Patent Publication Number: US-9406293-B2

Title: Apparatuses and methods to detect and obtain desired audio

Description:
RELATED APPLICATIONS 
     This application is a continuation-in-part of co-pending non-provisional U.S. patent application Ser. No. 12/157,426 entitled “Cardioid Beam With A Desired Null Based Acoustic Devices, Systems And Methods;” which is a continuation of U.S. patent application Ser. No. 10/206,242 (Now U.S. Pat. No. 7,386,135), entitled “Cardioid Beam With A Desired Null Based Acoustic Devices, Systems And Methods,” which claims priority to U.S. provisional patent application No. 60/309,462 entitled “Adaptive Noise Cancellation System,” filed on Aug. 1, 2001, U.S. patent application Ser. No. 12/157,426 entitled “Cardioid Beam With A Desired Null Based Acoustic Devices, Systems And Methods is hereby fully incorporated by reference. U.S. provisional patent application No. 60/309,462 entitled “Adaptive Noise Cancellation System,” is hereby fully incorporated by reference. 
    
    
     FIELD OF THE INVENTION 
     The present invention relates to the fields of acoustics and signal processing. More specifically, the present invention is related to audio devices, systems and methods for sensing and/or discerning desired audio in a noisy environment, where the environmental noises are statistically uncorrelated to the desired audio, and located at the directions other than the direction of the desired audio. 
     BACKGROUND OF THE INVENTION 
     Interference from background noises is one of the main barriers to advance acoustic applications or systems. whether it is audio acquiring microphone systems for communication or automatic speech recognition (ASR), hydrophone systems, sonar systems, or other acoustic systems of the like. The problem has found to be especially difficult with multiple background noise sources that are non-stationary, broadband, bursting and intermittent in a reverberant environment. 
     For example, in the case of ASR systems, it is increasingly desirable to introduce ASR technology to the large number of mobile communication devices, such as cell phones, car phones, and PDA, recently deployed as a result of the recent rapid advances in mobile communication and related technologies. However, most of these devices are frequently operated in a relatively noisy acoustic environment, such as on the street, in a car, bus, subway, train or airplane, or inside a noisy mail, factory or office. The background noises of these reverberant environments often exhibit the earlier mentioned non-stationary, broadband, bursting and intermittent characteristics. Resultantly, new applications utilizing speech recognition interface, whether for dictation or command-and-control, remain scarce. 
     To overcome these kinds of noise problems, others have resorted to close-talk handset, headset, or ear-set devices. However, these solutions introduce a number of inconveniences for the users. The wires of these additional headset/ear-set devices are often tangled with other objects. Wireless alternatives are more user friendly, however, they themselves have other limitations and inconveniences, e.g., higher cost. Multi-microphone arrays may avoid some of these limitations, however prior art multi-microphone arrays tend to be physically large, and unsuitable for most applications. 
     Consequently, there is a need for a more effective solution, especially one that is more compact that can allow a more natural human-machine interface that is hands-free, headset-free, and most importantly noise-free, for certain acoustic applications, such as ASR. In addition, noise reduction and/or cancellation preferably not only increase the clarity and intelligibility of the desired audio, but the reduction/cancellation may perhaps even reduce the load of digital communication networks, thereby resulting in more effective use of their capacities. 
     Other applications include noise robust headphone, teleconferencing system, digital voice recorder and hearing aid, etc. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       The present invention will be described by way of exemplary embodiments, but not limitations, illustrated in the accompanying drawings in which like references denote similar elements, and in which: 
         FIG. 1  illustrates an overview of the present invention, in accordance with one embodiment; 
         FIGS. 2 a -2 g    illustrates beam forming of the acoustic device of  FIG. 1  in further detail, in accordance with various embodiments; 
         FIG. 3  illustrates supplemental logic suitable for use in conjunction with the acoustic device of  FIG. 1  to generate the beam pattern of  FIG. 2 g   , in accordance with one embodiment; 
         FIG. 4 a    illustrates the acoustic beams generated by the various acoustic devices of  FIG. 2 a -2 c   , and  FIG. 2 g    (when supplemented with the circuitry of  FIG. 3 ) in the form of a polar sensitivity plots 
         FIG. 4 b    illustrates the acoustic beams of  FIG. 2 d -2 f    in the form of a polar sensitivity plot; 
         FIGS. 4 c -4 d    illustrate other possible primary signal beams in the form of polar sensitivity plots; 
         FIG. 5  illustrates the signal processing subsystem of  FIG. 1  in further detail, in accordance with one embodiment; 
         FIGS. 6 a -6 b    illustrate the sampling component of  FIG. 5  in further detail, in accordance with one embodiment; 
         FIGS. 7 a -7 b    illustrate a pre-whitening and a de-whitening component suitable for use as the optional signal conditioning and re-conditioning components of  FIG. 5  for certain acoustic applications, in accordance with one embodiment; 
         FIG. 8  illustrates the signal extraction component of  FIG. 5  in farther detail, in accordance with one embodiment; 
         FIG. 9  illustrates the mean amplitude computation component of  FIG. 8  in further detail, in accordance with one embodiment; 
         FIGS. 10 a -10 b    illustrate the detector components of  FIG. 8  in further details, in accordance with two embodiments; 
         FIGS. 11 a -11 e    illustrate the echo cancellation like logic of  FIG. 8 , in accordance with various embodiments; 
         FIG. 12  illustrates the operational logic of the inhibitor component of  FIG. 8 , in accordance with one embodiment; and 
         FIG. 13  illustrates the signal extraction logic of  FIG. 5 , in accordance with another embodiment. 
