Patent Publication Number: US-8989815-B2

Title: Far field noise suppression for telephony devices

Description:
FIELD OF THE INVENTION 
     The present invention relates generally to telephony devices and particularly to suppressing far field noise in telephony devices. 
     BACKGROUND 
     Telephones and conference units are commonly used for providing communication to near end and far end participants. These telephony devices include at least one microphone to capture the voice of the near end participants. Examples of such a microphone can be the microphone on a handset of a desktop telephone. In noisy environments, the microphone would pick up far field noise signals in addition to the voice signal, both of which signals get transmitted to the far end. This results in noisy voice signals being heard at the far end speaker. To overcome the noise signal, the near end speaker would have to speak louder so that the far end can hear his/her voice clearly over the noise signal. Clearly, this is inconvenient for the speaker. Furthermore, if the level of noise varies during the call, the speaker&#39;s voice may appear too loud or too low at the far end. Thus, the listening experience of far end listeners may be unsatisfactory. 
     One traditional solution to solving the problem of noise is to use a microphone array in a beamforming configuration. The microphones in the microphone array are arranged with a fixed distance between them. A signal processor coupled to the microphone array aims the audio beamforming in the direction of a speaker, providing directional sensitivity. As a result, sound from the speaker is emphasized, while noise from other directions surrounding the user is de-emphasized. Thus, the signal to noise ratio of the audio signal sent to the far end is improved. 
     In another solution, a reference microphone is used to capture stationary noise, which is then subtracted from the main microphone signal. Stationary noise is typically sensed over a long period of time (e.g., 1-2 s) by averaging the ambient noise signal generated by the reference microphone. The stationary noise signal is then subtracted from the main microphone signal using digital processing techniques. 
     The above mentioned techniques are used in several telephony applications for suppressing noise. One such application is a wireless headset, which, for example, uses a Bluetooth link to communicate with a communication device to provide hands free operation to the user. The wireless headset typically includes a microphone and a speaker that can be placed in close proximity with the user&#39;s mouth and ear respectively. The wireless headset can be affixed on or around an ear of the user so that the speaker is placed near the ear and the microphone extends to be close to the mouth. The wireless headset collects user&#39;s voice with the microphone and wirelessly transmits the voice signal to the communication device, which, in turn, transmits the voice signal to the far end. Furthermore, the communication device receives voice signals from the far end and wirelessly transmits the far end voice signals to the headset, which, in turn, reproduces the voice signal from the speaker. 
     The wireless headset can include one or more additional microphones to provide noise suppression using beamforming or stationary noise subtraction techniques described above. The noise suppression can be carried out at the headset itself or at the communication device. The additional microphone is typically permanently affixed to the headset, and therefore at a fixed distance from the headset microphone. 
     However, the inventors recognize a few drawbacks with the above techniques. Beamforming technique is less effective when the number of microphones in the microphone array is reduced. Because of cost and space considerations, mobile phones and wireless handsets can include only a small number of microphones—typically only two. As a result, the directionality of beamforming suffers by including a larger angle of sound sources. Consequently, the speaker&#39;s voice signal in addition to other sound source, many of them unwanted, located around the speaker are picked up by the microphones and sent to the far end. 
     Stationary noise cancellation techniques capture sound sources that are relatively constant over a large period of time. For example, sounds made by fans, machines, etc., which are repetitive can be effectively captured and subtracted using stationary noise sensing techniques. However, instantaneous noise, such as random ambient noise, people talking at a distance, background music, keyboard typing noise, etc. cannot be captured by stationary noise cancellation techniques. In some instances, the duration for which the reference microphone captures sound is reduced to allow capturing near instantaneous noise sources. However, even these techniques fail because the reference microphone signals, while including sounds from noise sources, also include the speaker&#39;s voice. Thus, when the reference microphone signals are subtracted from the main microphone signal, the subtraction can also remove some of the voice signal. Clearly, removing the signal of interest from the main microphone signal is undesirable. 
     The following disclosure addresses these and other drawbacks with noise cancellation and suppression in telephony devices. 
     SUMMARY 
     Noise suppression systems and methods presented herein suppress far field noise in a microphone signal. A near end communication system can include a main microphone for generating a main microphone signal. The main microphone signal can include voice of near end participants in addition to far field noise at the near end. The communication system can also include a reference microphone for generating a reference microphone signal. The reference microphone signal can be used for determining an estimate of the far field noise present in the main microphone signal or can be used to detect whether a local participant is currently talking. Telephony devices can include desktop telephones, conference units, mobile phones, etc. 
