Patent Abstract:
An active noise suppression system for use in noisy environments includes a dual microphone noise suppression system in which the echo between the two microphones is substantially canceled or suppressed. Noise is cancelled by the use of first and second line echo cancellers, which model the delay and transmission characteristics of the acoustic path between the two microphones. In a first embodiment, a noise suppression system acts as an ear protector, canceling substantially all or most of the noise striking the dual microphones of the ear set. In a second embodiment, a noise suppression system in accordance with the present invention acts a noise suppression communication system, suppressing background noise while allowing speech signals to be heard by the wearer.

Full Description:
FIELD OF THE INVENTION 
     The present invention relates to the field of acoustic devices. In particular, the present invention relates to a noise suppression system for use in noisy environments. 
     BACKGROUND OF THE INVENTION 
     Many environments have unwanted noise. For example, factory machinery, aircraft engines, motor vehicle traffic and the like generate noise levels that can be detrimental to hearing and interfere with voice communication. 
     The primary component of unwanted noise is the direct sound wave d(t) from the noise source. The secondary component of unwanted noise is the echo of the direct sound wave off a reflecting surface, such as the exterior surface of a building or an interior wall. In open environments, noise is primarily direct noise. In a typical environment, there is some reflection (echo) of the primary sound wave off buildings or walls, which adds reflected noise to the original direct noise. In confined enclosed environments the interior surfaces and surfaces of objects contained inside the enclosed environment generate multiple echoes r(t) of the same sound wave. Multiple echoes of the reflected sound wave combined with the direct sound wave d(t) is the noise s(t) captured by the microphone. 
     One example of an environment in which reverberation noise is considered a significant problem is mining. Mines are typically located underground in closed quarters surrounded by reflecting walls of substantially homogenous materials. Powerful mining equipment generating acoustic waves is used on a daily basis. The noisy environment makes voice communication between mine workers very difficult. In addition, the accumulation of direct acoustic waves and their reverberation from the inner surfaces of the mine tunnel and other mining equipment in the tunnels leads to a high noise level detrimental to the ear. The risk of hearing loss after long exposure to high ambient noise levels has been well documented. 
     Noise Cancellation 
     Various devices have been proposed to reduce noise levels. One of the most direct means for reducing the sound intensity is to surround the source of the noise with acoustic baffles. Such baffles placed on or in front of reflecting walls and other objects, cut off the reflecting acoustic propagation path. Various absorbing materials dissipate incident sound energy by converting it to heat energy. Sound absorbers work well for the high frequency range. However, acoustic baffles are bulky and do not work well for low frequencies. In certain industrial environments such as mining, acoustic baffles are not practical. 
     Active noise cancellation, where the cancellation of noise is sought by emitting an artificial sound to cancel the unwanted sound, is known. An active noise cancellation system uses a microphone, an amplifier and a loud speaker, in an arrangement to cancel the sound in a particular area, typically the area in the vicinity of an operator. The microphone provides a measure of the noise in a local area relatively distant from the direct noise source. The amplifier drives the loudspeaker to produce equal amplitude and opposite phase acoustic signal to cancel out the sound in the local area. Although a significant sound reduction is experienced, it is experienced only for that particular area and no other areas where the sound may be equally objectionable. In addition, such an arrangement is prone to the production of interference patterns, which may even increase the noise intensity in other locations. 
     A variation of the above system includes a second microphone disposed at a noise receiving point. The output of the second microphone is a measure of the cancellation error, which is used to adjust the coefficients of an adaptive filter in a closed loop recursive system to further reduce the noise received at the second microphone. 
     In another type of active noise cancellation system a microphone is placed very close to the acoustic noise source, which is approximated a point source. The signal processing circuit produces a phase opposition signal, which is adjustable by adjusting the distance between the microphone and the loudspeaker. Such systems are restricted to a point source of acoustic radiation of a single frequency, and do not work well when the noise is produced by large vibrating surfaces that may be vibrating in a complex mode to produce a wide spectrum of frequencies. 
     Another type of active noise cancellation system uses a pair of microphones and a headset worn by the operator. A first microphone picks up a first sample of the background noise. A second microphone placed some distance away from the first microphone, picks up a second sample of the background noise. To cancel the noise, the signal from the second microphone is processed in an adaptive filter and combined in opposite phase relationship to the signal from the first microphone. The processed second signal from the second microphone tends to cancel the noise signal arriving at the first microphone. The headset actively reduces the level of noise reaching the ears, thereby providing ear protection for workers when worn in high noise areas. However, such headsets prevent workers from hearing alarm signals and block speech communication between workers. 