     
    
    
     SUMMARY OF THE INVENTION 
     Briefly, the present invention includes acoustic devices, systems and methods. 
     In accordance with one aspect, an acoustic device is formed with first and second plurality of one or more acoustic elements. The first plurality of one or more acoustic elements are designed and arranged to facilitate generation of a first signal that includes mostly undesired audio, substantially void of desired audio. The second plurality of one or more acoustic elements are designed and arranged to facilitate generation of a second signal that includes both the desired and undesired audio. 
     In one embodiment, the first one or more elements, in response to the presence of audio, desired or undesired, output a cardioid shaped acoustic beam with a null at the originating direction of the desired audio. The second one or more elements output an audio beam shaped in one of a number of complementary manner to encompass or maximizing the desired audio. 
     In accordance with another aspect, a signal processing subsystem is provided to extract the desired audio, using the two signals. 
     In various embodiments, the signal processing subsystems may practice various echo cancellation like signal extraction techniques, by introducing deterministic delays to the second signal, or practice blind signal separation techniques. 
     In various echo cancellation like signal extraction embodiments, pre-whitening and de-whitening components may be provided to condition and re-condition signals. 
     In one embodiment the desired audio is speech, in particular, speech to be recognized in a noisy reverberant environment, where noise is stochastic and uncorrelated to the desired speech, such as in an automobile or in an office. 
     DETAILED DESCRIPTION OF EMBODIMENTS 
     In the following description, various embodiments of the present invention will be described, in particular, ASR oriented embodiments. However, from the descriptions to follow, those skilled in the art would appreciate that the present invention is not limited to ASR only. The present invention may be practiced in other acoustic applications, including but are not limited to communication devices, recording devices, hearing aids, as well as hydrophones and sonar. 
     For purposes of explanation, specific numbers, materials and configurations are set forth in order to provide a thorough understanding of the present invention. However, it will be apparent to t hose skilled in the art that the present invention may be practiced with only some of these details, and/or with other elements. In other instances, well-known features are omitted or simplified. 
     Terminology 
     Parts of the description will be presented in acoustic and signal processing terms, such as acoustic beams, impulse, response, sampling, signal conditioning, signal extractions and so forth, consistent with the manner commonly employed by those skilled in the art to convey the substance of their work to others skilled in the art. As well understood by those skilled in the art, even in software implementations of some of the aspects of the present invention, these quantities take the form of electrical, magnetic, or optical signals capable of being stored, transferred, combined, and otherwise manipulated through electrical and/or optical components of a processor and its subsystems. 
     Part of the descriptions will employ various abbreviations, including but not limited to:
     ASR Automatic Speech Recognition   BSS Blind Signal Separation or Blind Source Separation   FFT Fast Fourier Transform FIR Feedback Impulse Response   IFFT Inverse Fast Fourier Transform   LMS Least Mean Square   NLMS Normalized Least Mean Square
 
Section Headings, Order of Descriptions and Embodiments
   

     Section headings are merely employed to improve readability, and they are not to be construed to restrict or narrow the present invention. 
     Various operations will be described as multiple discrete steps in turn, in a manner that is most helpful in understanding the present invention, however, the order of description should not be construed as to imply that these operations are necessarily order dependent. In particular, these operations need not be performed in the order of presentation. 
     The phrase “in one embodiment” is used repeatedly. The phrase generally does not refer to the same embodiment, however, it may. The terms “comprising”, “having”, “including” and other constructs of the like, are synonymous, unless the context dictates otherwise. 