     In one example, the main microphone and the reference microphone can be located on the same device. Alternatively, the main microphone and the reference microphone can be located on separate devices connected by a communication link. For example, the main microphone can be located on a main device and the reference microphone can be located on a reference device. If the main device communicates with the far end, then the reference microphone signal can be transmitted over the communication link to the main device, where a processor (e.g., digital signal processor or DSP) can carry out noise suppression of the main microphone signal using the received reference microphone signal. Conversely, if the reference device communicates with the far end, then the main microphone signal can be transmitted over the communication link to the reference device, where the processor can carry out noise suppression of the received main microphone signal using the reference microphone signal. In both scenarios, the processor can insert time delays in the main microphone signal and/or the reference microphone signal such that their respective audio frames are time aligned. 
     In another example, the processor can choose to send the main microphone signal to a far end without noise suppression, if it determines that a latency of the main microphone signal due to inserted delays and/or noise suppression exceeds a predetermined limit. However, the processor can momentarily resume sending noise suppressed main microphone signals to the far end, despite high latency, during durations when the speaker at the near end is detected be in a state of monologue. 
     The DSP splits the main and reference microphone signals into subbands. In one approach, for each subband, the DSP determines a reference noise estimate of far field noise in the reference microphone signal. The DSP also determines a coupling estimate between the main and reference microphone signals. The DSP then determines an estimate of far field noise in the main microphone signal based on the reference noise estimate and the coupling estimate. The DSP then subtracts the far field noise estimate from the main microphone signal to produce a far field noise suppressed main microphone signal. 
     In another approach, for each subband, the DSP compares the levels of the main and reference microphone signal to determine whether a local participant is talking. If the local participant is detected not to be talking, the DSP can mute the main microphone signal, effectively suppressing any far field noise captured by the main microphone signal from being transmitted to the far end. When a local participant is detected to be talking, the DSP un-mutes the main microphone and allows the main microphone signal to be transmitted to the far end. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       Exemplary embodiments of the present invention will be more readily understood from reading the following description and by reference to the accompanying drawings, in which: 
         FIG. 1  shows an exemplary desktop telephone device having a handset microphone and a hands-free microphone, either of which can be used as a main microphone and a reference microphone; 
         FIG. 2  shows an exemplary conference unit having base-unit microphones and an extension microphone, either of which can be used as a main microphone and a reference microphone; 
         FIG. 3  shows an exemplary mobile telephone handset device having a main microphone and a reference microphone. 
         FIG. 4  depicts an exemplary block diagram of a communication device for suppressing far field noise; 
         FIG. 5  shows an example in which the main microphone and the reference microphone are located in separate devices, and in which the reference microphone is transmitted from one device to the other device over a communication link; 
         FIG. 6  shows another example of the main microphone and the reference microphones located in separate devices, of which one device is a conference unit; 
         FIG. 7  shows an exemplary block diagram for the examples of  FIGS. 5 and 6 ; 
         FIG. 8  illustrates a relative delay between the main and reference microphone signals; 
         FIG. 9  illustrates an exemplary digital signal processor having time delay blocks for aligning the main microphone and reference microphone signals; 
         FIG. 10  illustrates aligned main microphone and reference microphone signals by the processor of  FIG. 9 ; 
         FIG. 11  shows an exemplary digital signal processor having a monologue detector; 
         FIG. 12  shows another example in which the main microphone and the reference microphone are located on separate devices, but in which the main microphone signal, instead of the reference microphone signal, is transmitted from one device to the other device; 
         FIGS. 13 and 14  show two more examples similar to the example of  FIG. 12 , but in which only one of the two separate devices can communicate with the far end; 
         FIG. 15  shows an exemplary block diagram for the examples of  FIGS. 12-14 ; 
         FIG. 16  illustrates an exemplary signal flow diagram for suppressing far field noise in the main microphone signal; 
         FIG. 17  shows an exemplary flow chart for the steps executed by the digital signal processor for suppressing far field noise in the main microphone signal; 
         FIGS. 18A and 18B  shows an example of the difference between the spectrums of the main and reference microphone signals when the local talker is inactive and active; 
         FIG. 19  illustrates an exemplary block diagram for quantifying the difference between the main and reference microphone signals; 
         FIG. 20  shows an exemplary flowchart for steps executed by the digital signal processor for detecting activity of the local talker and suppressing far field noise; and 
         FIG. 21  compares spectral profiles of the main and reference microphone signals in response to an unvoiced or weak syllable to spectral profiles in response to high frequency percussive noise. 
