Patent Publication Number: US-2020294521-A1

Title: Microphone configurations for eyewear devices, systems, apparatuses, and methods

Description:
RELATED APPLICATIONS 
     This patent application is a continuation-in-part of U.S. Non-provisional patent application titled “HEAD WEARABLE ACOUSTIC SYSTEM WITH NOISE CANCELING MICROPHONE GEOMETRY APPARATUSES AND METHODS” filed on Oct. 18, 2015, Ser. No. 14/886,077, which is a continuation-in-part of U.S. Non-Provisional patent application titled “Dual Stage Noise Reduction Architecture For Desired Signal Extraction,” filed on Mar. 12, 2014, Ser. No. 14/207,163 which claims priority from U.S. Provisional Patent Application titled “Noise Canceling Microphone Apparatus,” filed on Mar. 13, 2013, Ser. No. 61/780,108 and from U.S. Provisional Patent Application titled “Systems and Methods for Processing Acoustic Signals,” filed on Feb. 18, 2014, Ser. No. 61/941,088. 
     Patent application Ser. No. 14/886,077 is also a continuation-in-part of U.S. Non-Provisional patent application titled “Eye Glasses With Microphone Array,” filed on Feb. 14, 2014, Ser. No. 14/180,994 which claims priority from U.S. Provisional Patent Application Ser. No. 61/780,108 filed on Mar. 13, 2013, and from U.S. Provisional Patent Application Ser. No. 61/839,211 filed on Jun. 25, 2013, and from U.S. Provisional Patent Application Ser. No. 61/839,227 filed on Jun. 25, 2013, and from U.S. Provisional Patent Application Ser. No. 61/912,844 filed on Dec. 6, 2013. 
     This patent application also claims priority to U.S. Provisional Patent Application titled “MICROPHONE CONFIGURATIONS FOR EYEWEAR DEVICES APPARATUSES AND METHODS” filed on Feb. 5, 2019 Ser. No. 62/801,618. 
     U.S. Provisional Patent Application Ser. No. 62/801,618 is hereby incorporated by reference. U.S. Provisional Patent Application Ser. No. 61/780,108 is hereby incorporated by reference. U.S. Provisional Patent Application Ser. No. 61/941,088 is hereby incorporated by reference. U.S. Non-Provisional patent application Ser. No. 14/207,163 is hereby incorporated by reference. U.S. Non-Provisional patent application Ser. No. 14/180,994 is hereby incorporated by reference. U.S. Provisional Patent Application Ser. No. 61/839,211 is hereby incorporated by reference. U.S. Provisional Patent Application Ser. No. 61/839,227 is hereby incorporated by reference. U.S. Provisional Patent Application Ser. No. 61/912,844 is hereby incorporated by reference. 
    
    
     BACKGROUND OF THE INVENTION 
     1. Field of Invention 
     The invention relates generally to wearable devices which detect and process acoustic signal data and more specifically to reducing noise in head wearable acoustic systems and to assist a user&#39;s hearing. 
     2. Art Background 
     Acoustic systems employ acoustic sensors such as microphones to receive audio signals. Often, these systems are used in real world environments which present desired audio and undesired audio (also referred to as noise) to a receiving microphone simultaneously. Such receiving microphones are part of a variety of systems such as a mobile phone, a handheld microphone, a hearing aid, etc. These systems often perform speech recognition processing on the received acoustic signals. Simultaneous reception of desired audio and undesired audio have a negative impact on the quality of the desired audio. Degradation of the quality of the desired audio can result in desired audio which is output to a user and is hard for the user to understand. Degraded desired audio used by an algorithm such as in speech recognition (SR) or Automatic Speech Recognition (ASR) can result in an increased error rate which can render the reconstructed speech hard to understand. Either of which presents a problem. 
     Handheld systems require a user&#39;s fingers to grip and/or operate the device in which the handheld system is implemented. Such as a mobile phone for example. Occupying a user&#39;s fingers can prevent the user from performing mission critical functions. This can present a problem. 
     Undesired audio (noise) can originate from a variety of sources, which are not the source of the desired audio. Thus, the sources of undesired audio are statistically uncorrelated with the desired audio. The sources can be of a non-stationary origin or from a stationary origin. Stationary applies to time and space where amplitude, frequency, and direction of an acoustic signal do not vary appreciably. For, example, in an automobile environment engine noise at constant speed is stationary as is road noise or wind noise, etc. In the case of a non-stationary signal, noise amplitude, frequency distribution, and direction of the acoustic signal vary as a function of time and or space. Non-stationary noise originates for example, from a car stereo, noise from a transient such as a bump, door opening or closing, conversation in the background such as chit chat in a back seat of a vehicle, etc. Stationary and non-stationary sources of undesired audio exist in office environments, concert halls, football stadiums, airplane cabins, everywhere that a user will go with an acoustic system (e.g., mobile phone, tablet computer etc. equipped with a microphone, a headset, an ear bud microphone, etc.) At times, the environment that the acoustic system is used in is reverberant, thereby causing the noise to reverberate within the environment, with multiple paths of undesired audio arriving at the microphone location. Either source of noise, i.e., non-stationary or stationary undesired audio, increases the error rate of speech recognition algorithms such as SR or ASR or can simply make it difficult for a system to output desired audio to a user which can be understood. All of this can present a problem. 
     Various noise cancellation approaches have been employed to reduce noise from stationary and non-stationary sources. Existing noise cancellation approaches work better in environments where the magnitude of the noise is less than the magnitude of the desired audio, e.g., in relatively low noise environments. Spectral subtraction is used to reduce noise in speech recognition algorithms and in various acoustic systems such as in hearing aids. Systems employing Spectral Subtraction do not produce acceptable error rates when used in Automatic Speech Recognition (ASR) applications when a magnitude of the undesired audio becomes large. This can present a problem. 
     In addition, existing algorithms, such as Spectral Subtraction, etc., employ non-linear treatment of an acoustic signal. Non-linear treatment of an acoustic signal results in an output that is not proportionally related to the input. Speech Recognition (SR) algorithms are developed using voice signals recorded in a quiet environment without noise. Thus, speech recognition algorithms (developed in a quiet environment without noise) produce a high error rate when non-linear distortion is introduced in the speech process through non-linear signal processing. Non-linear treatment of acoustic signals can result in non-linear distortion of the desired audio which disrupts feature extraction which is necessary for speech recognition, this results in a high error rate. All of which can present a problem. 
     Various methods have been used to try to suppress or remove undesired audio from acoustic systems, such as in Speech Recognition (SR) or Automatic Speech Recognition (ASR) applications for example. One approach is known as a Voice Activity Detector (VAD). A VAD attempts to detect when desired speech is present and when undesired speech is present. Thereby, only accepting desired speech and treating as noise by not transmitting the undesired speech. Traditional voice activity detection only works well for a single sound source or a stationary noise (undesired audio) whose magnitude is small relative to the magnitude of the desired audio. Therefore, traditional voice activity detection renders a VAD a poor performer in a noisy environment. Additionally, using a VAD to remove undesired audio does not work well when the desired audio and the undesired audio are arriving simultaneously at a receive microphone. This can present a problem. 
     Acoustic systems used in noisy environments with a single microphone present a problem in that desired audio and undesired audio are received simultaneously on a single channel. Undesired audio can make the desired audio unintelligible to either a human user or to an algorithm designed to use received speech such as a Speech Recognition (SR) or an Automatic Speech Recognition (ASR) algorithm. This can present a problem. Multiple channels have been employed to address the problem of the simultaneous reception of desired and undesired audio. Thus, on one channel, desired audio and undesired audio are received and on the other channel an acoustic signal is received which also contains undesired audio and desired audio. Over time the sensitivity of the individual channels can drift which results in the undesired audio becoming unbalanced between the channels. Drifting channel sensitivities can lead to inaccurate removal of undesired audio from desired audio. Non-linear distortion of the original desired audio signal can result from processing acoustic signals obtained from channels whose sensitivities drift over time. This can present a problem. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       The invention may best be understood by referring to the following description and accompanying drawings that are used to illustrate embodiments of the invention. The invention is illustrated by way of example in the embodiments and is not limited in the figures of the accompanying drawings, in which like references indicate similar elements. 
         FIG. 1  illustrates a general process for microphone configuration on a head wearable device according to embodiments of the invention. 
         FIG. 2  illustrates microphone placement geometry according to embodiments of the invention. 
         FIG. 3A  illustrates generalized microphone placement with a primary microphone at a first location according to embodiments of the invention. 
         FIG. 3B  illustrates signal-to-noise ratio difference measurements for main microphone as located in  FIG. 3A , according to embodiments of the invention. 
         FIG. 3C  illustrates signal-to-noise ratio difference versus increasing microphone acoustic separation distance for the data shown in  FIG. 3B  according to embodiments of the invention. 
         FIG. 4A  illustrates generalized microphone placement with a primary microphone at a second location according to embodiments of the invention. 
         FIG. 4B  illustrates signal-to-noise ratio difference measurements for main microphone as located in  FIG. 4A , according to embodiments of the invention. 
         FIG. 4C  illustrates signal-to-noise ratio difference versus increasing microphone acoustic separation distance for the data shown in  FIG. 4B  according to embodiments of the invention. 
         FIG. 5A  illustrates generalized microphone placement with a primary microphone at a third location according to embodiments of the invention. 
         FIG. 5B  illustrates signal-to-noise ratio difference measurements for main microphone as located in  FIG. 5A , according to embodiments of the invention. 
         FIG. 5C  illustrates signal-to-noise ratio difference versus increasing microphone acoustic separation distance for the data shown in  FIG. 5B  according to embodiments of the invention. 
         FIG. 6  illustrates microphone directivity patterns according to embodiments of the invention. 
         FIG. 7  illustrates a misaligned reference microphone response axis according to embodiments of the invention. 
         FIG. 8  is a diagram illustrating an embodiment of eyeglasses of the invention having two embedded microphones. 
         FIG. 9  is a diagram illustrating an embodiment of eyeglasses of the invention having three embedded microphones. 
         FIG. 10  is an illustration of another embodiment of the invention employing four omni directional microphones at four acoustic ports in place of two bidirectional microphones. 
         FIG. 11  is a schematic representation of eyewear of the invention employing two omni directional microphones placed diagonally across the lens opening defined by the front frame of the eyewear. 
         FIG. 12  is an illustration of another embodiment of the invention employing four omni directional microphones placed along the top and bottom portions of the eyeglasses frame. 
         FIG. 13  is an illustration of another embodiment of the invention wherein microphones have been placed at a temple portion of the eyewear facing inward and at a lower center corner of the front frame of the eyewear and facing down. 
         FIG. 14  is an illustration of another embodiment of the invention wherein microphones have been placed at a temple portion of the eyewear facing inward and at a lower center corner of the front frame of the eyewear and facing down. 
         FIG. 15  illustrates an eye glass with built-in acoustic noise cancellation system according to embodiments of the invention. 
         FIG. 16  illustrates a primary microphone location in the head wearable device from  FIG. 15  according to embodiments of the invention. 
         FIG. 17  illustrates goggles with built-in acoustic noise cancellation system according to embodiments of the invention. 
         FIG. 18  illustrates a visor with built-in acoustic noise cancellation system according to embodiments of the invention. 
         FIG. 19  illustrates a helmet with built-in acoustic noise cancellation system according to embodiments of the invention. 
         FIG. 20  illustrates a process for extracting a desired audio signal according to embodiments of the invention. 
         FIG. 21  illustrates system architecture, according to embodiments of the invention. 
         FIG. 22  illustrates filter control, according to embodiments of the invention. 
         FIG. 23  illustrates another diagram of system architecture, according to embodiments of the invention. 
         FIG. 24A  illustrates another diagram of system architecture incorporating auto-balancing, according to embodiments of the invention. 
         FIG. 24B  illustrates processes for noise reduction, according to embodiments of the invention. 
         FIG. 25A  illustrates beamforming according to embodiments of the invention. 
         FIG. 25B  presents another illustration of beamforming according to embodiments of the invention. 
         FIG. 25C  illustrates beamforming with shared acoustic elements according to embodiments of the invention. 
         FIG. 26  illustrates multi-channel adaptive filtering according to embodiments of the invention. 
         FIG. 27  illustrates single channel filtering according to embodiments of the invention. 
         FIG. 28A  illustrates desired voice activity detection according to embodiments of the invention. 
         FIG. 28B  illustrates a normalized voice threshold comparator according to embodiments of the invention. 
         FIG. 28C  illustrates desired voice activity detection utilizing multiple reference channels, according to embodiments of the invention. 
         FIG. 28D  illustrates a process utilizing compression according to embodiments of the invention. 
         FIG. 28E  illustrates different functions to provide compression according to embodiments of the invention. 
         FIG. 29A  illustrates an auto-balancing architecture according to embodiments of the invention. 
         FIG. 29B  illustrates auto-balancing according to embodiments of the invention. 
         FIG. 29C  illustrates filtering according to embodiments of the invention. 
         FIG. 30  illustrates a process for auto-balancing according to embodiments of the invention. 
         FIG. 31  illustrates an acoustic signal processing system according to embodiments of the invention. 
         FIG. 32A  illustrates microphone configurations on a head wearable device in perspective view according to embodiments of the invention. 
         FIG. 32B  illustrates microphone configurations on a head wearable device in top view corresponding to  FIG. 32A  according to embodiments of the invention. 
         FIG. 32C  illustrates microphone configurations on a head wearable device in bottom view corresponding to  FIG. 32A  according to embodiments of the invention. 
         FIG. 32D  illustrates another set of microphone placements in perspective view on a head wearable device according to embodiments of the invention. 
         FIG. 32E  illustrates microphone placement on a head wearable device in bottom view corresponding to  FIG. 32D  according to embodiments of the invention. 
         FIG. 33  illustrates the head wearable devices from  FIG. 32A-D  relative to different sound sources according to embodiments of the invention. 
         FIG. 34  illustrates processing acoustic signals from an array of microphones configured with a head wearable device, according to embodiments of the invention. 
     
    
    
     DETAILED DESCRIPTION 
     In the following detailed description of embodiments of the invention, reference is made to the accompanying drawings in which like references indicate similar elements, and in which is shown by way of illustration, specific embodiments in which the invention may be practiced. These embodiments are described in sufficient detail to enable those of skill in the art to practice the invention. In other instances, well-known circuits, structures, and techniques have not been shown in detail in order not to obscure the understanding of this description. The following detailed description is, therefore, not to be taken in a limiting sense, and the scope of the invention is defined only by the appended claims. 
     Apparatuses and methods are described for detecting and processing acoustic signals containing both desired audio and undesired audio within a head wearable device. In one or more embodiments, noise cancellation architectures combine multi-channel noise cancellation and single channel noise cancellation to extract desired audio from undesired audio. In one or more embodiments, multi-channel acoustic signal compression is used for desired voice activity detection. In one or more embodiments, acoustic channels are auto-balanced. In one or more embodiments, a system automatically selects a subset of microphones for acoustic signal extraction from an array of possible microphones. In one or more embodiments, a user is provided with hearing assistance to facilitate hearing sounds from a local environment. 
       FIG. 1  illustrates a general process at  100  for microphone configuration on a head wearable device according to embodiments of the invention. With reference to  FIG. 1 , a process starts at a block  102 . At a block  104 , a “main” or “primary” microphone channel is created on a head wearable device using one or more microphones. The main microphone(s) is positioned to optimize reception of desired audio thereby enhancing a first signal-to-noise ratio associated with the main microphone, indicated as SNR M . At a block  106 , a reference microphone channel is created on the head wearable device using one or more microphones. The reference microphone(s) is positioned on the head wearable device to provide a lower signal-to-noise ratio with respect to detection of desired audio from the user, thereby resulting in a second signal-to-noise ratio indicated as SNR R . Thus, at a block  108  a signal-to-noise ratio difference is accomplished by placement geometry of the microphones on the head wearable device, resulting in the first signal-to-noise ratio SNR M  being greater than the second signal-to-noise ratio SNR R . 
     At a block  110  a signal-to-noise ratio difference is accomplished through beamforming by creating different response patterns (directivity patterns) for the main microphone channel and the reference microphone channel(s). Utilizing different directivity patterns to create a signal-to-noise ratio difference is described more fully below in conjunction with the figures that follow. 
     In various embodiments, at a block  112  a signal-to-noise ratio difference is accomplished through a combination of one or more of microphone placement geometry, beamforming, and utilizing different directivity patterns for the main and reference channels. At a block  114  the process ends. 
       FIG. 2  illustrates, generally at  200 , microphone placement geometry according to embodiments of the invention. With reference to  FIG. 200 , a source of desired audio, a user&#39;s mouth is indicated at  202 , from which desired audio  204  emanates. The source  202  provides desired audio  204  to the microphones mounted on a head wearable device. A first microphone  206  is positioned at a distance indicated by d 1    208  from the source  202 . A second microphone  210  is positioned at a distance indicated by d 2    212  from the source  202 . The system of  200  is also exposed to undesired audio as indicated by  218 . 
     With respect to the source  202 , the first microphone  206  and the second microphone  210  are at different acoustic distances from the source  202  as represented by ΔL at  214 . The difference in acoustic distances ΔL  214  is given by equation  216 . As used in this description of embodiments, the distances d 1  and d 2  represent the paths that the acoustic wave travels to reach the respective microphones  206  and  210 . Thus, these distances might be linear or they might be curved depending on the particular location of a microphone on a head wearable device and the acoustic frequency of interest. For clarity in illustration, these paths and the corresponding distances have been indicated with straight lines however, no limitation is implied thereby. 