     Overview 
     We refer now to  FIG. 1 , wherein a block diagram illustrating an overview of an acoustic system of the present invention, in accordance with one embodiment. As illustrated, for the embodiment, acoustic system  100  includes acoustic device  102  and signal processing subsystem  104 , both incorporated with the teachings of the present invention. The two elements are coupled to each other as shown. 
     Acoustic device  102 , in accordance with the present invention, is designed to respond to audio presence, desired and undesired (i.e. noise), by outputting two audio beams  103   a  and  103   b , with audio beam  103   a  having mostly undesired audio, substantially void of desired audio, and audio beam  103   b  having both the desired and the undesired audio. 
     The two acoustic beams (hereinafter, simply beams) are sampled by signal processing subsystem  104  to generate two corresponding audio signals (hereinafter, simply signals), which in turn are used by signal processing subsystem  104  to recover the desired audio, by removing the first signal corresponding to the first beam from the second signal corresponding to the second beam. 
     As will be appreciated by those skilled in the art, based on the descriptions to follow, acoustic device  102  may be formed compactly using as little as two acoustic elements, one each for the corresponding responsive generation of one of the two beams. Resultantly, the present invention is able to provide a more compact and user-friendly human interface for acoustic applications that have to address the issue of recovering desired audio from a complex noisy environment, such as in many ASR applications. 
     Further, under the present invention, audio beam  103   a  may be generated by an element with Cardioid beam, pattern, which has considerable sensitivity in all directions except the desired audio direction in space. Resultantly, unlike the prior arts where there are some “blind spot” directions with incomplete noise cancellation, the present invention may cancel noise coming from virtually any one of a number of directions. 
     Acoustic Device 
     As alluded to earlier, acoustic device  102  may be formed compactly using as few as two acoustic elements.  FIGS. 2 a -2 g    illustrate a number of these embodiments, from the perspective of the audio beams formed. However, the present invention is not so limited. In alternate embodiments, two or more acoustic elements may be used to responsively generate beams  103   a  and  103   b  instead. For ease of understanding, the description will primarily be presented in the context of the various two elements embodiments. Moreover, acoustic device  102  will simply be referred to as “microphone”. 
       FIGS. 2 a -2 c    illustrate three two-element embodiments, where one element  202  in response to the presence of audio generates a cardioid beam and the other element  204 , in response to the presence of audio generates an omni directional beam. In each of these embodiments, cardioid beam generating acoustic element (hereinafter, simply “mic”)  202 , is arranged with the null of the cardioid mic facing the expected originating direction of desired audio. 
     For the embodiment of  FIG. 2 a   , omni-directional beam generating mic  204  is arranged to face the expected originating direction of desired audio in parallel with cardioid beam generating mic  202 . For each of the embodiments of  FIG. 2 b -2 c   , omni-directional beam generating mic  204  is arranged to face the expected originating direction of desired audio in series with cardioid beam generating mic  202 . For the embodiment of  FIG. 2 b   , omni-directional beam generating mic  204  is arranged to be disposed “behind” cardioid beam generating mic  202 . For the embodiment of  FIG. 2 c   , omni-directional beam generating mic  204  is arranged to be disposed “in front of” cardioid beam generating mic  202  (both viewed from the perspective of the expected originating direction of the desired audio). 
       FIG. 4 a    illustrates the corresponding acoustic beams  103   a  and  103   b  (in the form of a “polar sensitivity” plot) responsively generated by elements  202  and  204  of the arrangements of  FIGS. 2 a -2 c   . As described earlier, beam  103   a  comprises a null facing in the originating direction of desired audio. For these embodiments, beam  103   b  “radiates” in all directions, and does not contain any null. 
     The null of cardioid beam generating element  202  is an attempt to eliminate the leakage of desired acoustic into beam  103   a . In reality, the null can often achieve as much as −20 dB attenuation relative to the sensitivity at the opposite direction. Nevertheless, experience has shown that the present invention still exhibits consistent improved results over that of the prior arts. 
     Typically, the two acoustic elements are disposed proximally adjacent to each other to enable the desired compact human interface be formed (for certain applications). For these applications, the separating distance between two discrete mic elements may be in the range as small as 0.2 cm to 1 cm. For semiconductor acoustic devices, the separating distance may be in the order of microns or even sub-microns. While care should be exercised to reduce the likelihood of cross interference between the elements, as illustrated by  FIG. 2 a -2 c   , their relative dispositions, i.e. whether facing the expected originating direction of desired audio in parallel or in series, are not as important as their respective beam patterns. 
       FIGS. 2 d -2 f    illustrate three alternate two-element embodiments, where both elements are cardioid beam generating mics  202   a - 202   b . In each of these embodiments, one of the two cardioid beam generating mics  202   a  is arranged with its null facing the expected originating direction of desired audio, and the other cardioid beam generating mic  202   b  arranged with its null facing away from the expected originating direction of desired audio. 