     
    
    
     DETAILED DESCRIPTION 
       FIG. 1  discloses an exemplary device for carrying out noise cancellation. Desktop telephone  101  includes a base station  103  and a handset  102  interconnected by a cord  107 . Telephone  101  can be used by a near end user to communicate with one or more far end users. Telephone  101  can be a PSTN, VoIP, video phone, etc. Examples of telephone  101  can include products such as POLYCOM® SoundPoint® series, POLYCOM® VVX® series, etc. Telephone  101  can also include a hands-free microphone  106 , which is typically used by the user during hands-free operation. In hands-free operation, the user can replace the handset  102  on the base station  103  without disconnecting the call. While in hands-free operation, the telephone  101  can capture the users voice using the hands-free microphone  106  and reproduce the far end&#39;s audio signal via a speaker (not shown) on the base station. 
     In one embodiment, the telephone  101  can use the hands-free microphone  106  as a reference microphone while using the handset microphone  105  on the handset  102  as the main microphone. The main microphone  105  captures the near end audio signal, which includes the voice signal of the user in addition to the far field noise signal of the environment. Note that far field noise can be any noise generated at a distance of at least 6 inches from the main microphone. The reference microphone  105  is used to capture a reference signal, which, in turn, is used to suppress the far field noise signal in the near end audio signal captured by the main microphone  105 . After suppression, the near end audio signal is transmitted to the far end by the telephone  101 . 
     One advantage offered by the telephone  101  in noise suppression is that it does not require affixing additional microphones, such as the microphone array used in beamforming, to enable noise suppression. Instead, the telephone  101  uses the existing hands-free microphone  105  as a reference microphone. As a result, costs associated with adding additional reference microphones and the accompanying circuitry is avoided. 
     In another embodiment, the main microphone and the reference microphone can dynamically reverse roles. For example, referring to  FIG. 1 , the hands-free microphone  106 , which was previously acting as the reference microphone, can perform as a main microphone. Similarly, the handset microphone  105 , which was previously acting as the main microphone, can perform the function of the reference microphone. Which one of the two microphones performs what role can be based on, for example, the mode of operation of the telephone  101 , the relative distance of the microphones from the speaker, etc. 
     One can envision a scenario in which the telephone  101  is operating in the hands-free mode, i.e., the hands-free microphone  106  is being used to capture the user&#39;s voice. In such a scenario, the hands-free microphone  106  can be used as the main microphone, while the handset microphone  105  (whether on or off the cradle) can be used as the reference microphone. 
     In another scenario, while the telephone is operating in the hands-free mode, the user may move relative to the telephone  101  while talking. The user&#39;s motion may result in the handset microphone  105  being closer to the user than the hands-free microphone  106  during one time period but may also result in the opposite where the hands-free microphone  106  is closer to the user than the handset microphone. As will be discussed in detail below, it is advantageous to use the microphone that is closer to the user as the main microphone and user the other as the reference microphone. A measure of relative strength of the user&#39;s voice signal within the individual microphone signals can provide an indication of the user being closer to one of the two microphones. The telephone  101  can monitor the relative strength of the user&#39;s voice signal in signals received from both the hands-free microphone  106  and the handset microphone  105 . The microphone with the higher strength of user&#39;s voice signal can be selected as the main microphone signal, while the other can be selected as the reference microphone signal. 
       FIG. 2  shows a conference unit  110  typically used for audio conference calls. The conference unit  110  includes a base station  111  and an extension microphone  112  connected to the base station  111  via cord  113 . Examples of conference unit  110  can include products such as POLYCOM® SoundStation® series, POLYCOM® VoiceStation® series, etc. Typically, the base station  111  is placed at the center of a conference table, and the extension microphone  112  is extended to a position that is close to a speaker. Audio signals generated by the extension microphone  112  are transmitted to the base station via cord  113 . The base station  111  also includes a base microphones  114   a - c , which are typically used as an additional source of audio signal. However, for noise suppression, the conference unit  111  can use the extension microphone  112  as the main microphone for capturing the audio signal of the near end, and use any one of the base microphones  114   a - c  as a reference microphone for capturing a reference microphone signal. The reference microphone signal is used to suppress noise in the audio signal captured by the main microphone  111 . 