     Undesired audio  218  typically results from various sources that are located at distances that are much greater than the distances dr and d 2 . For example, construction noise, car noise, airplane noise, etc. all originate at distances that are typically several orders of magnitude larger than d 1  and d 2 . Thus, undesired audio  218  is substantially correlated at microphone locations  206  and  210  or is at least received at a fairly uniform level at each location. The difference in acoustic distance ΔL at  214  decreases an amplitude of the desired audio  204  received at the second microphone  210  relative to the first microphone  208 , due to various mechanisms. One such mechanism is, for example, spherical spreading which causes the desired audio signal to fall off as a function of 1/r 2 , where r is the distance (e.g.  208  or  212 ) between a source (e.g.,  202 ) and a receive location (e.g.,  206  or  210 ). Reduction in desired audio at the second microphone location  210  decreases a signal-to-noise ratio at  210  relative to  206  since the noise amplitude is substantially the same at each location but the signal amplitude is decreased at  210  relative to the amplitude received at  206 . Another related mechanism to path length is a difference in an acoustic impendence along one path versus another, thereby resulting in a curved acoustic path instead of a straight path. Collectively, the mechanisms combine to decrease an amplitude of desired audio received at a reference microphone location relative to a main microphone location. Thus, placement geometry is used to provide a signal-to-noise ratio difference between two microphone locations which is used by the noise cancellation system, which is described further below, to reduce undesired audio from the main microphone channel. 
     Microphone placement geometry admits various configurations for placement of a primary microphone and a reference microphone. In various embodiments, a general microphone placement methodology is described and presented in conjunction with  FIG. 3A  through  FIG. 5C  immediately below which permit microphones to be placed in various locations on a headwear device. 
       FIG. 3A  illustrates, generally at  300 , generalized microphone placement with a primary microphone at a first location according to embodiments of the invention. With reference to  FIG. 3A , a head wearable device  302  is illustrated. As used in this detailed description of embodiments a head wearable device can be any of the devices that are configured to wear on a user&#39;s head such as but not limited to glasses, goggles, a helmet, a visor, a head band, etc. In the discussion presented in conjunction with FIG.  3 A through  FIG. 5C  immediately below it is recognized that this discussion is equally applicable to any head wear device, such as those shown in  FIG. 8  through  FIG. 19  as well as to those head wearable devices not specifically shown in the figures herein. Thus, embodiments of the invention are applicable to head wearable devices that are as of yet unnamed or yet to be invented. 
     Referring back to  FIG. 3A , in one embodiment, the head wearable device has a frame  302  with attached temple  304  and temple  306 , a glass  308 , and a glass  310 . In various embodiments, the head wearable device  302  is a pair of glasses that are worn on a user&#39;s head. A number of microphones are located on the head wearable device  302 , such as a microphone  1 , a microphone  2 , a microphone  3 , a microphone  4 , a microphone  5 , a microphone  6 , a microphone  7 , a microphone  8 , and optionally a microphone  9  and a microphone  10 . In various embodiments, the head wearable device including frame  302 /temples  304  and  306  as illustrated, can be sized to include electronics  318  for signal processing as described further below. Electronics  318  provides electrical coupling to the microphones mounted on the head wearable device  302 . 
     The head wearable device  302  has an internal volume, defined by its structure, within which electronics  318  can be mounted. Alternatively electronics  318  can be mounted externally to the structure. In one or more embodiments, an access panel is provided to access the electronics  318 . In other embodiments no access door is provided explicitly but the electronics  318  can be contained within the volume of the head wearable device  302 . In such cases, the electronics  318  can be inserted prior to assembly of a head wearable device where one or more parts interlock together thereby forming a housing which captures the electronics  318  therein. In yet other embodiments, a head wearable device is molded around electronics  318  thereby encapsulating the electronics  318  within the volume of the head wearable device  302 . In various non-limiting embodiments, electronics  318  include an adaptive noise cancellation unit, a single channel noise cancellation unit, a filter control, a power supply, a desired voice activity detector, a filter, etc. Other components of electronics  118  are described below in the figures that follow. 
     The head wearable device  302  can include a switch (not shown) which is used to power up or down the head wearable device  302 . The head wearable device  302  can contain a data processing system within its volume for processing acoustic signals which are received by the microphones associated therewith. The data processing system can contain one or more of the elements of the system illustrated in  FIG. 31  described further below. Thus, the illustrations of  FIG. 3A  through  FIG. 5C  do not limit embodiments of the invention. 
     The headwear device of  FIG. 3A  illustrates that microphones can be placed in any location on the device. The ten locations chosen for illustration within the figures are selected merely for illustration of the general principles of placement geometry and do not limit embodiments of the invention. Accordingly, microphones can be used in different locations other than those illustrated and different microphones can be used in the various locations. For the purpose of illustration and without any limitation, the measurements that were made in conjunction with the illustrations of  FIG. 3A  through  FIG. 5C  omni-directional microphones were used. In other embodiments, directive microphones are used. In the example configuration used for the signal-to-noise ratio measurements, each microphone was mounted within a housing and each housing had a port opening to the environment. A direction for a port associated with microphone  1  is shown by arrow  1   b . A direction for a port associated with microphone  2  is shown by arrow  2   b . A direction for a port associated with microphone  3  is shown by arrow  3   b . A direction for a port associated with microphone  4  is shown by arrow  4   b . A direction for a port associated with microphone  5  is shown by arrow  5   b . A direction for a port associated with microphone  6  is shown by arrow  6   b . A direction for a port associated with microphone  7  is shown by arrow  7   b . A direction for a port associated with microphone  8  is shown by arrow  8   b.    
     A user&#39;s mouth is illustrated at  312  and is analogous to the source of desired audio shown in  FIG. 2  at  202 . An acoustic path length (referred to herein as acoustic distance or distance) from the user&#39;s mouth  312  to each microphone is illustrated with an arrow from the user&#39;s mouth  312  to the respective microphone locations. For example, d 1  indicates the acoustic distance from the user&#39;s mouth  312  to microphone  1 . d 2  indicates the acoustic distance from the user&#39;s mouth  312  to microphone  2 . d 3  indicates the acoustic distance from the user&#39;s mouth  312  to microphone  3 . d 4  indicates the acoustic distance from the user&#39;s mouth  312  to microphone  4 . d 5  indicates the acoustic distance from the user&#39;s mouth  312  to microphone  5 . d 6  indicates the acoustic distance from the user&#39;s mouth  312  to microphone  6 . d 7  indicates the acoustic distance from the user&#39;s mouth  312  to microphone  7 . d 8  indicates the acoustic distance from the user&#39;s mouth  312  to microphone  8 . Similarly, optional microphone  9  and microphone  10  have acoustic distances as well; however they are not so labeled to preserve clarity in the figure. 
     In  FIG. 3A , microphones  1 ,  2 ,  3 , and  6  and the user&#39;s mouth  312  fall substantially in an X-Z plane (see coordinate system  316 ), the corresponding acoustic distances d 1 , d 2 , d 3 , and d 6  have been indicated with substantially straight lines. The paths to microphones  4 ,  5 ,  7 , and  8 , i.e., d 4 , d 5 , d 7 , and d 8  are represented as curved paths which reflect the fact that the user&#39;s head is not transparent to the acoustic field. Thus, in such cases, the acoustic path is somewhat curved. In general, the acoustic path between the source of desired audio and a microphone on the head wearable device can be linear or curved. As long as the path length is sufficiently different between a main microphone and a reference microphone the requisite signal-to-noise ratio difference will be obtained which is needed by the noise cancellation system in order to achieve an acceptable level of noise cancellation. 
     To make the measurements presented in  FIG. 3B  and  FIG. 3C , an acoustic test facility was used to measure signal-to-noise ratio difference between primary and reference microphone locations. The test facility included a manikin with a built-in speaker was used to simulate a user wearing a head wearable device. A speaker positioned at a location of the user&#39;s mouth was used to produce the desired audio signal. The manikin was placed inside of an anechoic chamber of the acoustic test facility. Background noise was generated within the anechoic chamber with an array of speakers. A pink noise spectrum was used during the measurements; however, other weightings in frequency can be used for the background noise field. During these measurements, the spectral amplitude level of the background noise was set to 75 dB/uPa/Hz. A head wearable device was placed on the manikin. During the test, microphones were located at the positions shown in  FIG. 3A  on the head wearable device. A microphone for a main or primary channel is selected as microphone  1  for the first sequence of measurements which are illustrated in  FIG. 3B  and  FIG. 3C  directly below. 
     The desired audio signal consisted of the word “Camera.” This word was transmitted through the speaker in the manikin. The received signal corresponding to the word “Camera” at microphone  1  was processed through the noise cancellation system (as described below in the figures that follow), gated in time, and averaged to produce the “signal” amplitude corresponding with microphone  1 . The corresponding signal corresponding to the word “Camera” was measured in turn at each of the other microphones at locations  2 ,  3 ,  4 ,  5 ,  6 ,  7 , and  8 . Similarly, at each microphone location, background noise spectral levels were measured. With these measurements, signal-to-noise ratios were computed at each microphone location and then signal-to-noise ratio difference was computed for microphone pairs as shown in the figures directly below. 
       FIG. 3B  illustrates, generally at  320 , signal-to-noise ratio difference measurements for a main microphone as located in  FIG. 3A , according to embodiments of the invention. With reference to  FIG. 3B  and  FIG. 3A , microphone  1  is used as the main or primary microphone at  314 . A variety of locations were then used to place the reference microphone, such as microphone  2 , microphone  3 , microphone  6 , microphone  4 , microphone  5 , microphone  7 , and microphone  8 . In  FIG. 3B , column  322  indicates the microphone pair used for a set of measurements. A column  324  indicates the approximate difference in acoustic path length between the given microphone pair of column  322 . Approximate acoustic path length difference ˜ΔL is given by equation  216  in  FIG. 2 . Column  326  lists a non-dimensional number ranging from 1 to 7 for the seven different microphone pairs used for signal-to-noise ratio measurements. A column  328  lists the signal-to-noise ratio difference for the given microphone pair listed in the column  322 . Each row,  330 ,  332 ,  334 ,  336 ,  338 ,  340 , and  342  lists a different microphone pair, where the reference microphone has changed while the main microphone  314  is held constant as microphone  1 . Note that the approximate difference in acoustic path lengths for the various microphone pairs can be arranged in increasing order as shown by equation  344 . The microphone pairs have been arranged in the rows  330 - 342  in increasing approximate acoustic path length difference  324  according to equation  344 . Signal-to-noise ratio difference varies from 5.55 dB for microphone  2  used as a reference microphone to 10.48 dB when microphone  8  is used as the reference microphone. 
       FIG. 3C  illustrates, generally at  350 , signal-to-noise ratio difference versus increasing microphone acoustic separation distance for the data shown in  FIG. 3B  according to embodiments of the invention. With reference to  FIG. 3C , signal-to-noise ratio difference is plotted on a vertical axis at  352  and the non-dimensional X value from column  326  ( FIG. 3B ) is plotted on the horizontal axis at  354 . Note, as described above, the non-dimensional X value is representative of approximate acoustic path length difference ˜ΔL. The X axis  354  does not correspond exactly with ˜ΔL, but it is related to ˜ΔL because the data have been arranged and plotted in increasing approximate acoustic path length difference ˜ΔL. Such ordering of the data helps to illustrate the character of signal-to-noise ratio difference described above in conjunction with  FIG. 2 , i.e., signal-to-noise ratio difference will increase with increasing acoustic path length difference between main and reference microphones. This behavior is discerned by observing that signal-to-noise ratio difference is increasing as a function of ˜ΔL, with a curve  356  which plots data from columns  328  as a function of the data from column  326  ( FIG. 3B ). 
       FIG. 4A  illustrates, generally at  420  generalized microphone placements with a primary microphone at a second location according to embodiments of the invention. In  FIG. 4A , the second location for the main microphone  414  is the location occupied by microphone  2 . The tests described above were repeated with microphone  2  as the main microphone and the reference microphone locations were alternatively those of microphone  6 , microphone  3 , microphone  4 , microphone  5 , microphone  7 , and microphone  8 . These data are described below in conjunction with  FIG. 48  and  FIG. 4C . 
       FIG. 4B  illustrates signal-to-noise ratio difference measurements for main microphone as located in  FIG. 4A , according to embodiments of the invention. With reference to  FIG. 4B  and  FIG. 4A , microphone  2  is used as the main or primary microphone  414 . A variety of locations were then used to place the reference microphone, such as microphone  6 , microphone  3 , microphone  4 , microphone  5 , microphone  7 , and microphone  8 . In  FIG. 4B , column  422  indicates the microphone pair used for a set of measurements. A column  424  indicates the approximate difference in acoustic path length between the given microphone pair of column  422 . Approximate acoustic path length difference ˜ΔL is given by equation  216  in  FIG. 2 . Column  426  lists a non-dimensional number ranging from 1 to 6 for the six different microphone pairs used for signal-to-noise ratio measurements. A column  428  lists the signal-to-noise ratio difference for the given microphone pair listed in the column  422 . Each row,  430 ,  432 ,  434 ,  336 ,  438 , and  440  lists a different microphone pair, where the reference microphone has changed while the main microphone  414  is held constant as microphone  2 . Note that the approximate difference in acoustic path lengths for the various microphone pairs can be arranged in increasing order as shown by equation  442 . The microphone pairs have been arranged in the rows  430 - 440  in increasing approximate acoustic path length difference  424  according to equation  442 . Signal-to-noise ratio difference varies from 1.2 dB for microphone  6  used as a reference microphone to 5.2 dB when microphone  8  is used as the reference microphone. 
       FIG. 4C  illustrates signal-to-noise ratio difference versus increasing microphone acoustic separation distance for the data shown in  FIG. 4B  according to embodiments of the invention. With reference to  FIG. 4C , signal-to-noise ratio difference is plotted on a vertical axis at  452  and the non-dimensional X value from column  426  ( FIG. 48 ) is plotted on the horizontal axis at  454 . Note, as described above, the non-dimensional X value is representative of approximate acoustic path length difference ˜ΔL. The X axis  454  does not correspond exactly with ˜ΔL, but it is related to ˜ΔL because the data have been arranged and plotted in increasing approximate acoustic path length difference ˜ΔL. Such ordering of the data helps to illustrate the character of signal-to-noise ratio difference described above in conjunction with  FIG. 2 , i.e., signal-to-noise ratio difference will increase with increasing acoustic path length difference between main and reference microphones. This behavior is discerned by observing that signal-to-noise ration difference is increasing as a function of ˜ΔL, with a curve  456 , which plots data from columns  428  as a function of the data from column  426  ( FIG. 4B ). 
       FIG. 5A  illustrates generalized microphone placement with a primary microphone at a third location according to embodiments of the invention. In  FIG. 5A , the third location for the main microphone  514  is the location occupied by microphone  3 . The tests described above were repeated with microphone  3  as the main microphone and the reference microphone locations were alternatively those of microphone  6 , microphone  4 , microphone  5 , microphone  7 , and microphone  8 . These data are described below in conjunction with  FIG. 5B  and  FIG. 5C . 
       FIG. 5B  illustrates signal-to-noise ratio difference measurements for main microphone as located in  FIG. 5A , according to embodiments of the invention. With reference to  FIG. 5B  and  FIG. 5A , microphone  3  is used as the main or primary microphone  514 . A variety of locations were then used to place the reference microphone, such as microphone  6 , microphone  4 , microphone  5 , microphone  7 , and microphone  8 . In  FIG. 5B , column  522  indicates the microphone pair used for a set of measurements. A column  524  indicates the approximate difference in acoustic path length between the given microphone pair of column  522 . Approximate acoustic path length difference ˜ΔL is given by equation  216  in  FIG. 2 . Column  526  lists a non-dimensional number ranging from 1 to 5 for the five different microphone pairs used for signal-to-noise ratio measurements. A column  528  lists the signal-to-noise ratio difference for the given microphone pair listed in the column  522 . Each row,  530 ,  532 ,  534 ,  536 , and  538  lists a different microphone pair, where the reference microphone has changed while the main microphone  514  is held constant as microphone  3 . Note that the approximate difference in acoustic path lengths for the various microphone pairs can be arranged in increasing order as shown by equation  540 . The microphone pairs have been arranged in the rows  530 - 538  in increasing approximate acoustic path length difference  524  according to equation  540 . Signal-to-noise ratio difference varies from 0 dB for microphone  6  used as a reference microphone to 5.16 dB when microphone  7  is used as the reference microphone. 
       FIG. 5C  illustrates signal-to-noise ratio difference versus increasing microphone acoustic separation distance for the data shown in  FIG. 5B  according to embodiments of the invention. With reference to  FIG. 5C , signal-to-noise ratio difference is plotted on a vertical axis at  552  and the non-dimensional X value from column  526  ( FIG. 58 ) is plotted on the horizontal axis at  554 . Note, as described above, the non-dimensional X value is representative of approximate acoustic path length difference ˜ΔL. The X axis  554  does not correspond exactly with ˜ΔL, but it is related to ˜ΔL because the data have been arranged and plotted in increasing approximate acoustic path length difference ˜ΔL. Such ordering of the data helps to illustrate the character of signal-to-noise ratio difference described above in conjunction with  FIG. 2 , i.e., signal-to-noise ratio difference will increase with increasing acoustic path length difference between main and reference microphones. This behavior is discerned by observing that signal-to-noise ratio difference is increasing as a function of ˜ΔL, with a curve  556 , which plots data from columns  528  as a function of the data from column  526  ( FIG. 5B ). 