     For the embodiment of  FIG. 2 d   , the other cardioid beam generating mic  202   b  is arranged to have its null face away from the expected originating direction of desired audio, in parallel with the first cardioid beam generating mic  202   a . Similarly, for each of the embodiments of  FIG. 2 e -2 f   , the other cardioid beam generating mic  202   b  is also arranged with its null to face away from the expected originating direction of desired audio, except in series with the first cardioid beam generating mic  202   a.    
     For the embodiment of  FIG. 2 e   , the other cardioid beam generating mic  202   b  is arranged to be disposed “behind” the first cardioid beam generating mic  202   a , whereas for the embodiment of  FIG. 2 f   , the other cardioid beam generating mic  202   b  is arranged to be disposed “in front of” the first cardioid beam generating mic  202   a  (both viewed from the perspective of the expected originating direction of the desired audio). 
       FIG. 4 b    illustrates the corresponding acoustic beams  103   a  and  103   b  (also in the form of a “polar sensitivity” plot) responsive generated by elements  202   a  and  202   b  of the arrangements of  FIGS. 2 d -2 f   . As described earlier, beam  103   a  comprises a null facing in the originating direction of desired audio. For these embodiments, beam  103   b  comprises a null facing away from the originating direction of desired audio. 
       FIG. 2 g    illustrates yet another alternate two-element embodiment of acoustic device  102 . For this embodiment, two omni-directional beam generating mics  204   a  and  204   b  are used instead. The two elements  204   a  and  204   b  are arranged to face the expected originating direction of desired audio in series. The arrangement is supplemented with the circuitry of  FIG. 3  comprising delay  312 , amplifier  314  and adder  316 , implementing a “delay and sum” beam forming method. 
     As before, the responsive output of the second omni beam generating mic  204   b  provides beam  103   b . However, beam  103   a  is formed by having a delay added to the responsive output of the first omni beam generating mic  204   a , using delay  312 , amplified using amplifier  314 , and then subtracted from beam  103   b.    
     The delay should be chosen so that the cardioid null is sufficiently deep across all frequency in the bandwidth. The two acoustic elements may be balanced by adjusting the gain of amplifier  314 , to avoid mismatch and reduction of the null. 
     The circuitry of  FIG. 3  may be integrally disposed as part of acoustic device  102 , or it may be integrally disposed as part of signal processing subsystem  104 . In yet other embodiments, the role of the two omni beam generating mic  204   a  and  204   b  may be reversed. 
     Additionally, in addition to the “no null” and “single face away null” shape of  FIGS. 4 a  and 4 b   , beam  103   b  may comprise two or more nulls, as long as none of the nulls is facing the originating direction of desired audio. 
     For example,  FIG. 4 c    illustrates an alternate “clover leaf” beam shape (in a “polar sensitivity” plot) for beam  103   b  having two “leafs”, forming two nulls, with the two nulls facing two directions  406   a  and  406   b  that are substantially orthogonal to the originating direction of desired audio.  FIG. 4 d    illustrates yet another alternate “clover leaf” beam shape (in a “polar sensitivity” plot) for beam  103   b  having also two “leafs”, forming two nulls, with the two nulls facing two directions  406   c  and  406   d , each forming an obtuse angle with the originating direction of desired audio. 
     In summary, acoustic device  102  comprises two or more acoustic elements designed and arranged in a manner that facilitates generation of two signals with one signal comprising mostly undesired audio, substantially void of desired audio, and another signal comprising both desired and undesired audio. The two or more acoustic elements may e.g. respond to the presence of audio, desired and undesired, outputting a cardioid beam having a null facing the originating direction of desired audio, and another beam having any one of a number of complementary beam shapes (as long as it does not comprise a null facing the originating direction of desired audio). 
     Signal Processing Subsystem 
       FIG. 5  illustrates signal processing subsystem of  FIG. 1  in further detail, in accordance with one embodiment. As illustrated, for the embodiment, signal processing subsystem  104  comprises two channels of input (labeled as “reference” and “primary”), sampling components  502 , optional pre-extraction signal conditioning components  504 , signal extraction component  506 , and optional post-extraction signal conditioning component  508 . The elements are coupled to each other as shown. 
     Reference channel is employed to receive beam  103   a , whereas primary channel is employed to receive beam  103   b.    
     Sampling components  504  are employed to digitized beams  103   a  and  103   b . Typically, they are both digitized synchronically at the same sampling frequency, which is application dependent, and chosen according to the system bandwidth. In the case of ASR applications, the sampling frequency e.g. may be 8 kHz, 11 kHz, 12 kHz, or 16 kHz. 