     In general, prior art noise cancellation systems do not work well in relatively high background noise environments with complex reverberating structures especially in confined spaces, such as are commonly found in the mining industry. 
     Enhancing Speech Communication 
     In addition to ear protection, noise suppression systems are used in communications systems to help workers hear speech signals in noisy environments. Noise reduction communication systems distinguish the desired speech component from the background noise component of the combined signal. By canceling or reducing the background noise component, the signal-to-noise ratio is increased thereby enhancing the quality of the received speech. 
     One type of noise suppression system uses a pair of microphones connected to a headset worn by the operator. A first microphone (for voice) picks up a first signal containing the intended speech plus the background noise. A second microphone (for noise) placed some distance away from the first microphone, picks up a sample consisting mostly of the background noise and less of the speech signal. The signal from the second microphone (background noise) is processed in an adaptive filter and subtracted from the signal from the first microphone (speech plus the background noise) to reduce or cancel the background noise component of the first signal. 
     Since the second microphone is placed some distance away from the first microphone, the background noise sample (at the second microphone) is not exactly the same background noise signal that is arriving at the first microphone. The function of the adaptive filter is to compensate for the difference in acoustic paths of background noise arriving at the first and second microphones. 
     U.S. Pat. No. 5,754,665 to Hosoi shows a dual noise canceller with dual microphones and dual adaptive filters intended for use in an automobile telephone speaker system. First and second microphones are placed near the driver and passenger, respectively. When one microphone is used for conversation, the other microphone is used for collecting noise, and vice versa. A noise-reduced version of the first voice signal is obtained by using one of the adaptive filters. When the second microphone is used for conversation, the first microphone is used for collecting noise. A noise-reduced version of the second voice signal is obtained by using the second adaptive filter. The two noise reduced versions are added to form the outgoing telephone voice signal. 
     SUMMARY OF THE INVENTION 
     In order to cancel unwanted noise, it is necessary to obtain an accurate estimate of the noise to be cancelled. In an open environment, where the noise source can be approximated as a point source, microphones can be spaced far apart as necessary and each will still receive a substantially similar estimate of the background noise. However in a confined environment containing reverberation noise caused by multiple sound reflections, the sound field is very complex and each point in the environment has a very different background noise signal. The further apart the microphones are, the more dissimilar the sound field. As a result, it is difficult to obtain an accurate estimate of the noise to be cancelled in a confined environment by using widely spaced microphones. 
     If the two microphones are moved closer together, the second microphone should provide a better estimate of the noise to be cancelled in the first microphone. However, if the two microphones are placed very close together, each microphone will cause an additional echo to strike the other microphone. That is, the first microphone will act like a speaker (a sound source) transmitting an echo of the sound field striking the second microphone. Similarly, the second microphone will act like a speaker (a sound source) transmitting an echo of the sound field striking the first microphone. Therefore, the signal from the first microphone contains the sum of the background noise plus a reflection of the background noise, which results in a poorer estimate of the background noise to be cancelled. 
     The present invention is embodied in a dual microphone noise suppression system in which the echo between the two microphones is substantially canceled or suppressed. Reverberations from one microphone to the other are cancelled by the use of first and second line echo cancellers. Each line echo canceller models the delay and transmission characteristics of the acoustic path between the first and second microphones. 
     The present invention is further embodied in an ear set to be worn in the outer ear. The ear set is a self-contained molded unit, with integral dual microphones, battery, ear canal speaker, signal processing electronics that is convenient to wear and will not interfere with communication between workers or physical activity while working. 
     In a first embodiment, a noise suppression system in accordance with the present invention acts as an ear protector, canceling substantially all or most of the noise striking the dual microphones of the ear set. In a second embodiment, a noise suppression system in accordance with the present invention acts a noise suppression communication system, suppressing background noise while allowing communication signals to be heard by the wearer. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS 
     FIG. 1 is a block diagram of a dual echo predictive line canceller used in conjunction with the present invention. 
     FIG. 2 is a pictorial representation of the sound field reaching an ear set in accordance with the present invention intended to be worn in the human ear. 
     FIG. 3 is a pictorial representation of the reverberation noise field in confined spaces. 
     FIG. 4 is a diagram illustrating the various paths that reverberation sound reaches the dual microphones of a noise suppression system in accordance with the present invention. 
     FIGS. 5 and 6 illustrate the reverberations between the dual microphones of the present invention. 