     Similar to the telephone  101 , conference unit  110  does not require affixing a microphone array to enable noise suppression. Instead, existing microphones that are typically used in normal operations are adapted as main and reference microphones. Here too, additional costs associated with microphone arrays are avoided. 
     Also similar to the telephone  101  of  FIG. 1 , the conference unit  110  can dynamically reverse the roles of the microphones from being the reference microphone to being the main microphone. For example, if user movement causes the user to be closer to one of the base microphones  114   a - c  than to the extension microphone  112 , then the conference unit  110  can switch to using one of the base microphones  114   a - c  as the main microphone and using the extension microphone  112  as the reference microphone. In some instances, one of the base microphones  114   a - c  could be used as the main microphone while another one of the base microphones  114   a - c  could be used as the reference microphone. 
     Another advantage offered by both the telephone  101  and the conference unit  110  is that the distance between the main microphone ( 105 / 112 ) and the reference microphone ( 106 / 114 ) is not fixed. This allows positioning the main microphone closer to the speaker than the reference microphone. With this arrangement, the proportion of voice signal in the audio signal captured by the reference microphone is much less than that in the audio signal captured by the main microphone. As will be discussed further below, this aspect allows the reference microphone to be able to cancel out the noise signal from the main microphone signal more effectively, without affecting the voice signal, and to allow the detection of active or inactive local talker. As an example, the reference microphone can be placed at a distance of at least four inches away from where the main microphone is located. In another example, the reference microphone can be placed at a distance that produces a reference microphone signal of at least 6 dB less than the main microphone signal when the local participant is talking. However, the reference microphone may not be placed so far away from the main microphone, that the far field noise captured by the reference microphone is no longer representative of the far field noise present in the main microphone signal. As such, the reference microphone signal may not be able to provide an adequate estimate of the noise in the main microphone signal. As one can imagine, using such a reference microphone signal my result in inadequate noise suppression or at worst may result in undesirable noise being added to the main microphone signal. As an example, reference microphone can be placed at no more than four-six feet from the main microphone. 
     In yet another example,  FIG. 3  shows the back side of an exemplary mobile phone or a wireless handset  130 . Handset  130  includes a reference microphone  131  that is used for far field noise suppression and a battery pack  133  removably affixed to the handset  130  by a battery hatch  132 . Handset  130  also includes a main microphone (not shown) for capturing the signal of interest, e.g., voice signal of the speaker, and is typically located at the bottom of the front side. Note that the location of the reference microphone  131  as shown in  FIG. 3  is only exemplary. The reference microphone  131  can be located anywhere on the handset  130 . In one example, the reference microphone  131  can be located as far as possible from the main microphone. The reference microphone signal can be used to estimate the far field noise present in the main microphone signal. Using this estimate, the far field noise in the main microphone signal can be suppressed. Note that while the prior art method of stationary noise suppression also subtracts an estimate of noise from the main microphone, it does so by measuring noise over a long period of time (e.g., 1-2 seconds). In contrast, the noise suppression methods (described in detail further below) used in the handset  130  carry out noise suppression in a near instantaneous fashion by suppressing noise every audio frame. As a result, short duration noise events and non-stationary noise, which would not be suppressed using stationary noise suppression method, would be advantageously suppressed using the noise suppression methods employed in handset  130 . 
     Discussion now turns to the description of exemplary functional block diagrams of noise suppression in telephone  101 , as shown in  FIG. 4  (similar block diagram can represent conference unit  110  of  FIG. 2 , and the handset of  FIG. 3 ). For simplicity, only those portions of telephone  101  that are relevant to noise suppression are disclosed in  FIG. 4 . Audio signals from both the main microphone  105  and the reference microphone  106  are amplified by amplifiers A 1   121  and A 2   122 , respectively. The audio signal generated by the main microphone  105  includes the voice signal of the speaker in addition to far field noise signal. The reference microphone signal  106  also includes the voice signal and the far field noise signal; however, the proportion of voice signal is much smaller than that in the audio signal of the main microphone  105 . As was described previously, the microphones  105  and  106  may also reverse roles, such that microphone  105 , which is currently shown as the main microphone, can assume the role of the reference microphone and the microphone  106 , which is currently shown as the reference microphone can assume the role of the main microphone. 