     Typically, optional pre-extraction and post-extraction signal conditioning components  504  and  508  are application and/or extraction technique dependent. For example, in the case of ASR applications, and certain signal extraction techniques, such as echo cancellation like NLMS processing, pre-extraction and post-extraction signal conditioning components  504  and  508  may be pre-whitening and de-whitening filters. The pre-whitening and de-whitening filters are employed to level, and reverse the leveling of the spectrum density of both signals. Leveling of the spectrum density of both channels improve NLMS converging speed, for uneven frequency distribution of the signals. Other single channel noise cancellation technique, such as spectrum subtraction, can be added as additional stage of optional post extraction signal conditioning components. 
     Sampling Component 
       FIGS. 6 a -6 b    illustrate the sampling component of  FIG. 5  in further details, in accordance with two embodiments. For the embodiment of  FIG. 6 a   , sampling component  502  comprises two A/D converters  606 , one each for the two beams  103   a  and  103   b . Further sampling component  502  includes pre-amps  602  and anti-aliasing filters  604 . The elements are coupled to each other as shown. 
     The signal from each acoustic element is amplified by a corresponding pre-amp  602 , and then band-limited by a corresponding anti-aliasing filter  604 , before being digitized by a corresponding A/D converter at the sampling frequency Fs. 
       FIG. 6 b    illustrates an alternate embodiment, where only one A/D converter  606  is used. However, sampling component  502  further includes sample and hold components  608  and multiplexor  610 . The elements are coupled to each other as shown. 
     Each signal goes through the same processing as in  FIG. 6 a    until after anti-aliasing filtering (using anti-aliasing filters  604 ), then it is sampled by sample-and-hold (S/H) unit  608  to produce a discreet signal. The output is then multiplexed (using multiplexor  610 ) with the discreet signal from the other channel. Finally, the multiplexed signal is digitized by A/D converter  606  into digital signal at twice the sampling frequency (2×Fs). 
     Pre-Whitening and De-Whitening 
     As described earlier, for certain acoustic applications, such as ASR applications, which tend to have stronger lower frequency components than higher frequency components, it may be desirable to perform pre-extraction conditioning of the signals, such as spectrum density leveling through pre-whitening filtering, and therefore, post-extraction reverse conditioning, such as reversing the spectrum density leveling through de-whitening filtering. 
     For these applications, a pre-whitening filter (also referred to as de-colorization filter) is placed on both the primary and reference inputs before they are sent to signal extraction component  506 , in particular, if component  506  implements NMLS noise cancellation processing, to alleviate the potential slow convergence rate brought about by narrow band (highly auto-correlated) input signal. 
     One embodiment each of a pre-whitening filter, and a de-whitening filter is illustrated in  FIGS. 7 a  and 7 b    respectively. 
     For the embodiment of  FIG. 7 a   , pre-whitening filter  504  is in the form of a pre-emphasis filter characterized by the equation:
 
 y   n   =x   n   −α*x   n-1  
 
     For the illustrated implementation, pre-whitening filter  504  includes storage elements  702  and  704  for storing the preceding input value, x n-1 , and the constant α, and multiplier  706  as well as adder  708 . The elements are coupled to each other as shown, and collectively operate to implement the processing to compute output y n , per the above equation. 
     In alternate embodiment, pre-whitening filter  504  may also be implemented in software. 
       FIG. 7 b    illustrates the complementary de-whitening filter, in the form of a de-emphasis filter, characterized by the equation:
   y   n   =x   n   +α*y   n-1    
     For the illustrated implementation, de-whitening filter  508  includes storage elements  722  and  724  for storing the preceding output value, y n-1 , and the constant α, and multiplier  726  as well as adder  728 . The elements are coupled to each other as shown, and collectively operate to implement the processing to compute output y n , per the above equation. 
     Similarly in alternate embodiment, de-whitening filter  508  may also be implemented in software. 
     Signal Extraction Component 
       FIG. 8  illustrates signal extraction component of  FIG. 5  in further detail, in accordance with one embodiment. The embodiment implements an echo cancellation like technique to recover desired audio by removing the reference signal from the primary channel. The technique is referred to as “echo cancellation” like because similar to conventional “echo cancellation”, one signal is subtracted from another. However, in classical “echo cancellation”, the original signal that generates “echo” is accessible; and that original signal is not corrupted with the desired audio. Whereas under the present invention, the original noise signals are not available. Although the reference signal in the current invention is “substantially void of desired audio”, it still contains some desired audio. Extra steps, such as inhibition, should be taken to avoid the cancellation of the desired signal. In classical “echo cancellation”, the “echo” signal that is being subtracted from the composite signal with the desired audio and echo, and more importantly, the “echo” signal bears a natural deterministic time lagged relationship to the original audio that generated the echo. In contrast, under the present invention, it is the filtered reference signal, substantially void of the desired audio, being subtracted from the signal with both desired and undesired audio, and the reference and the desired signals are acquired virtually concurrently responsive to the presence of desired and undesired audio. 