     FIG. 7A is a dual echo line canceller embodying the present invention. 
     FIG. 7B is a block diagram of an echo prediction circuit for the dual echo line canceller of FIG. 7A in accordance with the present invention. 
     FIG. 8 is a noise suppression system in accordance with a first embodiment of the present invention. 
     FIG. 9 is a noise suppression communication system in accordance with a second embodiment of the present invention. 
     FIG. 10 is an alternate scheme for a noise suppression system in accordance with a second embodiment of the present invention. 
    
    
     DETAILED DESCRIPTION 
     FIG. 1 is a general purpose block diagram a dual microphone acoustic noise suppression (ANS) system. First and second microphones, mic 1  and mic 2 , are coupled to a dual echo predictive line canceller  10 . The concept of ANS is based on the cancellation of noise in one microphone by means of the other microphone. In the prior art, the electronic portion  10  of an ANS system was first developed using an analog system. Such systems were much too bulky to be fitted into an ear set. 
     Each noise source (A or B) projects a different direct sound wave along different paths to mic 1  and mic 2 . The acoustic path from noise source A to mic 1  is represented by a transfer function E 2 (z). The acoustic path from noise source A to mic 2  is represented by a transfer function E 1 (z). Between mic 1  and mic 2  the acoustic path is represented by a transfer function E 3 (z). 
     FIG. 2 shows an ear set  14  embodying the present invention. The ear set  14  contains an ear canal speaker  12 , which is coupled to the human ear  36 . The ear set  14  further includes a pair of microphones, mic 1  and mic 2  closely mounted on the ear set  14 . Sound from a given source  21  reaches mic 1  and mic 2  by direct paths  26 ,  16  respectively. Sound from source  21  also reaches mic 1  by various reflecting paths. In particular, a sound wave  28  reflecting off a neighboring wall  23  reaches mic 1  as a reflected sound wave  30 . In addition, a sound wave  32  reflecting off a neighboring wall  23  reaches mic 1  as a reflected sound wave  34 . 
     With respect to mic 2 , sound from the source  21  arrives via a variety of paths. In particular, a sound wave  22  reflecting off a neighboring wall  23  reaches mic 2  as a reflected sound wave  24 . Yet another sound wave  20  from a different direction arrives at mic 2  via a sound wave  18  reflected off an opposite wall  25 . Thus, the sound fields at mic 1  and mic 2  contain a complex mixture of the original sound with many echoes. 
     The situation in a confined space is illustrated further in FIG. 3 in which a sound source  40  includes a direct path  44  and a plurality of reflecting paths such as  46 A,  46 B,  48 A,  48 B and  50 , known as reverberation (or reverberating) noise. 
     The relationship of the microphones to the ear set is illustrated in FIG.  4 . For simplicity, FIG. 4 is a simplified representation to the model illustrated in FIG. 1 where mic 1  acts as an echo source generator transmitting the signal toward mic 2 . The dual microphones mic 1  and mic 2  are fixed on the same axis  72  on either side of the ear set, perpendicular to a direct path  70  to the ear set. 
     In a homogeneous medium, each of the microphones will receive a reverberant sound. A sound wave  64  reflecting off a neighboring wall  53  reaches mic 1  as a reflected sound wave  68 , which tends to cancel a sound wave  60  reflecting off a neighboring wall  55  reaching mic 1  as a reflected sound wave  62 . Similarly, a sound wave  52  reflecting off a neighboring wall  53  reaches mic 2  as a reflected sound wave  54 , which tends to cancel a sound wave  56  reflecting off a neighboring wall  55  reaches mic 2  as a reflected sound wave  58 . All reverberant sound waves will tend to cancel each other at each microphone, except the reverberant sound wave r 3 (t) along the echo path from mic 1  to mic 2 . The reverberant sound wave r 3 (t) captured by mic 1  is out of phase with the reverberant sound wave −r 3 (t) captured by mic 2 . 
     Depending on the position of the noise source  51 , the received direct sound by each microphone will be a delayed version of the other. The direct sound wave d 1 (t) at one microphone is a delayed version of the direct sound wave d 2 (t) at the other microphone. The direct sound wave received directly from the source will be substantially similar if the noise source  51  were relocated along the perpendicular axis  70 , equidistant from the two microphones, i.e., d 1 (t)=d 2 (t). 