     Note that within the views presented in the figures above, specific locations for the microphones have been chosen for the purpose of illustration only. These locations do not limit embodiments of the invention. Other locations for microphones on a head wearable device are used in other embodiments. 
     Thus, as described above in conjunction with  FIG. 1  block  108  and  FIG. 2  through  FIG. 5C , in various embodiments, microphone placement geometry is used to create an acoustic path length difference between two microphones and a corresponding signal-to-noise ratio difference between a main and a reference microphone. The signal-to-noise ratio difference can also be accomplished through the use of different directivity patterns for the main and reference microphones. In some embodiments beamforming is used to create different directivity patterns for a main and a reference channel. For example, in  FIG. 5A , acoustic path lengths d 3  and d 6  are too similar in value, thus this choice of locations for the main and reference microphones did not produce an adequate signal-to-noise ratio difference (0 dB at column  528  row  530   FIG. 5B ). In such a case, variation in microphone directivity pattern (one or both microphones) and/or beamforming can be used to create the needed signal-to-noise ratio difference between the main and the reference channels. 
     A directional microphone can be used to decrease reception of desired audio and/or to increase reception of undesired audio, thereby lowering a signal-to-noise ratio of a second microphone (reference microphone), which results in an increase in the signal-to-noise ratio difference between the primary and reference microphones. An example is illustrated in  FIG. 3A  using a second microphone (not shown) and the techniques taught in  FIG. 6  and  FIG. 7  below. In some embodiments, the second microphone can be substantially co-located with microphone  1 . In other embodiments, the second microphone is located an equivalent distance from the source  312  as is the first microphone. In some embodiments, the second microphone is a directional microphone whose main response axis is substantially perpendicular to (or equivalently stated misaligned with) the acoustic path d 1 . Thus, a null or a direction of lesser response to desired audio from  312  for the second microphone exists in the direction of desired audio d 1 . This results in a decrease in the signal-to-noise ratio of the second microphone and an increase in a signal-to-noise ratio difference calculated between the first microphone and the second microphone. Note that the two microphones can be placed in any location on the head wearable device  302 , which includes co-location as described above. In other embodiments, one or more microphone elements are used as inputs to a beamformer resulting in main and reference channels having different directivity patterns and a resulting signal-to-noise ratio difference there between. 
       FIG. 6  illustrates, generally at  600 , microphone directivity patterns according to embodiments of the invention. With reference to  FIG. 6 , an omni-directional microphone directivity pattern is illustrated with circle  602  having constant radius  604  indicating uniform sensitivity as a function of angle alpha (a) at  608  measured from reference  606 . 
     An example of a directional microphone having a cardioid directivity pattern  622  is illustrated within plot  620  where the cardioid directivity pattern  622  has a peak sensitivity axis indicated at  624  and a null indicated at  626 . A cardioid directivity pattern can be formed with two omni-directional microphones or with an omni-directional microphone and a suitable mounting structure for the microphone. 
     An example of a directional microphone having a bidirectional directivity pattern  642 / 644  is illustrated within plot  640  where a first lobe  642  of the bidirectional directivity pattern has a first peak sensitivity axis indicated at  648  the second lobe  644  has a second peak sensitivity axis indicated at  646 . A first null exists at a direction  650  and a second null exists at a direction  652 . 
     An example of a directional microphone having a super-cardioid directivity pattern is illustrated with plot  660  where the super-cardioid directivity pattern  664 / 665  has a peak sensitivity axis indicated at a direction  662 , a minor sensitivity axis indicated at a direction  666  and nulls indicated at directions  668  and  670 . 
       FIG. 7  illustrates, generally at  700 , a misaligned reference microphone response axis according to embodiments of the invention. With reference to  FIG. 7 , a microphone is indicated at  702 . The microphone  702  is a directional microphone having a main response axis  706  and a null in its directivity pattern indicated at  704 . An incident acoustic field is indicated arriving from a direction  708 . In various embodiments, the microphone  702  is for example a bidirectional microphone as illustrated in  FIG. 6  above. Suitably positioned on a head wearable device, the directional microphone  702  decreases a signal-to-noise ratio when used as a reference microphone by limiting response to desired audio coming from direction  708  while responding to undesired audio, coming from a direction  710 . The response of the directive microphone  702  will produce an increase in a signal-to-noise ratio difference as described above. 
     Thus, within the teachings of embodiments presented herein one or more main microphones and one or more reference microphones are placed in locations on a head wearable device to obtain suitable signal-to-noise ratio difference between a main and a reference microphone. Such signal-to-noise ratio difference enables extraction of desired audio from an acoustic signal containing both desired audio and undesired audio as described below in conjunction with the figures that follow. Microphones can be placed at various locations on the head wearable device, including co-locating a main and a reference microphone at a common position on a head wearable device. 
     In some embodiments, the techniques of microphone placement geometry are combined together with different directivity patterns obtained at the microphone level or through beamforming to produce a signal-to-noise ratio difference between a main and a reference channel according to a block  112  ( FIG. 1 ). 
     In various embodiments, a head wearable device is an eyewear device as described below in conjunction with the figures that follow.  FIG. 8  is an illustration of an example of one embodiment of an eyewear device  800  of the invention. As shown therein, eyewear device  800  includes eyeglasses  802  having embedded microphones. The eyeglasses  802  have two microphones  804  and  806 . First microphone  804  is arranged in the middle of the eyeglasses  802  frame. Second microphone  806  is arranged on the side of the eyeglasses  802  frame. The microphones  804  and  806  can be pressure-gradient microphone elements, either bi- or uni-directional. In one or more embodiments, each microphone  804  and  806  is a microphone assembly within a rubber boot. The rubber boot provides an acoustic port on the front and the back side of the microphone with acoustic ducts. The two microphones  804  and  806  and their respective boots can be identical. The microphones  804  and  806  can be sealed air-tight (e.g., hermetically sealed). The acoustic ducts are filled with windscreen material. The ports are sealed with woven fabric layers. The lower and upper acoustic ports are sealed with a water-proof membrane. The microphones can be built into the structure of the eyeglasses frame. Each microphone has top and bottom holes, being acoustic ports. In an embodiment, the two microphones  804  and  806 , which can be pressure-gradient microphone elements, can each be replaced by two omni-directional microphones. 
       FIG. 9  is an illustration of another example of an embodiment of the invention. As shown in  FIG. 9 , eyewear device  900  includes eyeglasses  952  having three embedded microphones. The eyeglasses  952  of  FIG. 9  are similar to the eyeglasses  802  of  FIG. 8 , but instead employ three microphones instead of two. The eyeglasses  952  of  FIG. 9  have a first microphone  954  arranged in the middle of the eyeglasses  952 , a second microphone  956  arranged on the left side of the eyeglasses  952 , and a third microphone  958  arranged on the right side of the eyeglasses  952 . The three microphones can be employed in the three-microphone embodiment described above. 
       FIG. 10  is an illustration of an embodiment of eyewear  1000  of the present invention that replaces the two bi-directional microphones shown in  FIG. 8 , for example, with four omni-directional microphones  1002 ,  1004 ,  1006 ,  1008 , and electronic beam steering. Replacing the two bi-directional microphones with four omni-directional microphones provides eyewear frame designers more flexibility and manufacturability. In example embodiments having four omni-directional microphones, the four omni-directional microphones can be located anywhere on the eyewear frame, preferably with the pairs of microphones lining up vertically about a lens. In this embodiment, omni-directional microphones  1002  and  1004  are main microphones for detecting the primary sound that is to be separated from interference, and microphones  1004 ,  1008  are reference microphones that detect background noise that is to be separated from the primary sound. The array of microphones can be omni directional microphones, wherein the omni-directional microphones can be any combination of the following: electret condenser microphones, analog microelectromechanical systems (MEMS) microphones, or digital MEMS microphones. 
     Another example embodiment of the present invention, shown in  FIG. 11 , includes an eyewear device with a noise canceling microphone array, the eyewear device including an eyeglasses frame  1100 , an array of microphones coupled to the eyeglasses frame, the array of microphones including at least a first microphone  1102  and a second microphone  1104 , the first microphone coupled to the eyeglasses frame about a temple region, the temple region can be located approximately between a top corner of a lens opening and a support arm, and providing a first audio channel output, and the second microphone coupled to the eyeglasses frame about an inner lower corner of the lens opening, and providing a second audio channel output. The second microphone is located diagonally across lens opening  1106 , although it can be positioned anywhere along the inner frame of the lens, for example the lower corner, upper corner, or inner frame edge. Further, the second microphone can be along the inner edge of the lens at either the left or right of the nose bridge. 
     In yet another embodiment of the invention, the array of microphones can be coupled to the eyeglasses frame using at least one flexible printed circuit board (PCB) strip, as shown in  FIG. 12 . In this embodiment, eyewear device of the invention  1200  includes upper flexible PCB strip  1202  including the first  1204  and fourth  1206  microphones and a lower flexible PCB strip  1208  including the second  1210  and third  1212  microphones. 
     In further example embodiments, the eyeglasses frame can further include an array of vents corresponding to the array of microphones. The array of microphones can be bottom port or top port microelectromechanical systems (M EMS) microphones. As can be seen in  FIG. 13 , which is a microphone component of the eyewear of  FIG. 12 , MEMS microphone component  1300  includes MEMS microphone  1302  is affixed to flexible printed circuit board (PCB)  1304 . Gasket  1306  separates flexible PCB  1304  from device case  1308 . Vent  1310  is defined by flexible PCB  1304 , gasket  1306  and device case  1308 . Vent  1310  is an audio canal to channel audio waves to MEMS microphone  1302 . The first and fourth MEMS microphones can be coupled to the upper flexible PCB strip, the second and third MEMS microphones can be coupled to the lower flexible PCB strip, and the array of MEMS microphones can be arranged such that the bottom ports or top ports receive acoustic signals through the corresponding vents. 
       FIG. 14  shows another alternate embodiment of eyewear  1400  where microphones  1402 ,  1404  are placed at the temple region  1406  and front frame  1408 , respectively. 
       FIG. 15  illustrates, generally at  1500 , an eye glass with built-in acoustic noise cancellation system according to embodiments of the invention. With reference to  FIG. 15 , a head wearable device  1502  includes one or more microphones used for a main acoustic channel and one or more microphones used for a reference acoustic channel. The head wearable device  1502  is configured as a wearable computer with information display  1504 . In various embodiments, electronics are included at  1506  and/or at  1508 . In various embodiments, electronics can include noise cancellation electronics which are described more fully below in conjunction with the figures that follow. In other embodiments, noise cancellation electronics are not co-located with the head wearable device  1502  but are located externally from the head wearable device  1502 . In such embodiments, a wireless communication link such as is compatible with the Bluetooth® protocol, ZigBee®, etc. is provided to send the acoustic signals received from the microphones to an external location for processing by noise cancellation electronics. 
       FIG. 16  illustrates, generally at  1600 , a primary microphone location in the head wearable device from  FIG. 15  according to embodiments of the invention. With reference to  FIG. 16 , a main microphone location is illustrated at  1602 . 
       FIG. 17  illustrates, generally at  1700 , goggles with built-in acoustic noise cancellation system according to embodiments of the invention. With reference to  FIG. 17 , a head wearable device in the form of goggles  1702  is configured with a main microphone at a location  1704  and a reference microphone at a location  1706 . In various embodiments, noise cancellation electronics are included within goggles  1702 . Noise cancellation electronics are described more fully below in conjunction with the figures that follow. In other embodiments, noise cancellation electronics are not co-located with the head wearable device  1702  but are located external from the head wearable device  1702 . In such embodiments, a wireless communication link such as is compatible with the Bluetooth® protocol, ZigBee® protocol, etc. is provided to send the acoustic signals received from the microphones to an external location for processing by noise cancellation electronics. 
       FIG. 18  illustrates, generally at  1800 , a visor with built-in acoustic noise cancellation system according to embodiments of the invention. With reference to  FIG. 18 , a head wearable device in the form of a visor  1802  has a main microphone  1804  and a reference microphone  1806 . In various embodiments, noise cancellation electronics are included within the visor  1802 . Noise cancellation electronics are described more fully below in conjunction with the figures that follow. In other embodiments, noise cancellation electronics are not co-located with the head wearable device  1802  but are located external from the head wearable device  1802 . In such embodiments, a wireless communication link such as is compatible with the Bluetooth® protocol, ZigBee® protocol, etc. is provided to send the acoustic signals received from the microphones to an external location for processing by noise cancellation electronics. 
       FIG. 19  illustrates, generally at  1900 , a helmet with built-in acoustic noise cancellation system according to embodiments of the invention. With reference to  FIG. 19 , a head wearable device in the form of a helmet  1902  has a main microphone  1904  and a reference microphone  1906 . In various embodiments, noise cancellation electronics are included within the helmet  1902 . Noise cancellation electronics are described more fully below in conjunction with the figures that follow. In other embodiments, noise cancellation electronics are not co-located with the head wearable device  1902  but are located external from the head wearable device  1902 . In such embodiments, a wireless communication link such as is compatible with the Bluetooth® protocol, ZigBee® protocol, etc. is provided to send the acoustic signals received from the microphones to an external location for processing by noise cancellation electronics. 
       FIG. 20  illustrates, generally at  2000 , a process for extracting a desired audio signal according to embodiments of the invention. With reference to  FIG. 20 , a process starts at a block  2002 . At a block  2004 , a main acoustic signal is received from a main microphone located on a head wearable device. At a block  2006 , a reference acoustic signal is received from a reference microphone located on the head wearable device. At a block  2008 , a normalized main acoustic signal is formed. In various embodiments, the normalized main acoustic signal is formed using one or more reference acoustic signals as described in the figures below. At a block  2010  the normalized main acoustic signal is used to control noise cancellation using an acoustic signal processing system contained within the head wearable device. The process stops at a block  2012 . 
       FIG. 21  illustrates, generally at  2100 , system architecture, according to embodiments of the invention. With reference to  FIG. 21 , two acoustic channels are input into an adaptive noise cancellation unit  2106 . A first acoustic channel, referred to herein as main channel  2102 , is referred to in this description of embodiments synonymously as a “primary” or a “main” channel. The main channel  2102  contains both desired audio and undesired audio. The acoustic signal input on the main channel  2102  arises from the presence of both desired audio and undesired audio on one or more acoustic elements as described more fully below in the figures that follow. Depending on the configuration of a microphone or microphones used for the main channel the microphone elements can output an analog signal. The analog signal is converted to a digital signal with an analog-to-digital converter (AD) converter (not shown). Additionally, amplification can be located proximate to the microphone element(s) or AD converter. A second acoustic channel, referred to herein as reference channel  2104  provides an acoustic signal which also arises from the presence of desired audio and undesired audio. Optionally, a second reference channel  2104   b  can be input into the adaptive noise cancellation unit  2106 . Similar to the main channel and depending on the configuration of a microphone or microphones used for the reference channel, the microphone elements can output an analog signal. The analog signal is converted to a digital signal with an analog-to-digital converter (AD) converter (not shown). Additionally, amplification can be located proximate to the microphone element(s) or AD converter. In some embodiments the microphones are implemented as digital microphones. 
     In some embodiments, the main channel  2102  has an omni-directional response and the reference channel  2104  has an omni-directional response. In some embodiments, the acoustic beam patterns for the acoustic elements of the main channel  2102  and the reference channel  2104  are different. In other embodiments, the beam patterns for the main channel  2102  and the reference channel  2104  are the same; however, desired audio received on the main channel  2102  is different from desired audio received on the reference channel  2104 . Therefore, a signal-to-noise ratio for the main channel  2102  and a signal-to-noise ratio for the reference channel  2104  are different. In general, the signal-to-noise ratio for the reference channel is less than the signal-to-noise-ratio of the main channel. In various embodiments, by way of non-limiting examples, a difference between a main channel signal-to-noise ratio and a reference channel signal-to-noise ratio is approximately 1 or 2 decibels (dB) or more. In other non-limiting examples, a difference between a main channel signal-to-noise ratio and a reference channel signal-to-noise ratio is 1 decibel (dB) or less. Thus, embodiments of the invention are suited for high noise environments, which can result in low signal-to-noise ratios with respect to desired audio as well as low noise environments, which can have higher signal-to-noise ratios. As used in this description of embodiments, signal-to-noise ratio means the ratio of desired audio to undesired audio in a channel. Furthermore, the term “main channel signal-to-noise ratio” is used interchangeably with the term “main signal-to-noise ratio.” Similarly, the term “reference channel signal-to-noise ratio” is used interchangeably with the term “reference signal-to-noise ratio.” 
     The main channel  2102 , the reference channel  2104 , and optionally a second reference channel  2104   b  provide inputs to an adaptive noise cancellation unit  2106 . While a second reference channel is shown in the figures, in various embodiments, more than two reference channels are used. Adaptive noise cancellation unit  2106  filters undesired audio from the main channel  2102 , thereby providing a first stage of filtering with multiple acoustic channels of input. In various embodiments, the adaptive noise cancellation unit  2106  utilizes an adaptive finite impulse response (FIR) filter. The environment in which embodiments of the invention are used can present a reverberant acoustic field. Thus, the adaptive noise cancellation unit  2106  includes a delay for the main channel sufficient to approximate the impulse response of the environment in which the system is used. A magnitude of the delay used will vary depending on the particular application that a system is designed for including whether or not reverberation must be considered in the design. In some embodiments, for microphone channels positioned very closely together (and where reverberation is not significant) a magnitude of the delay can be on the order of a fraction of a millisecond. Note that at the low end of a range of values, which could be used for a delay, an acoustic travel time between channels can represent a minimum delay value. Thus, in various embodiments, a delay value can range from approximately a fraction of a millisecond to approximately 500 milliseconds or more depending on the application. Further description of the adaptive noise cancellation unit  1106  and the components associated therewith are provided below in conjunction with the figures that follow. 