     Accordingly, in addition to echo cancellation like logic  810 , signal extraction component  506  includes in particular a delay element  802  to artificially introduce a deterministic delay to the signal formed based on beam  103   b  (i.e. the signal on the primary channel). This artificially introduced delay enables modeling of reverberation between the acoustic elements of acoustic device  102 . Further, it enables adaptive FIR filters employed in the echo cancellation like signal processing technique to approximate a non-causal filter. 
     The amount of delay to be artificially introduced in order to model reverberation is application dependent. In general, it is approximately in the order of the duration of the impulse response of the environment, the delay then corresponds to an adaptive filter length which is approximately twice the impulse response of the environment in order to model reverberation; the delay allows the adaptive filter to model the reverberation history. In various applications, the amount ranges from 30 ms-60 ms for automotive environment, and 100 ms-200 ms for an office environment. 
     For the embodiment, the echo cancellation like extraction of desired audio is actually conditionally operated, only when the channels are considered to be both active. Thus, beside signal extraction logic  810  and delay element  802 , for the embodiment, signal extraction component  506  further includes mean amplitude estimation components  804 , channel signal detectors  806  and inhibition logic  808 . Channel signal detectors  806  are also collectively referred to as the “comparator” component, and in one embodiment, include in particular, two channel-active detectors, one each for the reference channel and the primary channel, and a desired audio detector. The elements are coupled to each other, and to the earlier enumerated elements as shown. 
     Mean amplitude estimator components  804  are employed to determine/estimate the power or amplitude of the signals of both channels for channel signal detectors  806 , i.e. channel-active detector and desired audio detector, as well as for the echo cancellation like signal extraction process. 
     In one embodiment, the echo cancellation like signal extraction process implemented is an adaptive noise cancellation process employing a NLMS FIR filter ( FIG. 11 a   ). In other embodiments, the echo cancellation like signal extraction processes implemented are adaptive noise cancellation processes employing a number of frequency domain LMS filters ( FIG. 11 b -11 c   ). In yet other embodiments, the echo cancellation like signal extraction processes implemented are adaptive noise cancellation processes employing a number subband LMS filter ( FIG. 11 d -11 e   ). 
     These elements are further described in turn below. 
     Mean Amplitude Estimator 
       FIG. 9  illustrates the mean amplitude estimation component of  FIG. 8  in further detail, in accordance with one embodiment. For the embodiment, mean amplitude estimation component  804  calculates a weighted running average of the absolute value of the input as characterized by the equation:
 
 y   n =(1−α)* y   n-1   +α*|x   n |
 
     The weight coefficient determines the length of the running window. 
     The embodiment includes various storage elements  902 - 908  for storing the values of |x n |, y n-1 , α, and (1−α) respectively, and multipliers  910  and adder  912  to perform the computations. 
     As with the earlier described pre-whitening and de-whitening components. mean amplitude estimation component  804  may also be implemented in software. 
     Detectors 
       FIGS. 10 a -10 b    illustrate the comparator component, i.e. detectors, of  FIG. 8  in further detail, in accordance with one embodiment. More specifically,  FIG. 10 a    shows the logic of a desired audio detector  806   a , whereas  FIG. 10 b    shows the logic of a channel active detector  806   b.    
     For the illustrated embodiment, desired audio detector  806   a  includes storage element  1002  for storing an audio threshold offset. Additionally, desired audio detector  806   a  further includes ratio calculator  1004 , long term running mean amplitude ratio value calculator  1006 , adder  1008  and comparator  1010 . The elements are coupled to each other as shown. The embodiment is a power based detector. 
     Ratio calculator  1004  is employed to calculate the ratio of the primary and reference signal mean amplitude. Running mean amplitude ratio calculator  1006  is employed to calculate the long term running mean value of the ratio, which provides the base or floor for the desired audio. Comparator  1010  and adder  1008  are employed to compare the current ratio to determine whether it is greater than the long term running ratio by at least a threshold offset. If it is above the base by at least the threshold offset, desired audio is considered detected; otherwise no desired audio is assumed. 
     The embodiment is designed for desired audio that tends to exhibit a bursty characteristic, such as speech. For other audio applications, suitably modified embodiments may be employed instead. 