     A simplified representation of the mutual echoes is illustrated in FIGS. 5 and 6. In FIG. 5, mic 2  acts as an echo source generator  512  transmitting the noise signal {circumflex over (d)} 2 (t) toward mic 1 . In FIG. 6 the process is reversed, where mic 1  is acting as an echo source generator transmitting noise signal {circumflex over (d)} 1 (t) toward mic 2 . A line echo canceller is implemented in order to duplicate the noise signal flowing through the inter-microphone acoustic path (E 3  in FIG.  1 ). 
     As indicated, the noise captured in mic 2  includes the echo from mic 1  and vise versa. Similar to the task to be performed by an echo canceller, s 1 (t) in FIG. 5 has a term to be cancelled: i.e., {circumflex over (d)} 2 (t) (the delayed version of d 2 (t) including some reverberations) by having an estimate of d 2 (t). Therefore, an Acoustic Noise Suppressor (ANS) and the Line Echo Canceller (LEC) share the common problem of finding the best estimate of the microphone to microphone echo path E 3  (in FIG.  1 ). 
     A noise suppression system formed by a pair of echo line cancellers for use in conjunction with the present invention is shown in FIG.  7 A. Mic 1  is coupled to a first echo prediction adaptive filter  710  and a first adder  712 . Mic 2  is coupled to a second echo prediction adaptive filter  714  a second adder  718 . The output of the first adder  712  is used to subtract the predictive noise {circumflex over (d)} 2 (t) from s 1 (t). The output of the second adder  718  is used to subtract the predictive noise {circumflex over (d)} 1 (t) from s 2 (t). The residual error terms at the respective outputs of the first and second adders  712 ,  718  are summed in adder  716  to drive the output speaker  717 . Suitable analog to digital converters (not shown) sample the microphones at a 48 kHz sampling rate. 
     The echo prediction filters  710  and  714  are shown in further detail in FIG.  7 B. Each echo prediction filter takes an input signal s(t) and subtracts (in adder  726 ) a delayed filtered  724  version p(t) of the input signal s(t). The delay  722  is selected to be equal to the acoustic delay between mic 1  and mic 2 . The filtered version of the input signal is obtained by use of an adaptive filter  724 . The delayed and filtered signal p(t) is subtracted in adder  726  (subtraction by signed addition). The difference is the error signal e(t) used to adjust the adaptive filter  724  coefficients. At convergence, the adaptive filter  724  models the transfer function E 3 (z)of the acoustic path between mic 1  and mic 2 , in order to generate the predictive noise term, {circumflex over (d)} 2 (t). 
     Adaptive filtering is a well-known technique useful in many signal processing applications. Adaptive filters are typically used in a closed loop system in which some measure of error (an error term) is to be minimized. An adaptive filter has an input terminal, an output terminal and an error terminal. Adaptive filters internally implement a suitable algorithm (responsive to the error input) to adjust the parameters of the adaptive filter so as to minimize the error term. 
     The filtered least means-square error (LMS) algorithm is a well-known method for adapting a filter. The LMS algorithm is simple and robust, has been widely adopted in many applications. Typically, an adaptive filter is implemented using a finite impulse response (FIR) filter using a digital tapped delay line with adjustable filter coefficients. The LMS algorithm is used to adjust the values of the filter coefficients responsive to an error input. In the present invention, the adaptive filters are used in a closed loop feedback system in which the adaptive filters are adjusted to model the characteristics of the acoustic path between mic 1  and mic 2 . In this sense, the implementation of each half of FIG. 7A is like a telephone line echo canceller which compensates for the acoustic path coupling between the microphone and ear piece of a telephone handset. 
     In operation in FIG. 7, the parameters of the adaptive filter  710  are set to an initial estimate. To the extent that the output of the adaptive filter  710  is not equal to the delayed version of the same signal, an error term e 1 (t) at the output  719  is fed back to adjust the adaptive filter  710 . After successive iterations, the parameters of the adaptive filter  710  are adjusted so as to minimize the error term at the output  719 . 
     Similarly, the parameters of the adaptive filter  714  are set to an initial estimate. To the extent that the output of the adaptive filter  714  is not equal to the delayed version of the same signal, an error term e 2 (t) at the output  720  is fed back to adjust the adaptive filter  714 . After successive iterations, the parameters of the adaptive filter  714  are adjusted so as to minimize the error term at the output  720 . 
     Each microphone signal mic 1 , mic 2  is used by each respective adaptive filter  714 ,  710  to generate a replica of the echo called {circumflex over (d)}(t), which is subtracted from the other microphone signal (including the echo). The echo canceller generates the echo replica by applying the reference signal to an adaptive filter (tapped-delay-line), as shown. At convergence, the adaptive filter&#39;s transfer function is identical to that of the echo path between the two microphones. 