     An output  2107  of the adaptive noise cancellation unit  2106  is input into a single channel noise cancellation unit  2118 . The single channel noise cancellation unit  2118  filters the output  2107  and provides a further reduction of undesired audio from the output  2107 , thereby providing a second stage of filtering. The single channel noise cancellation unit  2118  filters mostly stationary contributions to undesired audio. The single channel noise cancellation unit  2118  includes a linear filter, such as for example a Wiener filter, a Minimum Mean Square Error (MMSE) filter implementation, a linear stationary noise filter, or other Bayesian filtering approaches which use prior information about the parameters to be estimated. Filters used in the single channel noise cancellation unit  2118  are described more fully below in conjunction with the figures that follow. 
     Acoustic signals from the main channel  2102  are input at  2108  into a filter control  2112 . Similarly, acoustic signals from the reference channel  2104  are input at  2110  into the filter control  2112 . An optional second reference channel is input at  2108   b  into the filter control  2112 . Filter control  2112  provides control signals  2114  for the adaptive noise cancellation unit  2106  and control signals  2116  for the single channel noise cancellation unit  2118 . In various embodiments, the operation of filter control  2112  is described more completely below in conjunction with the figures that follow. An output  2120  of the single channel noise cancellation unit  2118  provides an acoustic signal which contains mostly desired audio and a reduced amount of undesired audio. 
     The system architecture shown in  FIG. 21  can be used in a variety of different systems used to process acoustic signals according to various embodiments of the invention. Some examples of the different acoustic systems are, but are not limited to, a mobile phone, a handheld microphone, a boom microphone, a microphone headset, a hearing aid, a hands free microphone device, a wearable system embedded in a frame of an eyeglass, a near-to-eye (NTE) headset display or headset computing device, a head wearable device of general configuration such as but not limited to glasses, goggles, a visor, a head band, a helmet, etc. The environments that these acoustic systems are used in can have multiple sources of acoustic energy incident upon the acoustic elements that provide the acoustic signals for the main channel  2102  and the reference channel  2104 . In various embodiments, the desired audio is usually the result of a user&#39;s own voice (see  FIG. 2  above). In various embodiments, the undesired audio is usually the result of the combination of the undesired acoustic energy from the multiple sources that are incident upon the acoustic elements used for both the main channel and the reference channel. Thus, the undesired audio is statistically uncorrelated with the desired audio. In addition, there is a non-causal relationship between the undesired audio in the main channel and the undesired audio in the reference channel. In such a case, echo cancellation does not work because of the non-causal relationship and because there is no measurement of a pure noise signal (undesired audio) apart from the signal of interest (desired audio). In echo cancellation noise reduction systems, a speaker, which generated the acoustic signal, provides a measure of a pure noise signal. In the context of the embodiments of the system described herein, there is no speaker, or noise source from which a pure noise signal could be extracted. 
       FIG. 22  illustrates, generally at  2112 , filter control, according to embodiments of the invention. With reference to  FIG. 22 , acoustic signals from the main channel  2102  are input at  2108  into a desired voice activity detection unit  2202 . Acoustic signals at  2108  are monitored by main channel activity detector  2206  to create a flag that is associated with activity on the main channel  2102  ( FIG. 21 ). Optionally, acoustic signals at  2110   b  are monitored by a second reference channel activity detector (not shown) to create a flag that is associated with activity on the second reference channel. Optionally, an output of the second reference channel activity detector is coupled to the inhibit control logic  2214 . Acoustic signals at  2110  are monitored by reference channel activity detector  2208  to create a flag that is associated with activity on the reference channel  2104  ( FIG. 21 ). The desired voice activity detection unit  2202  utilizes acoustic signal inputs from  2110 ,  2108 , and optionally  2110   b  to produce a desired voice activity signal  2204 . The operation of the desired voice activity detection unit  2202  is described more completely below in the figures that follow. 
     In various embodiments, inhibit logic unit  2214  receives as inputs, information regarding main channel activity at  2210 , reference channel activity at  2212 , and information pertaining to whether desired audio is present at  2204 . In various embodiments, the inhibit logic  2214  outputs filter control signal  2114 / 2116  which is sent to the adaptive noise cancellation unit  2106  and the single channel noise cancellation unit  2118  of  FIG. 21  for example. The implementation and operation of the main channel activity detector  2206 , the reference channel activity detector  2208  and the inhibit logic  2214  are described more fully in U.S. Pat. No. 7,386,135 titled “Cardioid Beam With A Desired Null Based Acoustic Devices, Systems and Methods,” which is hereby incorporated by reference. 
     In operation, in various embodiments, the system of  FIG. 21  and the filter control of  FIG. 22  provide for filtering and removal of undesired audio from the main channel  2102  as successive filtering stages are applied by adaptive noise cancellation unit  2106  and single channel nose cancellation unit  2118 . In one or more embodiments, throughout the system, application of the signal processing is applied linearly. In linear signal processing an output is linearly related to an input. Thus, changing a value of the input, results in a proportional change of the output. Linear application of signal processing processes to the signals preserves the quality and fidelity of the desired audio, thereby substantially eliminating or minimizing any non-linear distortion of the desired audio. Preservation of the signal quality of the desired audio is useful to a user in that accurate reproduction of speech helps to facilitate accurate communication of information. 
     In addition, algorithms used to process speech, such as Speech Recognition (SR) algorithms or Automatic Speech Recognition (ASR) algorithms benefit from accurate presentation of acoustic signals which are substantially free of non-linear distortion. Thus, the distortions which can arise from the application of signal processing processes which are non-linear are eliminated by embodiments of the invention. The linear noise cancellation algorithms, taught by embodiments of the invention, produce changes to the desired audio which are transparent to the operation of SR and ASR algorithms employed by speech recognition engines. As such, the error rates of speech recognition engines are greatly reduced through application of embodiments of the invention. 
       FIG. 23  illustrates, generally at  2300 , another diagram of system architecture, according to embodiments of the invention. With reference to  FIG. 23 , in the system architecture presented therein, a first channel provides acoustic signals from a first microphone at  2302  (nominally labeled in the figure as MIC  1 ). A second channel provides acoustic signals from a second microphone at  2304  (nominally labeled in the figure as MIC  2 ). In various embodiments, one or more microphones can be used to create the signal from the first microphone  2302 . In various embodiments, one or more microphones can be used to create the signal from the second microphone  2304 . In some embodiments, one or more acoustic elements can be used to create a signal that contributes to the signal from the first microphone  2302  and to the signal from the second microphone  2304  (see  FIG. 25C  described below). Thus, an acoustic element can be shared by  2302  and  2304 . In various embodiments, arrangements of acoustic elements which provide the signals at  2302 ,  2304 , the main channel, and the reference channel are described below in conjunction with the figures that follow. 
     A beamformer  2305  receives as inputs, the signal from the first microphone  2302  and the signal from the second microphone  2304  and optionally a signal from a third microphone  2304   b  (nominally labeled in the figure as MIC  3 ). The beamformer  2305  uses signals  2302 ,  2304  and optionally  2304   b  to create a main channel  2308   a  which contains both desired audio and undesired audio. The beamformer  2305  also uses signals  2302 ,  2304 , and optionally  2304   b  to create one or more reference channels  2310   a  and optionally  2311   a . A reference channel contains both desired audio and undesired audio. A signal-to-noise ratio of the main channel, referred to as “main channel signal-to-noise ratio” is greater than a signal-to-noise ratio of the reference channel, referred to herein as “reference channel signal-to-noise ratio.” The beamformer  2305  and/or the arrangement of acoustic elements used for MIC  1  and MIC  2  provide for a main channel signal-to-noise ratio which is greater than the reference channel signal-to-noise ratio. 
     The beamformer  2305  is coupled to an adaptive noise cancellation unit  2306  and a filter control unit  2312 . A main channel signal is output from the beamformer  2305  at  2308   a  and is input into an adaptive noise cancellation unit  2306 . Similarly, a reference channel signal is output from the beamformer  2305  at  2310   a  and is input into the adaptive noise cancellation unit  2306 . The main channel signal is also output from the beamformer  2305  and is input into a filter control  2312  at  2308   b . Similarly, the reference channel signal is output from the beamformer  2305  and is input into the filter control  2312  at  2310   b . Optionally, a second reference channel signal is output at  2311   a  and is input into the adaptive noise cancellation unit  2306  and the optional second reference channel signal is output at  2311   b  and is input into the filter control  2012 . 
     The filter control  2312  uses inputs  2308   b ,  2310   b , and optionally  2311   b  to produce channel activity flags and desired voice activity detection to provide filter control signal  2314  to the adaptive noise cancellation unit  2306  and filter control signal  2316  to a single channel noise reduction unit  2318 . 
     The adaptive noise cancellation unit  2306  provides multi-channel filtering and filters a first amount of undesired audio from the main channel  2308   a  during a first stage of filtering to output a filtered main channel at  2307 . The single channel noise reduction unit  2318  receives as an input the filtered main channel  2307  and provides a second stage of filtering, thereby further reducing undesired audio from  2307 . The single channel noise reduction unit  2318  outputs mostly desired audio at  2320 . 
     In various embodiments, different types of microphones can be used to provide the acoustic signals needed for the embodiments of the invention presented herein. Any transducer that converts a sound wave to an electrical signal is suitable for use with embodiments of the invention taught herein. Some non-limiting examples of microphones are, but are not limited to, a dynamic microphone, a condenser microphone, an Electret Condenser Microphone, (ECM), and a microelectromechanical systems (MEMS) microphone. In other embodiments a condenser microphone (CM) is used. In yet other embodiments micro-machined microphones are used. Microphones based on a piezoelectric film are used with other embodiments. Piezoelectric elements are made out of ceramic materials, plastic material, or film. In yet other embodiments, micromachined arrays of microphones are used. In yet other embodiments, silicon or polysilicon micromachined microphones are used. In some embodiments, bi-directional pressure gradient microphones are used to provide multiple acoustic channels. Various microphones or microphone arrays including the systems described herein can be mounted on or within structures such as eyeglasses or headsets. 
       FIG. 24A  illustrates, generally at  2400 , another diagram of system architecture incorporating auto-balancing, according to embodiments of the invention. With reference to  FIG. 24A , in the system architecture presented therein, a first channel provides acoustic signals from a first microphone at  2402  (nominally labeled in the figure as MIC  1 ). A second channel provides acoustic signals from a second microphone at  2404  (nominally labeled in the figure as MIC  2 ). In various embodiments, one or more microphones can be used to create the signal from the first microphone  2402 . In various embodiments, one or more microphones can be used to create the signal from the second microphone  2404 . In some embodiments, as described above in conjunction with  FIG. 23 , one or more acoustic elements can be used to create a signal that becomes part of the signal from the first microphone  2402  and the signal from the second microphone  2404 . In various embodiments, arrangements of acoustic elements which provide the signals  2402 ,  2404 , the main channel, and the reference channel are described below in conjunction with the figures that follow. 
     A beamformer  2405  receives as inputs, the signal from the first microphone  2402  and the signal from the second microphone  2404 . The beamformer  2405  uses signals  2402  and  2404  to create a main channel which contains both desired audio and undesired audio. The beamformer  2405  also uses signals  2402  and  2404  to create a reference channel. Optionally, a third channel provides acoustic signals from a third microphone at  2404   b  (nominally labeled in the figure as MIC  3 ), which are input into the beamformer  2405 . In various embodiments, one or more microphones can be used to create the signal  2404   b  from the third microphone. The reference channel contains both desired audio and undesired audio. A signal-to-noise ratio of the main channel, referred to as “main channel signal-to-noise ratio” is greater than a signal-to-noise ratio of the reference channel, referred to herein as “reference channel signal-to-noise ratio.” The beamformer  2405  and/or the arrangement of acoustic elements used for MIC  1 , MIC  2 , and optionally MIC  3  provide for a main channel signal-to-noise ratio that is greater than the reference channel signal-to-noise ratio. In some embodiments bi-directional pressure-gradient microphone elements provide the signals  2402 ,  2404 , and optionally  2404   b.    
     The beamformer  2405  is coupled to an adaptive noise cancellation unit  2406  and a desired voice activity detector  2412  (filter control). A main channel signal is output from the beamformer  2405  at  2408   a  and is input into an adaptive noise cancellation unit  2406 . Similarly, a reference channel signal is output from the beamformer  2405  at  2410   a  and is input into the adaptive noise cancellation unit  2406 . The main channel signal is also output from the beamformer  2405  and is input into the desired voice activity detector  2412  at  2408   b . Similarly, the reference channel signal is output from the beamformer  2405  and is input into the desired voice activity detector  2412  at  2410   b . Optionally, a second reference channel signal is output at  2409   a  from the beamformer  2405  and is input to the adaptive noise cancellation unit  2406 , and the second reference channel signal is output at  2409   b  from the beamformer  2405  and is input to the desired vice activity detector  2412 . 
     The desired voice activity detector  2412  uses input  2408   b ,  2410   b , and optionally  2409   b  to produce filter control signal  2414  for the adaptive noise cancellation unit  2408  and filter control signal  2416  for a single channel noise reduction unit  2418 . The adaptive noise cancellation unit  2406  provides multi-channel filtering and filters a first amount of undesired audio from the main channel  2408   a  during a first stage of filtering to output a filtered main channel at  2407 . The single channel noise reduction unit  2418  receives as an input the filtered main channel  2407  and provides a second stage of filtering, thereby further reducing undesired audio from  2407 . The single channel noise reduction unit  2418  outputs mostly desired audio at  2420   
     The desired voice activity detector  2412  provides a control signal  2422  for an auto-balancing unit  2424 . The auto-balancing unit  2424  is coupled at  2426  to the signal path from the first microphone  2402 . The auto-balancing unit  2424  is also coupled at  2428  to the signal path from the second microphone  2404 . Optionally, the auto-balancing unit  2424  is also coupled at  2429  to the signal path from the third microphone  2404   b . The auto-balancing unit  2424  balances the microphone response to far field signals over the operating life of the system. Keeping the microphone channels balanced increases the performance of the system and maintains a high level of performance by preventing drift of microphone sensitivities. The auto-balancing unit is described more fully below in conjunction with the figures that follow. 
       FIG. 24B  illustrates, generally at  2450 , processes for noise reduction, according to embodiments of the invention. With reference to  FIG. 24B , a process begins at a block  2452 . At a block  2454  a main acoustic signal is received by a system. The main acoustic signal can be for example, in various embodiments such a signal as is represented by  2102  ( FIG. 21 ),  2302 / 2308   a / 2308   b  ( FIG. 23 ), or  2402 / 2408   a / 2408   b  ( FIG. 24A ). At a block  2456  a reference acoustic signal is received by the system. The reference acoustic signal can be for example, in various embodiments such a signal as is represented by  2104  and optionally  2104   b  ( FIG. 21 ),  2304 / 2310   a / 2310   b  and optionally  2304   b / 2311   a / 2311   b  ( FIG. 23 ), or  2404 / 2410   a / 2410   b  and optionally  2404   b / 2409   a / 2409   b  ( FIG. 24A ). At a block  2458  adaptive filtering is performed with multiple channels of input, such as using for example the adaptive filter unit  2106  ( FIG. 21 ),  2306  ( FIG. 23 ), and  2406  ( FIG. 24A ) to provide a filtered acoustic signal for example as shown at  2107  ( FIG. 21 ),  2307  ( FIG. 23 ), and  2407  ( FIG. 24A ). At a block  2460  a single channel unit is used to filter the filtered acoustic signal which results from the process of the block  2458 . The single channel unit can be for example, in various embodiments, such a unit as is represented by  2118  ( FIG. 21 ),  2318  ( FIG. 23 ), or  2418  ( FIG. 24A ). The process ends at a block  2462 . 
     In various embodiments, the adaptive noise cancellation unit, such as  2106  ( FIG. 21 ),  2306  ( FIG. 23 ), and  2406  ( FIG. 24A ) is implemented in an integrated circuit device, which may include an integrated circuit package containing the integrated circuit. In some embodiments, the adaptive noise cancellation unit  2106  or  2306  or  2406  is implemented in a single integrated circuit die. In other embodiments, the adaptive noise cancellation unit  2106  or  2306  or  2406  is implemented in more than one integrated circuit die of an integrated circuit device which may include a multi-chip package containing the integrated circuit. 
     In various embodiments, the single channel noise cancellation unit, such as  2018  ( FIG. 21 ),  2318  ( FIG. 23 ), and  2418  ( FIG. 24A ) is implemented in an integrated circuit device, which may include an integrated circuit package containing the integrated circuit. In some embodiments, the single channel noise cancellation unit  2118  or  2318  or  2418  is implemented in a single integrated circuit die. In other embodiments, the single channel noise cancellation unit  2118  or  2318  or  2418  is implemented in more than one integrated circuit die of an integrated circuit device which may include a multi-chip package containing the integrated circuit. 
     In various embodiments, the filter control, such as  2112  ( FIGS. 21 &amp; 22 ) or  2312  ( FIG. 23 ) is implemented in an integrated circuit device, which may include an integrated circuit package containing the integrated circuit. In some embodiments, the filter control  2112  or  2312  is implemented in a single integrated circuit die. In other embodiments, the filter control  2112  or  2312  is implemented in more than one integrated circuit die of an integrated circuit device which may include a multi-chip package containing the integrated circuit. 