     In alternate embodiments, other desired audio detectors, e.g. correlation based desired signal detector, may be employed instead.  FIG. 10 b    shows a channel-active detector in further detail, in accordance with one embodiment. The embodiment is a power based comparator. As illustrated, channel-active detector  806   b  comprises storage element  1024  for storing a threshold value, and comparator  1026  for comparing the mean amplitude of the channel to the stored threshold value. If it&#39;s above the stored threshold value, the channel is assumed to be active; otherwise the channel is assumed to be inactive. 
     Further, as with the earlier described pre-whitening/de-whitening and mean amplitude estimation components, detectors  804   a - 804   b  may also be implemented in software. 
     Inhibit 
       FIG. 12  illustrates the operating logic of the inhibit component of  FIG. 8  in further details, in accordance with one embodiment. As illustrated, for the embodiment, designed for time domain implementation, inhibit component  808  using the inputs provided detectors  806  first determines whether both the primary and the reference channels are active, blocks  1202 - 1204 . If either the primary or the reference channel is determined to be inactive, the inhibit signal is set to “positive”, resulting in substantial inoperation of the signal extraction block (e.g. to conserve computing power), i.e. no signal extraction (filtering) or adjustment to the extraction (adaptation) is performed. Under the condition, for the embodiment, whatever signal is present on the primary channel is outputted, block  1210 . 
     However, if both channels are active, inhibit logic  808  further determines if either desired audio is present or a pause threshold (also referred to as hangover time) has not been reached, block  1206 - 1208 . The pause threshold (or hangover time) is application dependent. For example, in the case of ASR, the pause threshold may be a fraction of a second. 
     If the desired audio is detected or the pause time is not exceeded, the inhibit signal is set to “positive with filter adaptation disabled”, i.e. filtering coefficients frozen, block  1212 . The reference signal is filtered accordingly, and subtracted from the primary channel to generate desired audio. 
     If the desired audio is not detected, and the pause time is exceeded (but the channels are active), the inhibit signal is set to “negative with filter adaptation enabled”, block  1214 . Under the condition, the filtering coefficients of the employed filters will be adapted. 
     Note that the above described embodiment advantageously employs the detectors and inhibition before the primary signal (having the desired audio) delay, thus the likelihood of the desired audio negatively impacting the filter adaptation operation is reduced. 
     As alluded to earlier, described in more detail below, filtering may also be performed in the frequency and subband domains. For these embodiments, the above inhibit implementation may be practiced on a frequency by frequency or subband by subband basis. 
     Echo Cancellation Like Signal Extraction Component 
       FIGS. 11 a -11 e    illustrate echo cancellation like signal extraction component of  FIG. 8  in further detail, in accordance with various embodiments. More specifically,  FIG. 11 a    illustrates an implementation employing a NLMS adapted approach, whereas  FIGS. 11 b -11 c    illustrate two implementations employing a frequency domain LMS adapted approach.  FIG. 11 d -11 e    illustrate two implementations employing a subband LMS adapted approach. 
     As illustrated in  FIG. 11 a   , echo cancellation like signal extraction component  810  of the NLMS adaptive implementation comprises adaptive FIR filter  1102  and adder  1104 . The elements are coupled to each other as shown. 
     The (conditioned) signal of the reference channel is filtered by adaptive FIR filter  1102 , and subtracted from the delayed signal of the primary channel, using adder  1104 . The result is outputted as the desired audio. 
     The extraction logic operates as a loop running on a sample-by-sample basis. The reference signal is filtered by the adaptive FIR filter  1102 . Essentially, a transfer function is applied to the reference channel to model the acoustic path from the cardioid element to the other element, so that the filtered reference signal closely matches the noise component of the signal in the primary channel. The filtered reference signal is then subtracted from the delayed primary signal. What is left, is the desired audio. 
     The output of the NLMS is also called the NLMS error; it is used to adjust the adaptive FIR filter coefficients so that the NLMS error will be minimized when the desired audio is not present. 
     As illustrated in  FIG. 11 b   , echo cancellation like signal extraction component  810  of the first frequency LMS implementation comprises FFT components  1112 , a number of adaptive filters  1114  (two shown), a number of adders  1116  (two shown) and IFFT component  1118 . The elements are coupled to each other as shown. 
     The (conditioned) signals of the reference channel and the delayed primary channel are first “decomposed” into a number of frequency components (two shown), by the corresponding FFT components  1112 . Each of the frequency components of the reference signal is filtered by a corresponding adaptive filter  1114 , and subtracted from the corresponding frequency component of the delayed signal of the primary channel. using a corresponding adder  1116 . The resulted frequency components are “recombined”, using IFFT component  1118 , and the recombined signal is outputted as the desired audio. 
