25.0 millimeters of the first microphone  310  from the acoustic source  305 , with an angle of approximately 90 degrees relative to the axis between microphones  310  and  330  and approximately 120 millimeters of spacing between the microphones. 
     FIG. 4C shows waveform diagrams of the detected output signals from each microphone  310  and  330  and the linear ratio  470   a  from the divider  360  (FIG.  2 A). The linear ratio  470   a  is shown as a function of the ratio value versus time. The waveform  400   a  is received by the first input  365  and the waveform  405   a  is received by the second input  375  of the divider  360 . The waveform  400   a  and waveform  405   a  are each shown as smoothed absolute values of signals from their respective microphones. 
     FIG. 4D shows the logarithm  400   b  of the smoothed absolute value of waveform  400   a  and the logarithm  405   b  of the smoothed absolute value of waveform  405   a.  These logarithmic signals are generated by the logarithmic amplifier  450  and logarithmic amplifier  460 , respectively, of FIG.  3 . The waveform  470   b  is the difference of the logarithms  400   b  and  405   b,  and is generated by the differential amplifier  455  of FIG.  3 . 
     FIG. 4E shows the output waveforms from the proximity estimation block  325  (FIG. 2A) when the acoustic source  305  is close to the first microphone  310 . The waveform  470   a  is the output of proximity estimation block  325  with the output signal from divider  360  (FIG. 2A) clipped at 20 dB. The waveform  470   b  is the output of proximity estimation block  325  with the output signal clipped at 20 dB and then smoothed by filtering (by use of clip and filter block  395  in FIG.  2 A). 
     FIGS. 4F and 4I illustrate various waveforms when the acoustic source  305  is distant from the first microphone  310 . FIG. 4F shows the output waveform diagrams of each microphone  310  and  330  as a function of pressure versus time. The waveform  405  is delayed approximately 0.007 milliseconds relative to the waveform  400 , for a path length difference of approximately 2.40 millimeters between the first microphone  310  and the second microphone  330 . It is noted that in the examples of FIGS. 4F to  4 I, an attenuation of approximately 0.01 dB of the waveform  400  corresponds to a distance of approximately 3.0 meters of the first microphone  310  from the acoustic source  305 , with an angle of approximately 90 degrees relative to the axis between microphones  310  and  330  and approximately 120 millimeters of spacing between the microphones. 
     FIG. 4G shows waveform diagrams of the detected output signals from each microphone  310  and  330  and the linear ratio  470   a  from the divider  360  (FIG.  2 A). The waveform  400   a  (to input  365  in FIG. 2A) and the waveform  405   a  (to input  375 ) are substantially overlapping, and are each shown as smoothed absolute values of signals from their respective microphones. 
     FIG. 4H shows the logarithm  400   b  of the waveform  400   a  and the logarithm  405   b  of the waveform  405   a  of FIG.  4 G. The waveform  470   b  is the difference of the logarithms  400   b  and  405   b  in FIG.  4 H. 
     FIG. 4I shows the output waveforms from the proximity estimation block  325  (FIG. 2A) when the acoustic source  305  is distant from the first microphone  310 . The waveform  470   a  is the output of proximity estimation block  325  with the ratio signal from the divider  360  (FIG. 2A) clipped at 20 dB. The waveform  470   b  is the output of proximity estimation block  325  with the output signal clipped at 20 dB and then smoothed by filtering. 
     FIG. 5 illustrates a signal expander system  500  in accordance with another embodiment of the present invention. The signal expander system  500  is a digital implementation that would convert the signals at the microphones  310  and  330  and implement the same functional blocks described above in a digital signal processor (DSP). Analog processes are replaced by digital arithmetic blocks and algorithms. The digital implementation shown in FIG. 5 also enables more sophisticated processes to be utilized without requiring additional hardware. The signal expander system  500  includes an analog-to-digital (A/D) converter (not shown) coupled between each of the microphones  310  and  330  and the proximity estimation block  510 . The A/D converters convert the output of microphones  310  and  330 , respectively, into digital form. 