     In various embodiments, the beamformer, such as  2305  ( FIG. 23 ) or  2405  ( FIG. 24A ) is implemented in an integrated circuit device, which may include an integrated circuit package containing the integrated circuit. In some embodiments, the beamformer  2305  or  2405  is implemented in a single integrated circuit die. In other embodiments, the beamformer  2305  or  2405  is implemented in more than one integrated circuit die of an integrated circuit device which may include a multi-chip package containing the integrated circuit. 
       FIG. 25A  illustrates, generally at  2500 , beamforming according to embodiments of the invention. With reference to  FIG. 25A , a beamforming block  2506  is applied to two microphone inputs  2502  and  2504 . In one or more embodiments, the microphone input  2502  can originate from a first directional microphone and the microphone input  2504  can originate from a second directional microphone or microphone signals  2502  and  2504  can originate from omni-directional microphones. In yet other embodiments, microphone signals  2502  and  2504  are provided by the outputs of a bi-directional pressure gradient microphone. Various directional microphones can be used, such as but not limited to, microphones having a cardioid beam pattern, a dipole beam pattern, an omni-directional beam pattern, or a user defined beam pattern. In some embodiments, one or more acoustic elements are configured to provide the microphone input  2502  and  2504 . 
     In various embodiments, beamforming block  2506  includes a filter  2508 . Depending on the type of microphone used and the specific application, the filter  2508  can provide a direct current (DC) blocking filter which filters the DC and very low frequency components of Microphone input  2502 . Following the filter  2508 , in some embodiments additional filtering is provided by a filter  2510 . Some microphones have non-flat responses as a function of frequency. In such a case, it can be desirable to flatten the frequency response of the microphone with a de-emphasis filter. The filter  2510  can provide de-emphasis, thereby flattening a microphone&#39;s frequency response. Following de-emphasis filtering by the filter  2510 , a main microphone channel is supplied to the adaptive noise cancellation unit at  2512   a  and the desired voice activity detector at  2512   b.    
     A microphone input  2504  is input into the beamforming block  2506  and in some embodiments is filtered by a filter  2512 . Depending on the type of microphone used and the specific application, the filter  2512  can provide a direct current (DC) blocking filter which filters the DC and very low frequency components of Microphone input  2504 . A filter  2514  filters the acoustic signal which is output from the filter  2512 . The filter  2514  adjusts the gain, phase, and can also shape the frequency response of the acoustic signal. Following the filter  2514 , in some embodiments additional filtering is provided by a filter  2516 . Some microphones have non-flat responses as a function of frequency. In such a case, it can be desirable to flatten the frequency response of the microphone with a de-emphasis filter. The filter  2516  can provide de-emphasis, thereby flattening a microphone&#39;s frequency response. Following de-emphasis filtering by the filter  2516 , a reference microphone channel is supplied to the adaptive noise cancellation unit at  2518   a  and to the desired voice activity detector at  2518   b.    
     Optionally, a third microphone channel is input at  2504   b  into the beamforming block  2506 . Similar to the signal path described above for the channel  2504 , the third microphone channel is filtered by a filter  2512   b . Depending on the type of microphone used and the specific application, the filter  2512   b  can provide a direct current (DC) blocking filter which filters the DC and very low frequency components of Microphone input  2504   b . A filter  2514   b  filters the acoustic signal which is output from the filter  2512   b . The filter  2514   b  adjusts the gain, phase, and can also shape the frequency response of the acoustic signal. Following the filter  2514   b , in some embodiments additional filtering is provided by a filter  2516   b . Some microphones have non-flat responses as a function of frequency. In such a case, it can be desirable to flatten the frequency response of the microphone with a de-emphasis filter. The filter  2516   b  can provide de-emphasis, thereby flattening a microphone&#39;s frequency response. Following de-emphasis filtering by the filter  2516   b , a second reference microphone channel is supplied to the adaptive noise cancellation unit at  2520   a  and to the desired voice activity detector at  2520   b    
       FIG. 25B  presents, generally at  2530 , another illustration of beamforming according to embodiments of the invention. With reference to  FIG. 25B , a beam pattern is created for a main channel using a first microphone  2532  and a second microphone  2538 . A signal  2534  output from the first microphone  2532  is input to an adder  2536 . A signal  2540  output from the second microphone  2538  has its amplitude adjusted at a block  2542  and its phase adjusted by applying a delay at a block  2544  resulting in a signal  2546  which is input to the adder  2536 . The adder  2536  subtracts one signal from the other resulting in output signal  2548 . Output signal  2548  has a beam pattern which can take on a variety of forms depending on the initial beam patterns of microphone  2532  and  2538  and the gain applied at  2542  and the delay applied at  2544 . By way of non-limiting example, beam patterns can include cardioid, dipole, etc. 
     A beam pattern is created for a reference channel using a third microphone  2552  and a fourth microphone  2558 . A signal  2554  output from the third microphone  2552  is input to an adder  2556 . A signal  2560  output from the fourth microphone  2558  has its amplitude adjusted at a block  2562  and its phase adjusted by applying a delay at a block  2564  resulting in a signal  2566  which is input to the adder  2556 . The adder  2556  subtracts one signal from the other resulting in output signal  2568 . Output signal  2568  has a beam pattern which can take on a variety of forms depending on the initial beam patterns of microphone  2552  and  2558  and the gain applied at  2562  and the delay applied at  2564 . By way of non-limiting example, beam patterns can include cardioid, dipole, etc. 
       FIG. 25C  illustrates, generally at  2570 , beamforming with shared acoustic elements according to embodiments of the invention. With reference to  FIG. 25C , a microphone  2552  is shared between the main acoustic channel and the reference acoustic channel. The output from microphone  2552  is split and travels at  2572  to gain  2574  and to delay  2576  and is then input at  2586  into the adder  2536 . Appropriate gain at  2574  and delay at  2576  can be selected to achieve equivalently an output  2578  from the adder  2536  which is equivalent to the output  2548  from adder  2536  ( FIG. 25B ). Similarly gain  2582  and delay  2584  can be adjusted to provide an output signal  2588  which is equivalent to  2568  ( FIG. 25B ). By way of non-limiting example, beam patterns can include cardioid, dipole, etc. 
       FIG. 26  illustrates, generally at  2600 , multi-channel adaptive filtering according to embodiments of the invention. With reference to  FIG. 26 , embodiments of an adaptive filter unit are illustrated with a main channel  2604  (containing a microphone signal) input into a delay element  2606 . A reference channel  2602  (containing a microphone signal) is input into an adaptive filter  2608 . In various embodiments, the adaptive filter  2608  can be an adaptive FIR filter designed to implement normalized least-mean-square-adaptation (NLMS) or another algorithm. Embodiments of the invention are not limited to NLMS adaptation. The adaptive FIR filter filters an estimate of desired audio from the reference signal  2602 . In one or more embodiments, an output  2609  of the adaptive filter  2608  is input into an adder  2610 . The delayed main channel signal  2607  is input into the adder  2610  and the output  2609  is subtracted from the delayed main channel signal  2607 . The output of the adder  2616  provides a signal containing desired audio with a reduced amount of undesired audio. 
     Many environments that acoustic systems employing embodiments of the invention are used in present reverberant conditions. Reverberation results in a form of noise and contributes to the undesired audio which is the object of the filtering and signal extraction described herein. In various embodiments, the two channel adaptive FIR filtering represented at  2600  models the reverberation between the two channels and the environment they are used in. Thus, undesired audio propagates along the direct path and the reverberant path requiring the adaptive FIR filter to model the impulse response of the environment. Various approximations of the impulse response of the environment can be made depending on the degree of precision needed. In one non-limiting example, the amount of delay is approximately equal to the impulse response time of the environment. In another non-limiting example, the amount of delay is greater than an impulse response of the environment. In one embodiment, an amount of delay is approximately equal to a multiple n of the impulse response time of the environment, where n can equal 2 or 3 or more for example. Alternatively, an amount of delay is not an integer number of impulse response times, such as for example, 0.5, 1.4, 2.75, etc. For example, in one embodiment, the filter length is approximately equal to twice the delay chosen for  2606 . Therefore, if an adaptive filter having 200 taps is used, the length of the delay  2606  would be approximately equal to a time delay of 100 taps. A time delay equivalent to the propagation time through 100 taps is provided merely for illustration and does not imply any form of limitation to embodiments of the invention. 
     Embodiments of the invention can be used in a variety of environments which have a range of impulse response times. Some examples of impulse response times are given as non-limiting examples for the purpose of illustration only and do not limit embodiments of the invention. For example, an office environment typically has an impulse response time of approximately 100 milliseconds to 200 milliseconds. The interior of a vehicle cabin can provide impulse response times ranging from 30 milliseconds to 60 milliseconds. In general, embodiments of the invention are used in environments whose impulse response times can range from several milliseconds to 500 milliseconds or more. 
     The adaptive filter unit  2600  is in communication at  2614  with inhibit logic such as inhibit logic  2214  and filter control signal  2114  ( FIG. 22 ). Signals  2614  controlled by inhibit logic  2214  are used to control the filtering performed by the filter  2608  and adaptation of the filter coefficients. An output  2616  of the adaptive filter unit  2600  is input to a single channel noise cancellation unit such as those described above in the preceding figures, for example;  2118  ( FIG. 21 ),  2318  ( FIG. 23 ), and  2418  ( FIG. 24A ). A first level of undesired audio has been extracted from the main acoustic channel resulting in the output  2616 . Under various operating conditions the level of the noise, i.e., undesired audio can be very large relative to the signal of interest, i.e., desired audio. Embodiments of the invention are operable in conditions where some difference in signal-to-noise ratio between the main and reference channels exists. In some embodiments, the differences in signal-to-noise ratio are on the order of 1 decibel (dB) or less. In other embodiments, the differences in signal-to-noise ratio are on the order of 1 decibel (dB) or more. The output  2616  is filtered additionally to reduce the amount of undesired audio contained therein in the processes that follow using a single channel noise reduction unit. 
     Inhibit logic, described in  FIG. 22  above including signal  2614  ( FIG. 26 ) provide for the substantial non-operation of filter  2608  and no adaptation of the filter coefficients when either the main or the reference channels are determined to be inactive. In such a condition, the signal present on the main channel  2604  is output at  2616 . 
     If the main channel and the reference channels are active and desired audio is detected or a pause threshold has not been reached then adaptation is disabled, with filter coefficients frozen, and the signal on the reference channel  2602  is filtered by the filter  2608  subtracted from the main channel  2607  with adder  2610  and is output at  2616 . 
     If the main channel and the reference channel are active and desired audio is not detected and the pause threshold (also called pause time) is exceeded then filter coefficients are adapted. A pause threshold is application dependent. For example, in one non-limiting example, in the case of Automatic Speech Recognition (ASR) the pause threshold can be approximately a fraction of a second. 
       FIG. 27  illustrates, generally at  2700 , single channel filtering according to embodiments of the invention. With reference to  FIG. 27 , a single channel noise reduction unit utilizes a linear filter having a single channel input. Examples of filters suitable for use therein are a Wiener filter, a filter employing Minimum Mean Square Error (MMSE), etc. An output from an adaptive noise cancellation unit (such as one described above in the preceding figures) is input at  2704  into a filter  2702 . The input signal  2704  contains desired audio and a noise component, i.e., undesired audio, represented in equation  2714  as the total power (Ø DA +Ø UA ). The filter  2702  applies the equation shown at  2714  to the input signal  2704 . An estimate for the total power (Ø DA +Ø UA ) is one term in the numerator of equation  2714  and is obtained from the input to the filter  2704 . An estimate for the noise Ø UA , i.e., undesired audio, is obtained when desired audio is absent from signal  2704 . The noise estimate Ø UA  is the other term in the numerator, which is subtracted from the total power (Ø DA +Ø UA ). The total power is the term in the denominator of equation  2714 . The estimate of the noise Ø UA  (obtained when desired audio is absent) is obtained from the input signal  2704  as informed by signal  2716  received from inhibit logic, such as inhibit logic  2214  ( FIG. 22 ) which indicates when desired audio is present as well as when desired audio is not present. The noise estimate is updated when desired audio is not present on signal  2704 . When desired audio is present, the noise estimate is frozen and the filtering proceeds with the noise estimate previously established during the last interval when desired audio was not present. 
       FIG. 28A  illustrates, generally at  2800 , desired voice activity detection according to embodiments of the invention. With reference to  FIG. 28A , a dual input desired voice detector is shown at  2806 . Acoustic signals from a main channel are input at  2802 , from for example, a beamformer or from a main acoustic channel as described above in conjunction with the previous figures, to a first signal path  2807   a  of the dual input desired voice detector  2806 . The first signal path  2807   a  includes a voice band filter  2808 . The voice band filter  2808  captures the majority of the desired voice energy in the main acoustic channel  2802 . In various embodiments, the voice band filter  2808  is a band-pass filter characterized by a lower corner frequency an upper corner frequency and a roll-off from the upper corner frequency. In various embodiments, the lower corner frequency can range from 50 to 300 Hz depending on the application. For example, in wide band telephony, a lower corner frequency is approximately 50 Hz. In standard telephony the lower corner frequency is approximately 300 Hz. The upper corner frequency is chosen to allow the filter to pass a majority of the speech energy picked up by a relatively flat portion of the microphone&#39;s frequency response. Thus, the upper corner frequency can be placed in a variety of locations depending on the application. A non-limiting example of one location is 2,500 Hz. Another non-limiting location for the upper corner frequency is 4,000 Hz. 
     The first signal path  2807   a  includes a short-term power calculator  2810 . Short-term power calculator  2810  is implemented in various embodiments as a root mean square (RMS) measurement, a power detector, an energy detector, etc. Short-term power calculator  2810  can be referred to synonymously as a short-time power calculator  2810 . The short-term power detector  2810  calculates approximately the instantaneous power in the filtered signal. The output of the short-term power detector  2810  (Y 1 ) is input into a signal compressor  2812 . In various embodiments compressor  2812  converts the signal to the Log 2  domain, Log 10  domain, etc. In other embodiments, the compressor  2812  performs a user defined compression algorithm on the signal Y 1 . 
     Similar to the first signal path described above, acoustic signals from a reference acoustic channel are input at  2804 , from for example, a beamformer or from a reference acoustic channel as described above in conjunction with the previous figures, to a second signal path  2807   b  of the dual input desired voice detector  2806 . The second signal path  2807   b  includes a voice band filter  2816 . The voice band filter  2816  captures the majority of the desired voice energy in the reference acoustic channel  2804 . In various embodiments, the voice band filter  2816  is a band-pass filter characterized by a lower corner frequency an upper corner frequency and a roll-off from the upper corner frequency as described above for the first signal path and the voice-band filter  2808 . 
     The second signal path  2807   b  includes a short-term power calculator  2818 . Short-term power calculator  2818  is implemented in various embodiments as a root mean square (RMS) measurement, a power detector, an energy detector, etc. Short-term power calculator  2818  can be referred to synonymously as a short-time power calculator  2818 . The short-term power detector  2818  calculates approximately the instantaneous power in the filtered signal. The output of the short-term power detector  2818  (Y 2 ) is input into a signal compressor  2820 . In various embodiments compressor  2820  converts the signal to the Log 2  domain, Log 10  domain, etc. In other embodiments, the compressor  2820  performs a user defined compression algorithm on the signal Y 2 . 
     The compressed signal from the second signal path  2822  is subtracted from the compressed signal from the first signal path  2814  at a subtractor  2824 , which results in a normalized main signal at  2826  (Z). In other embodiments, different compression functions are applied at  2812  and  2820  which result in different normalizations of the signal at  2826 . In other embodiments, a division operation can be applied at  2824  to accomplish normalization when logarithmic compression is not implemented. Such as for example when compression based on the square root function is implemented. 
     The normalized main signal  2826  is input to a single channel normalized voice threshold comparator (SC-NVTC)  2828 , which results in a normalized desired voice activity detection signal  2830 . Note that the architecture of the dual channel voice activity detector provides a detection of desired voice using the normalized desired voice activity detection signal  2830  that is based on an overall difference in signal-to-noise ratios for the two input channels. Thus, the normalized desired voice activity detection signal  2830  is based on the integral of the energy in the voice band and not on the energy in particular frequency bins, thereby maintaining linearity within the noise cancellation units described above. The compressed signals  2814  and  2822 , utilizing logarithmic compression, provide an input at  2826  (Z) which has a noise floor that can take on values that vary from below zero to above zero (see column  2895   c , column  2895   d , or column  2895   e    FIG. 28E  below), unlike an uncompressed single channel input which has a noise floor which is always above zero (see column  2895   b    FIG. 28E  below). 
       FIG. 28B  illustrates, generally at  2850 , a single channel normalized voice threshold comparator (SC-NVTC) according to embodiments of the invention. With reference to  FIG. 28B , a normalized main signal  2826  is input into a long-term normalized power estimator  2832 . The long-term normalized power estimator  2832  provides a running estimate of the normalized main signal  2826 . The running estimate provides a floor for desired audio. An offset value  2834  is added in an adder  2836  to a running estimate of the output of the long-term normalized power estimator  2832 . The output of the adder  2838  is input to comparator  2840 . An instantaneous estimate  2842  of the normalized main signal  2826  is input to the comparator  2840 . The comparator  2840  contains logic that compares the instantaneous value at  2842  to the running ratio plus offset at  2838 . If the value at  2842  is greater than the value at  2838 , desired audio is detected and a flag is set accordingly and transmitted as part of the normalized desired voice activity detection signal  2830 . If the value at  2842  is less than the value at  2838  desired audio is not detected and a flag is set accordingly and transmitted as part of the normalized desired voice activity detection signal  2830 . The long-term normalized power estimator  2832  averages the normalized main signal  2826  for a length of time sufficiently long in order to slow down the change in amplitude fluctuations. Thus, amplitude fluctuations are slowly changing at  2833 . The averaging time can vary from a fraction of a second to minutes, by way of non-limiting examples. In various embodiments, an averaging time is selected to provide slowly changing amplitude fluctuations at the output of  2832 . 