     As illustrated in  FIG. 11 c   , echo cancellation like signal extraction component  810  of the second frequency LMS implementation comprises FFT components  1122   a - 1122   b , a number of adaptive filters  1124  (two shown), adder  1128  and IFFT component  1126 . The elements are coupled to each other as shown. 
     The (conditioned) signal of the reference channel is first “decomposed” into a number of frequency components (two shown), by FFT component  1122   a . Each of the frequency components of the reference signal is filtered by a corresponding adaptive filter  1124 . The filtered frequency components are recombined into a filtered reference signal, using IFFT component  1126 , which is then subtracted from the delayed signal of the primary channel, using adder  1128 , to generate desired audio. 
     The error signal (comprising the desired audio, if present) is also “decomposed” into a number of frequency components, using FFT component  1122   b , and the “decomposed” frequency components are used to adapt filters  1124 . 
     As illustrated in  FIG. 11 d   , echo cancellation like signal extraction component  810  of the first subband LMS implementation comprises analysis banks  1132   a - 1132   b , a number of down-sampling units  1134   a - 1134   b  (two sets of two shown), a number of adaptive filters  1136  (two shown), a number of adders  1138  (two shown), a number of up-sampling units  1140  (two shown), and a synthesis bank  1142 . The elements are coupled to each other as shown. 
     The (conditioned) signals of the reference channel and the delayed primary channel are first “decomposed” into a number of subband components (two shown), by the corresponding analysis banks  1132   a / 1132   b . Each of the subband components of the reference signal is first down-sampled by a predetermined factor, using a corresponding down-sampling unit  1134 , and then filtered by a corresponding adaptive filter  1136 . Each of the filtered subband components is then subtracted from the corresponding subband component of the delayed signal of the primary channel, using a corresponding adder  1138 . The resulted subband components are up-sampled by the same factor, using a corresponding up-sampling unit  1140 , and then “recombined”, using synthesis bank  1142 . The recombined signal is outputted as the desired audio. 
     As illustrated in  FIG. 11 e   , echo cancellation like signal extraction component  810  of the second subband LMS implementation comprises analysis banks  1152   a - 1152   b , a number of down-sampling units  1154   a  and  1154   b  (two sets of two shown), a number of adaptive filters  1156  (two shown), a number of up-sampling units  1158  (two shown), synthesis bank  1160  and adder  1162 . The elements are coupled to each other as shown. 
     The (conditioned) signal of the reference channel is first “decomposed” into a number of subband components (two shown), by analysis bank  1152   a . Each of the subband components of the reference signal is down sampled by a predetermined factor, using a corresponding down sampling unit  1154   a , and then filtered by a corresponding adaptive filter  1156 . The filtered subband components are up-sampled by corresponding up-sampling units  1158 , and then recombined into a filtered reference signal, using synthesis bank  1160 , which is then subtracted from the delayed signal of the primary channel, using adder  1162 , to generate desired audio. 
     The error signal (comprising the desired audio, if present) is also “decomposed” into a number of subband components, using analysis bank  1152   b  and down sampling unit  1154   b , and the “decomposed” subband components are used to adapt filters  1156 . 
     Each of these signal extraction component embodiments may also be implemented in software. 
     Blind Signal Separation 
       FIG. 13  illustrates signal extraction component  506  of  FIG. 5  in further detail, in accordance with another embodiment. As opposed to the earlier described echo cancellation like embodiments, signal extraction component  506  of  FIG. 13  implements a blind signal separation technique to remove the signal of the reference channel from the signal of the primary channel to extract desired audio. 
     As illustrated, signal extraction component  810  comprises a number of adaptive FIR filters  1302 , adders  1306  and a cost function  1304 . The elements are coupled to each other as shown. 
     Both the reference and the primary channels are filtered using adaptive FIR filters  1302 . The results are subtracted from each other using corresponding adders  1306 . The resulted signals are outputted, with the result of the reference signal having been subtracted from the primary signal being the desired audio. 
     The output signals are in turn feedback to a cost function, which outputs respective adaptation for adaptive FIR filters  1302  based on the two output signals. The cost function depends on specific BSS method. 
     CONCLUSION AND EPILOGUE 
     Thus, it can be seen from the above descriptions, various novel acoustic devices, systems and methods have been described. 
     While the present invention has been described in terms of the above described embodiments, those skilled in the art will recognize that the invention is not limited to the embodiments described. The present invention can be practiced with modification and alteration within the spirit and scope of the appended claims. Thus, the description is to be regarded as illustrative instead of restrictive on the present invention.