     The convergence and the stability of the system relies on the stability of the two line echo cancellers. The choice of a value for the step size parameter μ (used in the known LMS algorithm) is important for stability. A sufficient condition for convergence of the LEC algorithm is given by:                0   &lt;   μ   &lt;     2       λ   max          (     R   xx     )           ,           (   20   )                                
     where λ max  is the largest eigenvalue of the autocorrolation matrix. 
     The system of FIG. 7A will tend to cancel all noise without discriminating between unwanted sounds (background noise) and wanted sounds (speech). For any wanted disturbances (e.g., speech), a speech detector is utilized (not shown). 
     Detailed Embodiment of the Invention 
     In FIG. 8, the detailed version of FIG. 7A is an approach for canceling the echo in each microphone uses dual prediction circuits to predict the echoes p 1 (n) and p 2 (n). In particular, a delay element  812 , an adaptive filter  814  and an adder  816  form a first predictor circuit to predict p 1 (n) from mic 1  (via analog to digital converter  810 ). Similarly, a delay element  822 , an adaptive filter  824  and an adder  826  form a second predictor circuit to predict p 2 (n) from mic 2 . (via analog to digital converter  820 ). The output is formed by adders  818 ,  828  and  830  which drive the speaker  833  via a digital to analog converter  832 . 
     To predict the echo from mic 2  received by mic 1 , a delayed  812  version of the mic 1  signal is processed in an adaptive filter  814  and subtracted  816  from the signal from mic 1 . The delay  812  is set equal to the acoustic delay between mic 1  and mic 2 . At convergence, the parameters of the adaptive filter  814  have been adjusted so as to model the transmission characteristics of the acoustic path between mic 2  and mic 1 . Once having a predicted value for the echo from each microphone, each echo p 1 (n), p 2 (n) is subtracted  828 ,  818  from the signal s 2 (n), s 1 (n) received from the other microphone. Specifically, the predicted value of the mic 2  echo p 1 (n) in mic 1  is then subtracted  828  from the mic 2  signal. Similarly, the predicted value of the mic 1  echo p 2 (n) in mic 2  is then subtracted  818  from the mic 1  signal. 
     In operation, an A/D converter  810  converts the signal from mic 1  to digital form, which is then delayed in delay element  812 . The preset value of the delay  812  is a function of the spacing between microphone mic 1  and microphone mic 2 . The delay value is set equal to the time it takes a sound wave to travel between mic 1  and mic 2 . The delayed signal from mic 1  is processed in an adaptive filter  814 , which simulates the transfer characteristics of the acoustic path from mic 1  to mic 2 . The output of the adaptive filter  814  is subtracted  816  (using a signed addition convention for subtraction) from the mic 1  signal. To the extent that the error e 1 (n) is not equal to zero at the output of adder  816 , the coefficients of the adaptive filter  814  are adjusted using the LMS algorithm. At convergence, the output of the adaptive filter  814  is p 1 (n), a predicted (delayed) version of the echo at mic 2  received from mic 1 . 
     The predicted value of the echo from mic 1 , p 1 (n), is subtracted from the signal from mic 2  in adder  828  (using a signed addition convention for subtraction). In such manner, the (predicted) echo from mic 1  arriving at mic 2  is subtracted (cancelled) from the mic 2  signal, and appears at the output of adder  828 . 
     The operation of the second prediction circuit is similar. Specifically, A/D converter  820  converts the signal from mic 2  to digital form, which is then delayed in delay element  822 . The preset value of the delay  822  is also a function of the spacing between microphone mic 1  and microphone mic 2  and is set to the same delay value as delay  812 . The delayed signal from mic 2  is processed in an adaptive filter  824 , which simulates the transfer characteristics of the acoustic path from mic 2  to mic 1 . The output of the adaptive filter  824 , is subtracted  826  (using a signed addition convention for subtraction) from the mic 2  signal. To the extent that the error e 2 (n) is not equal to zero at the output of adder  826 , the coefficients of the adaptive filter  824  are adjusted using the LMS algorithm. At convergence, the output of the adaptive filter  824  is p 2 (n), a predicted (delayed) version of the echo at mic 1  received from mic 2 . 
     The predicted value of the echo from mic 2 , p 2 (n), is subtracted from the signal from mic 1  in adder  818  (using a signed addition convention for subtraction). In such manner, the (predicted) echo from mic 2  arriving at mic 1  is subtracted (cancelled) from the mic 1  signal, and appears at the output of adder  818 . 