       FIG. 28C  illustrates, generally at  2846 , desired voice activity detection utilizing multiple reference channels, according to embodiments of the invention. With reference to  FIG. 28C , a desired voice detector is shown at  2848 . The desired voice detector  2848  includes as an input the main channel  2802  and the first signal path  2807   a  (described above in conjunction with  FIG. 28A ) together with the reference channel  2804  and the second signal path  2807   b  (also described above in conjunction with  FIG. 28A ). In addition thereto, is a second reference acoustic channel  2850  which is input into the desired voice detector  2848  and is part of a third signal path  2807   c . Similar to the second signal path  2807   b  (described above), acoustic signals from the second reference acoustic channel are input at  2850 , from for example, a beamformer or from a second reference acoustic channel as described above in conjunction with the previous figures, to a third signal path  2807   c  of the multi-input desired voice detector  2848 . The third signal path  2807   c  includes a voice band filter  2852 . The voice band filter  2852  captures the majority of the desired voice energy in the second reference acoustic channel  2850 . In various embodiments, the voice band filter  2852  is a band-pass filter characterized by a lower corner frequency an upper corner frequency and a roll-off from the upper corner frequency as described above for the second signal path and the voice-band filter  2808 . 
     The third signal path  2807   c  includes a short-term power calculator  2854 . Short-term power calculator  2854  is implemented in various embodiments as a root mean square (RMS) measurement, a power detector, an energy detector, etc. Short-term power calculator  2854  can be referred to synonymously as a short-time power calculator  2854 . The short-term power detector  2854  calculates approximately the instantaneous power in the filtered signal. The output of the short-term power detector  2854  is input into a signal compressor  2856 . In various embodiments compressor  2856  converts the signal to the Log 2  domain, Log 10  domain, etc. In other embodiments, the compressor  2854  performs a user defined compression algorithm on the signal Y 3 . 
     The compressed signal from the third signal path  2858  is subtracted from the compressed signal from the first signal path  2814  at a subtractor  2860 , which results in a normalized main signal at  2862  (Z 2 ). In other embodiments, different compression functions are applied at  2856  and  2812  which result in different normalizations of the signal at  2862 . In other embodiments, a division operation can be applied at  2860  when logarithmic compression is not implemented. Such as for example when compression based on the square root function is implemented. 
     The normalized main signal  2862  is input to a single channel normalized voice threshold comparator (SC-NVTC)  2864 , which results in a normalized desired voice activity detection signal  2868 . Note that the architecture of the multi-channel voice activity detector provides a detection of desired voice using the normalized desired voice activity detection signal  2868  that is based on an overall difference in signal-to-noise ratios for the two input channels. Thus, the normalized desired voice activity detection signal  2868  is based on the integral of the energy in the voice band and not on the energy in particular frequency bins, thereby maintaining linearity within the noise cancellation units described above. The compressed signals  2814  and  2858 , utilizing logarithmic compression, provide an input at  2862  (Z 2 ) which has a noise floor that can take on values that vary from below zero to above zero (see column  2895   c , column  2895   d , or column  2895   e    FIG. 28E  below), unlike an uncompressed single channel input which has a noise floor which is always above zero (see column  2895   b    FIG. 28E  below). 
     The desired voice detector  2848 , having a multi-channel input with at least two reference channel inputs, provides two normalized desired voice activity detection signals  2868  and  2870  which are used to output a desired voice activity signal  2874 . In one embodiment, normalized desired voice activity detection signals  2868  and  2870  are input into a logical OR-gate  2872 . The logical OR-gate outputs the desired voice activity signal  2874  based on its inputs  2868  and  2870 . In yet other embodiments, additional reference channels can be added to the desired voice detector  2848 . Each additional reference channel is used to create another normalized main channel which is input into another single channel normalized voice threshold comparator (SC-NVTC) (not shown). An output from the additional single channel normalized voice threshold comparator (SC-NVTC) (not shown) is combined with  2874  via an additional exclusive OR-gate (also not shown) (in one embodiment) to provide the desired voice activity signal which is output as described above in conjunction with the preceding figures. Utilizing additional reference channels in a multi-channel desired voice detector, as described above, results in a more robust detection of desired audio because more information is obtained on the noise field via the plurality of reference channels. 
       FIG. 28D  illustrates, generally at  2880 , a process utilizing compression according to embodiments of the invention. With reference to  FIG. 28D , a process starts at a block  2882 . At a block  2884  a main acoustic channel is compressed, utilizing for example Log 10  compression or user defined compression as described in conjunction with  FIG. 28A  or  FIG. 28C . At a block  2886  a reference acoustic signal is compressed, utilizing for example Log 10  compression or user defined compression as described in conjunction with  FIG. 28A  or  FIG. 28C . At a block  2888  a normalized main acoustic signal is created. At a block  2890  desired voice is detected with the normalized acoustic signal. The process stops at a block  2892 . 
       FIG. 28E  illustrates, generally at  2893 , different functions to provide compression according to embodiments of the invention. With reference to  FIG. 28E , a table  2894  presents several compression functions for the purpose of illustration, no limitation is implied thereby. Column  2895   a  contains six sample values for a variable X. In this example, variable X takes on values as shown at  2896  ranging from 0.01 to 1000.0. Column  2895   b  illustrates no compression where Y=X. Column  2895   c  illustrates Log base 10 compression where the compressed value Y=Log 10(X). Column  2895   d  illustrates ln(X) compression where the compressed value Y=ln(X). Column  2895   e  illustrates Log base 2 compression where Y=Log 2 (X). A user defined compression (not shown) can also be implemented as desired to provide more or less compression than  2895   c ,  2895   d , or  2895   e . Utilizing a compression function at  2812  and  2820  ( FIG. 28A ) to compress the result of the short-term power detectors  2810  and  2818  reduces the dynamic range of the normalized main signal at  2826  (Z) which is input into the single channel normalized voice threshold comparator (SC-NVTC)  2828 . Similarly utilizing a compression function at  2812 ,  2820  and  2856  ( FIG. 28C ) to compress the results of the short-term power detectors  2810 ,  2818 , and  2854  reduces the dynamic range of the normalized main signals at  2826  (Z) and  2862  (Z 2 ) which are input into the SC-NVTC  828  and SC-NVTC  864  respectively. Reduced dynamic range achieved via compression can result in more accurately detecting the presence of desired audio and therefore a greater degree of noise reduction can be achieved by the embodiments of the invention presented herein. 
     In various embodiments, the components of the multi-input desired voice detector, such as shown in  FIG. 28A ,  FIG. 28B ,  FIG. 28C ,  FIG. 28D , and  FIG. 28E  are implemented in an integrated circuit device, which may include an integrated circuit package containing the integrated circuit. In some embodiments, the multi-input desired voice detector is implemented in a single integrated circuit die. In other embodiments, the multi-input desired voice detector is implemented in more than one integrated circuit die of an integrated circuit device which may include a multi-chip package containing the integrated circuit. 
       FIG. 29A  illustrates, generally at  2900 , an auto-balancing architecture according to embodiments of the invention. With reference to  FIG. 29A , an auto-balancing component  2903  has a first signal path  2905   a  and a second signal path  2905   b . A first acoustic channel  2902   a  (MIC  1 ) is coupled to the first signal path  2905   a  at  2902   b . A second acoustic channel  2904   a  is coupled to the second signal path  2905   b  at  2904   b . Acoustic signals are input at  2902   b  into a voice-band filter  2906 . The voice band filter  2906  captures the majority of the desired voice energy in the first acoustic channel  2902   a . In various embodiments, the voice band filter  1906  is a band-pass filter characterized by a lower corner frequency an upper corner frequency and a roll-off from the upper corner frequency. In various embodiments, the lower corner frequency can range from 50 to 300 Hz depending on the application. For example, in wide band telephony, a lower corner frequency is approximately 50 Hz. In standard telephony the lower corner frequency is approximately 300 Hz. The upper corner frequency is chosen to allow the filter to pass a majority of the speech energy picked up by a relatively flat portion of the microphone&#39;s frequency response. Thus, the upper corner frequency can be placed in a variety of locations depending on the application. A non-limiting example of one location is 2,500 Hz. Another non-limiting location for the upper corner frequency is 4,000 Hz. 
     The first signal path  2905   a  includes a long-term power calculator  2908 . Long-term power calculator  2908  is implemented in various embodiments as a root mean square (RMS) measurement, a power detector, an energy detector, etc. Long-term power calculator  2908  can be referred to synonymously as a long-time power calculator  2908 . The long-term power calculator  2908  calculates approximately the running average long-term power in the filtered signal. The output  2909  of the long-term power calculator  2908  is input into a divider  2917 . A control signal  2914  is input at  2916  to the long-term power calculator  2908 . The control signal  2914  provides signals as described above in conjunction with the desired audio detector, e.g.,  FIG. 28A ,  FIG. 28B ,  FIG. 28C  which indicate when desired audio is present and when desired audio is not present. Segments of the acoustic signals on the first channel  2902   b  which have desired audio present are excluded from the long-term power average produced at  2908 . 
     Acoustic signals are input at  2904   b  into a voice-band filter  2910  of the second signal path  2905   b . The voice band filter  2910  captures the majority of the desired voice energy in the second acoustic channel  2904   a . In various embodiments, the voice band filter  2910  is a band-pass filter characterized by a lower corner frequency an upper corner frequency and a roll-off from the upper corner frequency. In various embodiments, the lower corner frequency can range from 50 to 300 Hz depending on the application. For example, in wide band telephony, a lower corner frequency is approximately 50 Hz. In standard telephony the lower corner frequency is approximately 300 Hz. The upper corner frequency is chosen to allow the filter to pass a majority of the speech energy picked up by a relatively flat portion of the microphone&#39;s frequency response. Thus, the upper corner frequency can be placed in a variety of locations depending on the application. A non-limiting example of one location is 2,500 Hz. Another non-limiting location for the upper corner frequency is 4,000 Hz. 
     The second signal path  2905   b  includes a long-term power calculator  2912 . Long-term power calculator  2912  is implemented in various embodiments as a root mean square (RMS) measurement, a power detector, an energy detector, etc. Long-term power calculator  2912  can be referred to synonymously as a long-time power calculator  2912 . The long-term power calculator  2912  calculates approximately the running average long-term power in the filtered signal. The output  2913  of the long-term power calculator  2912  is input into a divider  2917 . A control signal  2914  is input at  2916  to the long-term power calculator  2912 . The control signal  2916  provides signals as described above in conjunction with the desired audio detector, e.g.,  FIG. 28A ,  FIG. 28B ,  FIG. 28C  which indicate when desired audio is present and when desired audio is not present. Segments of the acoustic signals on the second channel  2904   b  which have desired audio present are excluded from the long-term power average produced at  2912 . 
     In one embodiment, the output  2909  is normalized at  2917  by the output  2913  to produce an amplitude correction signal  2918 . In one embodiment, a divider is used at  2917 . The amplitude correction signal  2918  is multiplied at multiplier  2920  times an instantaneous value of the second microphone signal on  2904   a  to produce a corrected second microphone signal at  2922 . 
     In another embodiment, alternatively the output  2913  is normalized at  2917  by the output  2909  to produce an amplitude correction signal  2918 . In one embodiment, a divider is used at  2917 . The amplitude correction signal  2918  is multiplied by an instantaneous value of the first microphone signal on  1902   a  using a multiplier coupled to  2902   a  (not shown) to produce a corrected first microphone signal for the first microphone channel  2902   a . Thus, in various embodiments, either the second microphone signal is automatically balanced relative to the first microphone signal or in the alternative the first microphone signal is automatically balanced relative to the second microphone signal. 
     It should be noted that the long-term averaged power calculated at  2908  and  2912  is performed when desired audio is absent. Therefore, the averaged power represents an average of the undesired audio which typically originates in the far field. In various embodiments, by way of non-limiting example, the duration of the long-term power calculator ranges from approximately a fraction of a second such as, for example, one-half second to five seconds to minutes in some embodiments and is application dependent. 
       FIG. 29B  illustrates, generally at  2950 , auto-balancing according to embodiments of the invention. With reference to  FIG. 29B , an auto-balancing component  2952  is configured to receive as inputs a main acoustic channel  2954   a  and a reference acoustic channel  2956   a . The balancing function proceeds similarly to the description provided above in conjunction with  FIG. 29A  using the first acoustic channel  2902   a  (MIC  1 ) and the second acoustic channel  2904   a  (MIC  2 ). 
     With reference to  FIG. 29B , an auto-balancing component  2952  has a first signal path  2905   a  and a second signal path  2905   b . A first acoustic channel  2954   a  (MAIN) is coupled to the first signal path  2905   a  at  2954   b . A second acoustic channel  2956   a  is coupled to the second signal path  2905   b  at  2956   b . Acoustic signals are input at  2954   b  into a voice-band filter  2906 . The voice band filter  2906  captures the majority of the desired voice energy in the first acoustic channel  2954   a . In various embodiments, the voice band filter  2906  is a band-pass filter characterized by a lower corner frequency an upper corner frequency and a roll-off from the upper corner frequency. In various embodiments, the lower corner frequency can range from 50 to 300 Hz depending on the application. For example, in wide band telephony, a lower corner frequency is approximately 50 Hz. In standard telephony the lower corner frequency is approximately 300 Hz. The upper corner frequency is chosen to allow the filter to pass a majority of the speech energy picked up by a relatively flat portion of the microphone&#39;s frequency response. Thus, the upper corner frequency can be placed in a variety of locations depending on the application. A non-limiting example of one location is 2,500 Hz. Another non-limiting location for the upper corner frequency is 4,000 Hz. 
     The first signal path  2905   a  includes a long-term power calculator  2908 . Long-term power calculator  2908  is implemented in various embodiments as a root mean square (RMS) measurement, a power detector, an energy detector, etc. Long-term power calculator  2908  can be referred to synonymously as a long-time power calculator  2908 . The long-term power calculator  2908  calculates approximately the running average long-term power in the filtered signal. The output  2909   b  of the long-term power calculator  2908  is input into a divider  2917 . A control signal  2914  is input at  2916  to the long-term power calculator  2908 . The control signal  2914  provides signals as described above in conjunction with the desired audio detector, e.g.,  FIG. 28A ,  FIG. 28B ,  FIG. 28C  which indicate when desired audio is present and when desired audio is not present. Segments of the acoustic signals on the first channel  2954   b  which have desired audio present are excluded from the long-term power average produced at  2908 . 
     Acoustic signals are input at  2956   b  into a voice-band filter  2910  of the second signal path  2905   b . The voice band filter  2910  captures the majority of the desired voice energy in the second acoustic channel  2956   a . In various embodiments, the voice band filter  2910  is a band-pass filter characterized by a lower corner frequency an upper corner frequency and a roll-off from the upper corner frequency. In various embodiments, the lower corner frequency can range from 50 to 300 Hz depending on the application. For example, in wide band telephony, a lower corner frequency is approximately 50 Hz. In standard telephony the lower corner frequency is approximately 300 Hz. The upper corner frequency is chosen to allow the filter to pass a majority of the speech energy picked up by a relatively flat portion of the microphone&#39;s frequency response. Thus, the upper corner frequency can be placed in a variety of locations depending on the application. A non-limiting example of one location is 2,500 Hz. Another non-limiting location for the upper corner frequency is 4,000 Hz. 
     The second signal path  2905   b  includes a long-term power calculator  2912 . Long-term power calculator  2912  is implemented in various embodiments as a root mean square (RMS) measurement, a power detector, an energy detector, etc. Long-term power calculator  2912  can be referred to synonymously as a long-time power calculator  2912 . The long-term power calculator  2912  calculates approximately the running average long-term power in the filtered signal. The output  2913   b  of the long-term power calculator  2912  is input into the divider  2917 . A control signal  2914  is input at  2916  to the long-term power calculator  2912 . The control signal  2916  provides signals as described above in conjunction with the desired audio detector, e.g.,  FIG. 28A ,  FIG. 28B ,  FIG. 28C  which indicate when desired audio is present and when desired audio is not present. Segments of the acoustic signals on the second channel  2956   b  which have desired audio present are excluded from the long-term power average produced at  2912 . 
     In one embodiment, the output  2909   b  is normalized at  2917  by the output  2913   b  to produce an amplitude correction signal  2918   b . In one embodiment, a divider is used at  2917 . The amplitude correction signal  2918   b  is multiplied at multiplier  2920  times an instantaneous value of the second microphone signal on  2956   a  to produce a corrected second microphone signal at  2922   b.    
     In another embodiment, alternatively the output  2913   b  is normalized at  2917  by the output  2909   b  to produce an amplitude correction signal  2918   b . In one embodiment, a divider is used at  2917 . The amplitude correction signal  2918   b  is multiplied by an instantaneous value of the first microphone signal on  2954   a  using a multiplier coupled to  2954   a  (not shown) to produce a corrected first microphone signal for the first microphone channel  2954   a . Thus, in various embodiments, either the second microphone signal is automatically balanced relative to the first microphone signal or in the alternative the first microphone signal is automatically balanced relative to the second microphone signal. 
     It should be noted that the long-term averaged power calculated at  2908  and  2912  is performed when desired audio is absent. Therefore, the averaged power represents an average of the undesired audio which typically originates in the far field. In various embodiments, by way of non-limiting example, the duration of the long-term power calculator ranges from approximately a fraction of a second such as, for example, one-half second to five seconds to minutes in some embodiments and is application dependent. 