     A digital comparator  530  compares the values of the output signal  535  from proximity estimation block  510  with a threshold value Vref(digital) from a threshold register  540 . The digital comparator  530  will generate a comparator output signal  545  with a logical one value if the value of output signal  535  is greater than the threshold value Vref(digital). The comparator output signal  545  is then used for controlling the variable gain block  315 . 
     FIG. 6 illustrates a signal expander system  600  in accordance with another embodiment of the present invention. The output signal  470  from proximity estimation block  325  is fed into a control function block  715  for permitting linear, logarithmic, or non-linear gain to be provided by the variable gain block  315 . The control function block  715  has been shown as a separate element in order to assist in explaining the functionality of the present invention. However, the block  715  may be implemented within the variable gain block  315 . The block  715  may also be implemented in the variable gain block  315  shown in FIGS. 2A and 5. 
     FIG. 7 illustrates a signal expander  700  in accordance with another embodiment of the present invention. The positive input terminal (+) of a differential amplifier  705  receives the microphone output signal  400  from first microphone  310 . The negative input terminal (−) of differential amplifier  705  receives the microphone output signal  405  from the second microphone  330 . The differential amplifier  705  generates the output differential signal  710  which is an amplification of the difference between the microphone output signals  400  and  405 . Therefore, the microphones  310  and  330  are both used for providing the acoustical output  322  of signal expander  700 . The second microphone  330  is being used to create a second order (bi-directional or gradient) microphone for increased noise rejection capability by the signal expander  700 . In this embodiment, the microphones are closely spaced and located in a common housing (not shown). This method of creating a directional microphone is well established and confers the advantages of the use of a directional microphone in suppressing background noise in addition to attaining the advantages of the signal expander circuit. 
     The output differential signal  710  from differential amplifier  705  is amplified by the variable gain block  315 . The amount of amplification provided by the variable gain block  315  depends on the output signal  470  from proximity estimation block  325 . The output signal  470  can be fed directly as a gain control signal to the variable gain block  315 , or fed into the control function block  715  as previously described above. 
     The signal expander system  700  may optionally include microphone adjustment stages  725  and  730  coupled to the first microphone  310  and second microphone  330 , respectively. The microphone adjustment stages  725  and  730  permit the sensitivities of microphones  310  and  330  to be adjusted and substantially matched. In other words, the microphone adjustment stages  725  and  730  permit the microphones  310  and  330  to be equally sensitive to sound. The microphone adjustment stages  725  and  730  may also be implemented in the above-mentioned signal expander systems in accordance with the present invention. 
     In addition, an optional delay stage  731  may be coupled between microphone adjustment stage  730  and the negative (−) input of differential amplifier  705 . The delay stage  731  compensates for the delay between the output signals of the microphones  310  and  330 . 
     For each of the above-mentioned embodiments of the signal expander systems in accordance with the present invention, the direction of the acoustic source  305  is preferably known so that it can be determined which microphone is nearest to the acoustic source  305 . Additionally, for each of the above-mentioned signal expander systems, the microphones  310  and  330  are preferably omni-directional microphones. Alternatively, the microphones  310  and  330  can be bi-directional microphones or cardioid unidirectional microphones, as long as the patterns and sensitivities of both microphones are substantially matched. The microphone adjustment stages  725  and  730  in FIG. 7 may be coupled to the microphones  310  and  330 , respectively, for matching the microphone sensitivities. As a further alternative, the microphone nearest the acoustic source  305  (e.g., first microphone  310 ) may be set at a higher gain level than the microphone further from the acoustic source  305 . As a result, the signal expander systems in accordance with the present invention can further have increased ambient noise rejection capability. 
     The analog signal expander systems in FIGS. 6 and 7 may also be modified in a digital implementation. Analog-to-digital converters may be coupled to the microphone  310  and  330  outputs, and a digital divider may be used to determine the ratio of the digitized microphone outputs, as similarly described above with reference to FIG.  5 . 
     Thus, while the present invention has been described herein with reference to particular embodiments thereof, a latitude of modification, various changes and substitutions are intended in the foregoing disclosure, and it will be appreciated that in some instances some features of the invention will be employed without a corresponding use of other features without departing from the scope of the invention as set forth.