     The outputs of adders  818  and  828  are summed in adder  830  and form the signal output to drive speaker  833 . The circuit of FIG. 8 is a noise suppression system used primarily for ear protection. Substantially all noise will tend to be cancelled. 
     Speech Communication System 
     A noise suppression system that allows speech signals to be heard while suppressing background noise is shown in FIGS. 9 and 10. The noise suppression stage, which consists of dual prediction circuits and adders, is analogous to the noise suppression circuit shown in FIG.  8 . In particular, respective A/D converters  910 ,  920 , delay elements  912 ,  922 , adaptive filters  914 ,  924  and adders  916 ,  926 ,  918 ,  928  in FIG. 9 are connected and operate in the same manner as the corresponding A/D converters  810 ,  820 , delay elements  812 ,  822 , adaptive filters  814 ,  824  and adders  816 ,  826 ,  818 ,  828  in FIG.  8 . The noise suppression circuit is adaptive so long as the speech detector  913  does not detect speech. While speech is not present, respective AND gates  940 A,  940  couple the respective error signal outputs of adders  916 ,  926  to update the adaptive filter coefficients of the adaptive filters  914 ,  924 . 
     The output of adders  918  and  928  are connected to the input of a speech processing stage. The speech processing stage consists of two adaptive filters  930 ,  933 , adders  932 ,  936  and  934  and AND gates  940  and  942 . In FIG. 9, the speech processing stage conditions speech in independent adaptive filters  930 ,  933  before combining the processed speech signals in adder  934 . FIG. 10 shows an alternate embodiment of the speech processing stage. In FIG. 10 the operation of the adaptive filters  930 ,  933  are interrelated. In particular, the adaptive filters  930 ,  933  are cross coupled by connecting the output of adder  928  to the input of adder  932  (FIG. 10) instead of to the input of adder  936  (FIG.  9 ). Similarly, in FIG. 10 the adaptive filters  930 ,  933  are cross coupled by connecting the output of adder  918  to the input of adder  936  (FIG. 10) instead of to the input of adder  932  (FIG.  9 ). 
     A speech detector  913  coupled to mic 1  and mic 2  indicates when speech is present in the background noise. There are many known techniques to implement the speech detector  913 , including methods based frequency spectrum analysis, or time domain analysis. The output of adder  918  is coupled to a first adaptive filter  930  and a first adder  932 . The output of adder  928  is coupled to a second adaptive filter  933  a second adder  936 . The output of the first adder  936  is used as the error term e 4  to adjust the parameters of the second adaptive filter  933  via AND gate  942 . The other input of AND gate  942  is coupled to the signal that indicates speech is present. The output of the second adder  932  is used as the error term e 3  to adjust the parameters of the first adaptive filter  930  via AND gate  940 . The other input of AND gate  940  is coupled to the signal that indicates speech is present. 
     The residual error terms e 3  and e 4  at the respective outputs of the first and second adders  936 ,  932  are subtracted in adder  934  to drive the output speaker  938 . The speech processing stage enhances the resulting speech signal by taking the difference (e 3  minus e 4 ) between the two adder outputs  932 ,  936 . A suitable digital to analog converter converts the output of adder  934  to drive a speaker  938 . 
     In operation, when speech is not present, AND gates  940 ,  940 A,  942 ,  942 A permit each respective adaptive filter  930 ,  914 ,  933 ,  924  to use each respective error signal to update the respective coefficients. The adaptive filters  930 ,  914 ,  933  and  924  are continuously adjusted to cancel all sound as noise. As a result, input noise is cancelled by operation of the circuit. However, in order not to cancel the desired speech signal, the AND gates  940 ,  940 A,  942 ,  942 A are responsive to a speech present indication from the speech detector  913 , to suspend the update error function. In other words, when speech is present, the adaptive filters are “frozen” and do not adapt to cancel the desired speech signal. 
     When speech is detected, the AND gates  940 ,  940 A,  942 ,  942 A force the adaptive filters  930 ,  914 ,  933 ,  924  to stop adapting respective filter coefficients and keep the computed values equal to the values computed just prior to detection of speech. With the adaptive filter coefficients frozen, the subsequent speech is the error signal. Assuming that the background noise does not materially change while speech is present, the system output from the D/A converter to the speaker  938  is substantially equal to the input speech signal with the background noise suppressed.

Technology Classification (CPC): 7