     Embodiments of the auto-balancing component  2902  or  2952  are configured for auto-balancing a plurality of microphone channels such as is indicated in  FIG. 24A . In such configurations, a plurality of channels (such as a plurality of reference channels) is balanced with respect to a main channel. Or a plurality of reference channels and a main channel are balanced with respect to a particular reference channel as described above in conjunction with  FIG. 29A  or  FIG. 29B . 
       FIG. 29C  illustrates filtering according to embodiments of the invention. With reference to  FIG. 29C, 2960   a  shows two microphone signals  2966   a  and  2968   a  having amplitude  2962  plotted as a function of frequency  2964 . In some embodiments, a microphone does not have a constant sensitivity as a function of frequency. For example, microphone response  2966   a  can illustrate a microphone output (response) with a non-flat frequency response excited by a broadband excitation which is flat in frequency. The microphone response  2966   a  includes a non-flat region  2974  and a flat region  2970 . For this example, a microphone which produced the response  2968   a  has a uniform sensitivity with respect to frequency; therefore  2968   a  is substantially flat in response to the broadband excitation which is flat with frequency. In some embodiments, it is of interest to balance the flat region  2970  of the microphones&#39; responses. In such a case, the non-flat region  2974  is filtered out so that the energy in the non-flat region  2974  does not influence the microphone auto-balancing procedure. What is of interest is a difference  2972  between the flat regions of the two microphones&#39; responses. 
     In  2960   b  a filter function  2978   a  is shown plotted with an amplitude  2976  plotted as a function of frequency  2964 . In various embodiments, the filter function is chosen to eliminate the non-flat portion  2974  of a microphone&#39;s response. Filter function  2978   a  is characterized by a lower corner frequency  2978   b  and an upper corner frequency  2978   c . The filter function of  2960   b  is applied to the two microphone signals  2966   a  and  2968   a  and the result is shown in  2960   c.    
     In  2960   c  filtered representations  2966   c  and  2968   c  of microphone signals  2966   a  and  2968   a  are plotted as a function of amplitude  2980  and frequency  2966 . A difference  2972  characterizes the difference in sensitivity between the two filtered microphone signals  2966   c  and  2968   c . It is this difference between the two microphone responses that is balanced by the systems described above in conjunction with  FIG. 29A  and  FIG. 29B . Referring back to  FIG. 29A  and  FIG. 29B , in various embodiments, voice band filters  2906  and  2910  can apply, in one non-limiting example, the filter function shown in  2960   b  to either microphone channels  2902   b  and  2904   b  ( FIG. 29A ) or to main and reference channels  2954   b  and  2956   b  ( FIG. 29B ). The difference  2972  between the two microphone channels is minimized or eliminated by the auto-balancing procedure described above in  FIG. 29A  or  FIG. 29B . 
       FIG. 30  illustrates, generally at  3000 , a process for auto-balancing according to embodiments of the invention. With reference to  FIG. 30 , a process starts at a block  3002 . At a block  3004  an average long-term power in a first microphone channel is calculated. The averaged long-term power calculated for the first microphone channel does not include segments of the microphone signal that occurred when desired audio was present. Input from a desired voice activity detector is used to exclude the relevant portions of desired audio. At a block  3006  an average power in a second microphone channel is calculated. The averaged long-term power calculated for the second microphone channel does not include segments of the microphone signal that occurred when desired audio was present. Input from a desired voice activity detector is used to exclude the relevant portions of desired audio. At a block  3008  an amplitude correction signal is computed using the averages computed in the block  3004  and the block  3006 . 
     In various embodiments, the components of auto-balancing component  2903  or  2952  are implemented in an integrated circuit device, which may include an integrated circuit package containing the integrated circuit. In some embodiments, auto-balancing components  2903  or  2952  are implemented in a single integrated circuit die. In other embodiments, auto-balancing components  2903  or  2952  are implemented in more than one integrated circuit die of an integrated circuit device which may include a multi-chip package containing the integrated circuit. 
       FIG. 31  illustrates, generally at  3100 , an acoustic signal processing system in which embodiments of the invention may be used. The block diagram is a high-level conceptual representation and may be implemented in a variety of ways and by various architectures. With reference to  FIG. 31 , bus system  3102  interconnects a Central Processing Unit (CPU)  3104 , Read Only Memory (ROM)  3106 , Random Access Memory (RAM)  3108 , storage  3110 , display  3120 , audio  3122 , keyboard  3124 , pointer  3126 , data acquisition unit (DAU)  3128 , and communications  3130 . The bus system  3102  may be for example, one or more of such buses as a system bus, Peripheral Component Interconnect (PCI), Advanced Graphics Port (AGP), Small Computer System Interface (SCSI), Institute of Electrical and Electronics Engineers (IEEE) standard number 1394 (FireWire), Universal Serial Bus (USB), or a dedicated bus designed for a custom application, etc. The CPU  3104  may be a single, multiple, or even a distributed computing resource or a digital signal processing (DSP) chip. Storage  3110  may be Compact Disc (CD), Digital Versatile Disk (DVD), hard disks (HD), optical disks, tape, flash, memory sticks, video recorders, etc. The acoustic signal processing system  3100  can be used to receive acoustic signals that are input from a plurality of microphones (e.g., a first microphone, a second microphone, etc.) or from a main acoustic channel and a plurality of reference acoustic channels as described above in conjunction with the preceding figures. Note that depending upon the actual implementation of the acoustic signal processing system, the acoustic signal processing system may include some, all, more, or a rearrangement of components in the block diagram. In some embodiments, aspects of the system  3100  are performed in software. While in some embodiments, aspects of the system  3100  are performed in dedicated hardware such as a digital signal processing (DSP) chip, etc. as well as combinations of dedicated hardware and software as is known and appreciated by those of ordinary skill in the art. 
     Thus, in various embodiments, acoustic signal data is received at  3129  for processing by the acoustic signal processing system  3100 . Such data can be transmitted at  3132  via communications interface  3130  for further processing in a remote location. Connection with a network, such as an intranet or the Internet is obtained via  3132 , as is recognized by those of skill in the art, which enables the acoustic signal processing system  3100  to communicate with other data processing devices or systems in remote locations. 
     For example, embodiments of the invention can be implemented on a computer system  3100  configured as a desktop computer or work station, on for example a WINDOWS® compatible computer running operating systems such as WINDOWS® XP Home or WINDOWS® XP Professional, Linux, Unix, etc. as well as computers from APPLE COMPUTER, Inc. running operating systems such as OS X, etc. Alternatively, or in conjunction with such an implementation, embodiments of the invention can be configured with devices such as speakers, earphones, video monitors, etc. configured for use with a Bluetooth communication channel. In yet other implementations, embodiments of the invention are configured to be implemented by mobile devices such as a smart phone, a tablet computer, a wearable device, such as eyeglasses, a near-to-eye (NTE) headset, a head wearable device of general configuration such as but not limited to glasses, goggles, a visor, a head band, a helmet, etc. or the like. 
     In one or more embodiments, a user is provided with hearing assistance to facilitate hearing sounds from a local environment. 
       FIG. 32A  illustrates, generally at  3200 , microphone configurations on a head wearable device according to embodiments of the invention.  FIG. 32B  illustrates, generally at  3220 , microphone configurations on a head wearable device in top view corresponding to  FIG. 32A  according to embodiments of the invention.  FIG. 32C  illustrates, generally at  3240 , microphone configurations on a head wearable device in bottom view corresponding to  FIG. 32A  according to embodiments of the invention.  FIG. 33  illustrates, generally at  3300 , the head wearable device from  FIG. 32A  relative to different sound sources according to embodiments of the invention. With reference to  FIG. 32A  through  FIG. 33  collectively, a head wearable device  3201  is presented in the shape of eyeglasses for use in a three-dimensional space. The three-dimensional space is illustrated by X, Y, Z axes at  3301  ( FIG. 33 ). The three-dimensional space is illustrated with what is known in the art as a cartesian coordinate system. However, no limitation is implied thereby. The three-dimensional space could have been illustrated with another coordinate system. In other embodiments, the head wearable device is in the shape of goggles, etc. Without limitation implied thereby, the term “eyeglasses” or “eyewear device” will be used synonymously with head wearable device, herein. The head wearable device  3201  has a front frame, the front frame contains one or more lenses typically made from glass or plastic, a left frame  3214 , and a right frame  3212 . The left and right frames are also referred to in the art as temples. The head wearable device is illustrated with four microphones, Microphone  0  ( 3202 ), Microphone  1  ( 3204 ), Microphone  2  ( 3206 ), and Microphone  3  ( 3210 ). In one or more embodiments, Microphone  0  ( 3202 ) is located under the bottom of the left-side frame  3214  and Microphone  1  ( 3204 ) and Microphone  2  ( 3206 ) are located on a top of the left-side frame  3214 . Microphone  3  ( 3210 ) is located on a top of the right-side frame  3212 . Alternatively, Microphone  0  ( 3202 ), Microphone  1  ( 3204 ), and Microphone  2  ( 3206 ) are located on the right-side frame  3212  and Microphone  3  ( 3210 ) is located on the left side frame  3214 . 
     In various embodiments, the eyewear device includes an array of microphones coupled to at least one side frame member. The array of microphones includes at least a first, and a second microphone. In one or more embodiments, the first and second microphones, for example  3202  and  3204 , are located at the side frame member  3214  close to the front frame member. The distance of the first and second microphones from the front frame member is approximately between 5 mm and 30 mm and can be around 15 mm as indicated by L 2  at  3209  ( FIG. 32B ). The first microphone (Microphone  0  ( 3202 )) is located at the bottom side of the side frame member  3214  and the second microphone (Microphone  1  ( 3204 )) is located on the top side of the side member  3214  and directly or nearly on top of the side frame member  3214 . In another embodiment, a third microphone (Microphone  2  ( 3206 )) is located on the side frame member  3214  and further away from the front frame member. The location of the third microphone (Microphone  2  ( 3206 )) from the first and/or second microphone ( 3202 / 3204 ) is approximately between 10 mm and 20 mm and can be around 15 mm as indicated by L 1  at  3208 . If the distance L 1  is too long, the third microphone (Microphone  2  ( 3206 )) may be close to a speaker embedded in the side frame member and located near the ear of the wearer. In this case, there could be an echo from the speaker to Microphone  2  ( 3206 ). Such an echo is remedied by decreasing the distance L 1  for a particular implementation. Decreasing the distance L 1  increases a separation distance between Microphone  2  ( 3206 ) and a speaker  3350  and reduces any echo thereby. 
     In another embodiment, a fourth microphone (Microphone  3  ( 3210 )) is located at the other side frame member  3212 . The Microphone  3  ( 3210 ) is illustrated close to the front frame member but other locations along the frame member  3212  are possible. The distance between Microphone  1  ( 3204 ) and Microphone  3  ( 3210 ) is determined by a width of the glasses frame and the distance is big enough for the system to detect the signal level difference from the two microphones. The distance between Microphone  1  ( 3204 ) and Microphone  3  ( 3210 ) is not a fixed number but is instead generally provided by a geometry and dimensions of the head wearable device. Similarly, the distance between Microphone  0  ( 3202 ) and Microphone  3  ( 3210 ) is not a fixed number but is instead generally provided by a geometry and dimensions of the head wearable device. 
       FIG. 32D  illustrates, generally at  3260 , another set of microphone placements in perspective view on a head wearable device according to embodiments of the invention.  FIG. 32E  illustrates, generally at  3280 , microphone placements on a head wearable device in bottom view corresponding to  FIG. 32D  according to embodiments of the invention. Referring to  FIG. 32D , microphone  0  ( 3202 ) and Microphone  1  ( 3204 ) are located on an inside surface of the temple  3212 . Microphone  2  ( 3206 ) is located on a bottom surface of the right temple  3212  and is setback from Microphone  0  ( 3202 )/Microphone  1  ( 3204 ) by an amount equal to L 1  as described above. A distance between Microphone  0  ( 3202 )/Microphone  1  ( 3204 ) and the front frame is as described above with L 2  ( FIG. 32B ). Referring back to  FIG. 32D , microphone  3  ( 3210 ) is located on a bottom side of the left temple  3210 . Alternatively, one or both of Microphones  2  ( 3206 ) and Microphone  3  ( 3210 ) can be located on a top surface of their respective temples. 
     In an alternative embodiment, the microphone placements shown in  FIGS. 32D / 32 E can be reversed with respect to the temples. For example, Microphone  0  ( 3202 ), Microphone  1  ( 3204 ), and Microphone  2  ( 3206 ) can be located on an inside surface of the left temple  3214  and Microphone  3  ( 3210 ) can be located on the right temple  3212 . 
     The four microphones described support three or more microphone combinations for the different use scenarios described herein as Configuration 1 using Microphone  0  and Microphone  1 ; Configuration 2 using Microphone  1  and Microphone  2 ; and Configuration 3 using Microphone  1  and Microphone  3 . In some embodiments, software interfaces are used to control the switching between these combinations of microphones and sequencing between configurations. 
     In various embodiments, the eyeglasses will have more than four microphones or less than four microphones. Four microphones are used for illustration of one or more embodiments as described herein and do not limit embodiments of the invention. Three configurations of microphones are described below to receive and to process acoustic signals for use by the user of the head wearable device to assist the user&#39;s hearing and in some instances for use remotely by e.g., speech recognition, command and control, reception and hearing by another user, as well as for local use by embedded speech recognition, etc. The Configurations described below can be used to provide primary and reference acoustic signals for use in the noise cancellation systems as described above. 
     Configuration 1 
     In one or more embodiments, Microphone  0  and Microphone  1  are used to process acoustic signals when a user is speaking while wearing the head wearable device  101 . In Configuration 1, the signals output from Microphone  0  and Microphone  1  are beamformed to place a main acoustic response downward along an axis  3302 . The axis  3302  is in a nominal direction of the user&#39;s mouth  3310  but need not be precisely aligned thereto. Microphone  0  and Microphone  1  have different acoustic distances to the user&#39;s mouth  3320 , with an acoustic distance for Microphone  0  being less than an acoustic distance for Microphone  1 . Acoustic signals  3312 , emanating from a user&#39;s mouth  3310 , are received with maximum acoustic sensitivity to the direction of the user  3310  relative to the microphone pair Microphone  0  and Microphone  1 . An acoustic signal so obtained, is used as a primary signal for input into a multichannel noise cancellation system. A reference signal, containing mostly noise (mostly undesired audio), is obtained by beamforming the microphone pair Microphone  0  and Microphone  1  with a main response steered 180 degrees away from the acoustic source  3310 . Thus, the reference signal is obtained in a direction looking up along the axis  3302  away from the user&#39;s mouth  3310 , towards potential noise sources, such as a noise source represented by  3360 , emitting noise  3362  (undesired audio). A signal so obtained, looking away from the user&#39;s mouth  3310 , is used as a reference signal for input into a multichannel noise cancellation system as described above. The beamforming applied to the reference signal minimizes acoustic sensitivity to signals arriving from the user&#39;s mouth  3310  and maximizes sensitivity to noise generated away from the direction of the user&#39;s mouth. Thus, a signal-to-noise ratio difference between Microphone  0  and Microphone  1  is maximized thereby to provide a reduction of noise from the primary signal through subsequent application of noise cancellation. 
     Processing to reduce noise (undesired audio) from the signal of interest (desired audio) permits the combination of Microphone  0  and Microphone  1  to help enhance the user&#39;s voice for phone calls in noisy environments. It also helps the command and control performance of the system when used in noisy environments. In noisy environments, the user&#39;s voice is buried within the background noises and is hard to be understood by the far-side listener during a phone call or to be recognized by a speech engine. The Microphone  0  and Microphone  1  combination uses beamforming technology to improve both the signal-to-noise ratio (SNR) of the user&#39;s voice over the background noise (as well as increasing a signal-to-noise ratio difference between Microphone  0  and Microphone  1 ) and the voice activity detection accuracy for the noise cancellation is improved thereby. This combination provides useful performance gains even in a very noisy environment having a 90-dB or greater background noise amplitude. As described above, Microphone  0  and Microphone  1  can be implemented with omni-directional microphones. 
     Configuration 2 
     In one or more embodiments, Microphone  1  and Microphone  2  are used to process acoustic signals when a user is listening to a remote sound source such as  3330 , while wearing the head wearable device  3201 . In Configuration 2, the signals output from Microphone  1  and Microphone  2  are beamformed to place a main acoustic response forward along an axis  3304 , thereby receiving acoustic signals  3332  emanating from the sound source indicated at  3330  with maximum acoustic sensitivity steered to a direction of the sound source  3330  relative to the microphone pair Microphone  1  and Microphone  2 . A signal so obtained, is used as a primary signal for input into a multichannel noise cancellation system. A reference signal, containing mostly noise, can be obtained from Microphone  2  with or without beamforming. When omnidirectional microphones are used for Microphone  1  and Microphone  2 , beamforming Microphone  1  and Microphone  2  to obtain a primary signal while using Microphone  2  alone for the reference signal, without beamforming with Microphone  1 , increases a sensitivity of the beamformed pair in the direction of a source  3330  by approximately 6 dB relative to a sensitivity of Microphone  2  alone to the source  3330 . Such processing provides a significant signal-to-noise ratio difference between Microphone  1  and Microphone  2 , which is advantageous to noise cancellation performance. The axis  3304  is pointing in a nominal direction forward of the user but need not be precisely aligned thereto. Microphone  1  and Microphone  2  have different acoustic distances to a sound source located forward of the user such as  3330 . An acoustic distance between the sound source  3330  and Microphone  1  less than an acoustic distance between Microphone  2  and the sound source  3330 . Thus, Microphone  1  and Microphone  2  can be flexibly located on a head wearable device in order to provide different acoustic distances relative to a sound source located in front of the head wearable device while not necessarily pointing directly at the sound source  3330 . 
     In alternative embodiments, beamforming the microphone pair Microphone  1  and Microphone  2  with a main response steered 180 degrees away from the acoustic source  3330  can be used to provide a reference signal (mostly undesired audio). Note that it is desirable to obtain a reference signal with a minimum amount of desired audio combined therewith. The reference signal can be obtained according to both methods, compared, and then a selection can be made based on the best system performance. Thus, a reference signal, so obtained by either method, has a signal-to-noise ratio that is less than the signal-to-noise ratio of the primary signal. Therefore, a signal-to-noise ratio difference is obtained for the Microphone  1 /Microphone  2  pair with respect to signals of interest originating from a direction that is nominally in front of the head wearable device  3201 , such as for example  3330 / 3332 . Signals so obtained via either method described above, looking away from the source  3330 , are used as a reference signal for input into a multichannel noise cancellation system. The beamforming used for the reference signal is chosen to provide minimum acoustic sensitivity to signals arriving from in front of the user such as the source  3330  (desired audio) and maximizes sensitivity to noise generated from directions other than the source  3330 . Thus, a signal-to-noise ratio difference between Microphone  1  and Microphone  2  is maximized thereby to provide reduction of noise from the primary signal through subsequent application of noise cancellation. 
     The output of the noise cancellation system is then provided on a speaker(s)  3350  to assist the user&#39;s hearing of the sound source  3330 . The speaker  3350  is incorporated into one or both side frames of the eyeglasses  3201 . Thus, in various embodiments, the Microphone  1 , Microphone  2  combination is used to enhance a user&#39;s hearing, such as for example during some activities like watching television or having a conversation with a person in front of the user wearing the eyeglasses  3201 . Some people with hearing difficulties are not able to understand audio signal clearly especially in a noisy environment. Combination 2 applies beamforming technology to help the user focus on the audio signals of interest by spatially removing the background noise. 
     Configuration 3 
     In one or more embodiments, Microphone  1  and Microphone  3  are used to process acoustic signals when a user is listening to or interacting with a remote sound source such as  3320  or  3340 , arriving from one side or the other while wearing the head wearable device  3201 . Alternatively, Microphone  3  and Microphone  2  are used to process the signals for Configuration 3 or Microphone  3  and Microphone  0  are used. The description that follows for Configuration 3 is provided in terms of Microphone  3  and Microphone  1  with no limitation implied thereby. In Configuration 3, the sound energy output from Microphone  1  and Microphone  3  are compared to determine which side of the user the loudest sound is coming from. Such information is useful because, in a meeting for example, with people sitting around a table, various people will speak from time-to-time thereby producing different arrival directions relative to a user wearing the eyeglasses  3201 . In Configuration 3, the signals output from a selected pair of microphones are processed to place a main acoustic response along an axis  3306 . The axis  3306  is in a nominal direction of the sound source but need not be precisely aligned thereto. The selected pair of microphones, e.g., one of Microphone  3  and Microphone  0 , Microphone  3  and Microphone  1 , or Microphone  3  and Microphone  2  have different acoustic distances to the sound source. 
     Following one method of operation, a primary microphone is the microphone from the Microphone  1 , Microphone  3  pair with the largest sound energy output. The other microphone of the Microphone  1 , Microphone  3  pair is then assigned to be the reference microphone. Alternative processing of the primary and the reference signals can follow the determination of which microphone is outputting the largest sound energy. For example, in one or more embodiments, beamforming is applied to the signals output from Microphone  1  and Microphone  3 . In one example, the primary signal is obtained when the main response axis of the beamforming process is steered to the side (direction) where the largest sound energy is being measured. In this example, the reference signal is obtained by steering the main response axis of the beamforming process to the opposite side as that of the primary. 
     A variation on this process is to use beamforming to obtain the primary signal, i.e., beamforming the outputs of Microphone  1  and Microphone  3  (steered toward a side where the maximum acoustic energy is measured on one of Microphone  1  and Microphone  3 , while using a non-beamformed output of the microphone with the lower sound energy for the reference signal. 
     Yet another variation on this process is to use beamforming to obtain the reference signal, i.e., beamforming the outputs of Microphone  1  and Microphone  3  (steered toward a side where the minimum acoustic energy is measured on one of Microphone  1  and Microphone  3 , while using a non-beamformed output of the microphone with the maximum sound energy output for the primary signal. 
     In one non-limiting example referring to  FIG. 33 , a hypothetical use scenario exists when sound source  3320  is louder than sound source  3340 . In one or more embodiments, the system is designed to select Microphone  3  as the side to receive the primary signal. Receiving the primary signal can be accomplished by any one of the methods described directly above, such as beamforming Microphone  1  and Microphone  3  while placing a main response axis  3306  in a direction of sound source  3320 . Alternatively, an output from Microphone  3  can be used as the primary signal without beamforming. A reference signal can be obtained by beamforming Microphone  1  and Microphone  3  while placing a main response axis  3306  in a direction opposite to that of the sound source  3320 . Alternatively, an output from Microphone  1  can be used as the reference signal without beamforming. 
     In some embodiments, a system is implemented to sequence through the methods described above, e.g., beamforming to select a primary or reference signal verses using a non-beamformed output of a microphone for either the primary or the reference signal. A performance metric such as a signal-to-noise ratio difference between a primary and a reference signal for each method is computed and the method with the largest signal-to-noise ratio difference is the method that is used to process the signals from Microphone  1  and Microphone  3 . Sequencing through the methods can be performed at the onset of signal processing or the sequencing can be continuously performed to monitor a performance metric and then based on the evolution of the performance metric the method can be updated on the fly. Thus, many different methods are possible for use during the implementation of Configuration 3. The output of the noise cancellation system is then provided on one or more of speaker(s)  3350  to assist the user&#39;s hearing of the sound source  3320 . The speaker  3350  is incorporated into one or both side frames (temples) of the eyeglasses  3201 . 
     A similar process is implemented when a sound source  3340  is producing a larger sound energy  3342  on Microphone  1  relative to a sound energy level received on Microphone  3 . In such a case, the system can use a beamforming process to steer a main response axis of a microphone pair in a direction of the sound source  3340 . 
     The Microphone  1  and Microphone  3  pair helps the user to pick the stronger voice from around the user during a conversation, especially from the left and right side, by comparing the sound energies picked up from the Microphone  1  and Microphone  3 . During a group meeting or chatting, the speech signal may come from different directions (right or left sides) to the user. Configuration 3 compares the audio signal energy on each of the two microphones in order to determine which side the audio signal is coming from in order to help the user to focus on the active person speaking during the conversation. The output of the noise cancellation system is then provided on a speaker  3350  to assist the user&#39;s hearing of the sound source  3320  or  3340 . The speaker  3350  is incorporated into one or both side frames of the eyeglasses  3201 . 
     Configuration Switching and Scanning 
     In various embodiments, a system can be configured to switch between two, three, or more configurations. Scanning the configurations or scanning the different beams (or selected microphone pairs) formed from the array of microphones incorporated into a head wearable device can also be done automatically by the signal processing (hardware or combination of hardware and software) built into a head wearable device. Thus, in some embodiments, a system is implemented that scans through a number of directions relative to a user thereby forming beams (or processing selected microphone pairs) and providing assistance to a user with audio signals that have been received and improved by one or more of beamforming, noise cancellation, and or adjustment of volume before presentation to a user either locally or at a far side. 
     For example, while watching television and talking on the phone, a system can be configured to switch between Configuration 1 (phone call) and Configuration 2 (television viewing). A metric for switching to Configuration 1 (telephone function) can be related to detection of a change in sound energy on Microphone  0 . 
     Another example of configuration switching can be switching from Configuration 3 to Configuration 2 during a conversation. For example, in a meeting a person sitting to the right of a user wearing the eyeglasses  3201  begins speaking. Such a geometry is represented by the source  3320  outputting acoustic energy  3322  and an output of Microphone  3  being larger than an output from Microphone  1 . The system operates in Configuration 3 at this point. As the user listens and realizes that the speaker is to the right, the user might turn his or her head to the right to face the speaker. Now facing the speaker  3320 , the difference between the sound energy received on Microphone  1  and Microphone  3  has decreased, while the sound energy on Microphone  1  has increased. In such a situation the system switches to Configuration 2 as described above. 
     In one mode of operation, a user does not have to rotate his or her head from side-to-side to face a speaker in a meeting. As the active person speaking changes from position to position, for example, a position  3320  (on a right side relative to eyeglasses  3201 ) to a position  3340  (on a left side relative to eyeglasses  3201 ) to a position  3330  (in front of eyeglasses  3201 ) to a position  3380  (in back of eyeglasses  3201 ). The system will switch between microphone pairs and directions to select a primary microphone (either alone or a beamformed output) in the direction of the speaker and a reference microphone (either alone or a beamformed output) in the direction of the noise (mostly undesired audio). 
     Thus, embodiments of the invention are implemented by a system that switches between Configurations 1, 2, and 3 (or any subset thereof) operable by mechanical switching, audio switching, or by intelligent design operable through analysis of one or more performance metrics such as but not limited to maximum signal-to-noise ratio difference, maximum sound energy output from a microphone or a beamformed output, etc. 
     Three configurations, utilizing three or four microphones, have been described in conjunction with the figures above. Note that more than four microphones can be used with a head wearable device to provide a general number of n directions (axes) and potential configurations to process acoustic signals. Likewise, beamforming can be performed with more than two microphones. 
       FIG. 34  illustrates, generally at  3400 , processing acoustic signals from an array of microphones configured with a head wearable device, according to embodiments of the invention. With reference to  FIG. 34 , a process starts at a block  3402 . At a block  3404 , microphones that are part of an array of microphones that are attached to a head wearable device are scanned. The scanning includes analyzing the acoustic signals from the microphones for signal amplitude level and in some cases other parameters. At a block  3406 , a configuration is selected based on the scanning from the block  3404 . In some embodiments, selection logic is used to select between the configurations that are available utilizing a given array of microphones. At a block  3408 , the acoustic signals from the configuration selected at the block  3406  are processed to improve the acoustic signal. Improving the acoustic signal can include inputting the acoustic signals into a noise cancellation block to remove underside audio from a primary acoustic channel. Improving the acoustic signal can include amplifying the acoustic signal and presenting the amplified acoustic signal to a user of a head wearable device on a speaker incorporated with the head wearable device. The process stops at a block  3412 . 
     For purposes of discussing and understanding the embodiments of the invention, it is to be understood that various terms are used by those knowledgeable in the art to describe techniques and approaches. Furthermore, in the description, for purposes of explanation, numerous specific details are set forth in order to provide a thorough understanding of the present invention. It will be evident, however, to one of ordinary skill in the art that the present invention may be practiced without these specific details. In some instances, well-known structures and devices are shown in block diagram form, rather than in detail, in order to avoid obscuring the present invention. These embodiments are described in sufficient detail to enable those of ordinary skill in the art to practice the invention, and it is to be understood that other embodiments may be utilized and that logical, mechanical, electrical, and other changes may be made without departing from the scope of the present invention. 
     Some portions of the description may be presented in terms of algorithms and symbolic representations of operations on, for example, data bits within a computer memory. These algorithmic descriptions and representations are the means used by those of ordinary skill in the data processing arts to most effectively convey the substance of their work to others of ordinary skill in the art. An algorithm is here, and generally, conceived to be a self-consistent sequence of acts leading to a desired result. The acts are those requiring physical manipulations of physical quantities. Usually, though not necessarily, these quantities take the form of electrical or magnetic signals capable of being stored, transferred, combined, compared, and otherwise manipulated. It has proven convenient at times, principally for reasons of common usage, to refer to these signals as bits, values, elements, symbols, characters, terms, numbers, waveforms, data, time series or the like. 
     It should be borne in mind, however, that all of these and similar terms are to be associated with the appropriate physical quantities and are merely convenient labels applied to these quantities. Unless specifically stated otherwise as apparent from the discussion, it is appreciated that throughout the description, discussions utilizing terms such as “processing” or “computing” or “calculating” or “determining” or “displaying” or the like, can refer to the action and processes of a computer system, or similar electronic computing device, that manipulates and transforms data represented as physical (electronic) quantities within the computer system&#39;s registers and memories into other data similarly represented as physical quantities within the computer system memories or registers or other such information storage, transmission, or display devices. 
     An apparatus for performing the operations herein can implement the present invention. This apparatus may be specially constructed for the required purposes, or it may comprise a general-purpose computer, selectively activated or reconfigured by a computer program stored in the computer. Such a computer program may be stored in a computer readable storage medium, such as, but not limited to, any type of disk including floppy disks, hard disks, optical disks, compact disk read-only memories (CD-ROMs), and magnetic-optical disks, read-only memories (ROMs), random access memories (RAMs), electrically programmable read-only memories (EPROM)s, electrically erasable programmable read-only memories (EEPROMs), FLASH memories, magnetic or optical cards, etc., or any type of media suitable for storing electronic instructions either local to the computer or remote to the computer. 
     The algorithms and displays presented herein are not inherently related to any particular computer or other apparatus. Various general-purpose systems may be used with programs in accordance with the teachings herein, or it may prove convenient to construct more specialized apparatus to perform the required method. For example, any of the methods according to the present invention can be implemented in hard-wired circuitry, by programming a general-purpose processor, or by any combination of hardware and software. One of ordinary skill in the art will immediately appreciate that the invention can be practiced with computer system configurations other than those described, including hand-held devices, multiprocessor systems, microprocessor-based or programmable consumer electronics, digital signal processing (DSP) devices, network PCs, minicomputers, mainframe computers, and the like. The invention can also be practiced in distributed computing environments where tasks are performed by remote processing devices that are linked through a communications network. In other examples, embodiments of the invention as described above in  FIG. 1  through  FIG. 31  can be implemented using a system on a chip (SOC), a Bluetooth chip, a digital signal processing (DSP) chip, a codec with integrated circuits (ICs) or in other implementations of hardware and software. 
     The methods of the invention may be implemented using computer software. If written in a programming language conforming to a recognized standard, sequences of instructions designed to implement the methods can be compiled for execution on a variety of hardware platforms and for interface to a variety of operating systems. In addition, the present invention is not described with reference to any particular programming language. It will be appreciated that a variety of programming languages may be used to implement the teachings of the invention as described herein. Furthermore, it is common in the art to speak of software, in one form or another (e.g., program, procedure, application, driver, . . . ), as taking an action or causing a result. Such expressions are merely a shorthand way of saying that execution of the software by a computer causes the processor of the computer to perform an action or produce a result. 
     It is to be understood that various terms and techniques are used by those knowledgeable in the art to describe communications, protocols, applications, implementations, mechanisms, etc. One such technique is the description of an implementation of a technique in terms of an algorithm or mathematical expression. That is, while the technique may be, for example, implemented as executing code on a computer, the expression of that technique may be more aptly and succinctly conveyed and communicated as a formula, algorithm, mathematical expression, flow diagram or flow chart. Thus, one of ordinary skill in the art would recognize a block denoting A+B=C as an additive function whose implementation in hardware and/or software would take two inputs (A and B) and produce a summation output (C). Thus, the use of formula, algorithm, or mathematical expression as descriptions is to be understood as having a physical embodiment in at least hardware and/or software (such as a computer system in which the techniques of the present invention may be practiced as well as implemented as an embodiment). 
     Non-transitory machine-readable media is understood to include any mechanism for storing information in a form readable by a machine (e.g., a computer). For example, a machine-readable medium, synonymously referred to as a computer-readable medium, includes read only memory (ROM); random access memory (RAM); magnetic disk storage media; optical storage media; flash memory devices; except electrical, optical, acoustical or other forms of transmitting information via propagated signals (e.g., carrier waves, infrared signals, digital signals, etc.); etc. 
     As used in this description, “one embodiment” or “an embodiment” or similar phrases means that the feature(s) being described are included in at least one embodiment of the invention. References to “one embodiment” in this description do not necessarily refer to the same embodiment; however, neither are such embodiments mutually exclusive. Nor does “one embodiment” imply that there is but a single embodiment of the invention. For example, a feature, structure, act, etc. described in “one embodiment” may also be included in other embodiments. Thus, the invention may include a variety of combinations and/or integrations of the embodiments described herein. 
     Thus, embodiments of the invention can be used to reduce or eliminate undesired audio from acoustic systems that process and deliver desired audio. Some non-limiting examples of systems are, but are not limited to, use in short boom headsets, such as an audio headset for telephony suitable for enterprise call centers, industrial and general mobile usage, an in-line “ear buds” headset with an input line (wire, cable, or other connector), mounted on or within the frame of eyeglasses, a near-to-eye (NTE) headset display or headset computing device, a long boom headset for very noisy environments such as industrial, military, and aviation applications as well as a gooseneck desktop-style microphone which can be used to provide theater or symphony-hall type quality acoustics without the structural costs. Other embodiments of the invention are readily implemented in a head wearable device of general configuration such as but not limited to glasses, goggles, a visor, a head band, a helmet, etc. or the like. 
     While the invention has been described in terms of several embodiments, those of skill in the art will recognize that the invention is not limited to the embodiments described but can be practiced with modification and alteration within the spirit and scope of the appended claims. The description is thus to be regarded as illustrative instead of limiting.