     Once amplified, the audio signals are sampled and digitized and fed to the digital signal processor (DSP)  125 . The audio signals are typically sampled at a given sampling rate and a given frame rate. For example, the audio signals may be sampled at a rate of 48 Ksamples/s with a frame rate of 10 ms. DSP  125  can be a microcontroller, a microprocessor, an application specific integrated circuit, a field programmable gate array, or a combination of one or more of the above. The DSP  125  can also include volatile memory such as RAM, and can be coupled to non-volatile memory such as Flash memory, ROM, etc. The DSP  125  can be configured to carry out operations such as noise suppression (explained in detail below), echo cancellation, filtering, mixing, amplification, encoding, decoding, compression, decompression, etc. One of the outputs of the DSP  125  can be a noise suppressed main microphone signal s 1 [n]. 
     The noise suppressed main microphone signal s 1 [n] can be outputted to a communication module  126 , which can transmit the signal to the far end via network  127 . The communication module  127  can include sub-modules for digital to analog conversion, amplification, modulation, network interfacing, etc. The communication module  126  can also receive communication signals such as far end microphone signals. Accordingly, the communication module  126  can additionally include sub modules for analog to digital conversion, demodulation, etc. that may be required for receiving such far end signals. 
     While the examples of  FIGS. 1-3  showed the main microphone and the reference microphone located on the same device, the following examples illustrate scenarios in which the main microphone and the reference microphone can be located on separate devices. 
       FIG. 5  shows an example in which the main microphone is located on main device  135  and the reference microphone is located on a separate reference device  140 . Main device  135  can be any telecommunication device that can transmit near end audio signals to a far end. For example, main device  135  can be a radio transceiver, a cellular phone, a wireless phone, smartphone, a personal digital assistant, etc. Reference device  140  can be any device that can capture sound through a microphone and can transmit the captured sound to the main device  135 . As an example, reference device  140  can also be a telecommunication device similar to main device  135 . Reference device  140  and main device  135  can communicate over a communication link  139 . Preferably, the communication link  139  can be a wireless link such as a radio frequency (RF) link, a Bluetooth® link, a Zigbee® link, WiFi link, etc. In some instances, communication link  139  may be, or additionally include, a serial link (e.g., USB, IEEE 1394), a parallel link, etc. Reference device  140  can capture sound using the reference microphone  141  and transmits the resultant reference audio signal to the main device  135  via link  139 . Main device  135  can use the reference microphone signal to suppress far field noise present in the audio signal captured using main microphone  136 . The noise suppressed audio signal can then be transmitted by the main device  135  to the far end. 
     It will be appreciated that by utilizing microphones of separate devices, noise suppression can be used on devices that do not have two microphones on the same device. Instead, a nearby reference device can be used to provide the reference microphone signal for carrying out noise suppression of the main microphone signal. Furthermore, the distance between the main microphone and the reference microphone can be flexible or variable, and not fixed as in the prior art. 
     Note that a handset  102  and base station  103  of the telephone  101  shown in  FIG. 1 , may not be considered as separate devices as the ones shown in  FIG. 5 . This is because the audio signal (represented by electric current or voltage) from the handset  102  is transferred to the base station  103  via a two wire link without any transformation. On the other hand, in the example shown in  FIG. 5 , the device  140  transforms the audio signal using transceivers to transfer the audio signal from one device to another. These transceivers can be wireless transceivers (e.g., Bluetooth, Zigbee, WiFi, etc.) or wired transceivers (e.g., USB, IEEE 1394, etc.) and use some protocol to transfer the audio signal. 
       FIG. 6  shows an example similar to the one shown in  FIG. 5 , but in which the main device  135  can be a conference unit  110 . The main microphone is located on the conference unit  110 , while the reference microphone is located on the reference device  140 . The Conference unit  110  and the reference device  140  can communicate wirelessly over communication link  139 . Any of the existing microphones  114   a - c  can be used as a main microphone to generate the main microphone signal. Typically, the conference unit  110  can process the generated main microphone signal and transmit it to the far end via network  127 . But, for noise suppression, the conference unit  110  can receive a reference microphone signal from the reference device  140  over communication link  139 , and use the received reference microphone signal to suppress far field noise present in the main microphone signal before transmitting it to the far end. 
       FIG. 7  shows an exemplary block diagram for the system described in  FIG. 5  (a similar block diagram can represent the system disclosed in  FIG. 6 ). As mentioned before, main device  135  and reference device  140  can communicate with each other over wireless link  139 . This communication can be accomplished via transceivers  142  and  143  in the main device  135  and the reference device  140 , respectively. The nature of the transceivers  142  and  143  would depend upon the nature of the communication link  139 . For example, if the communication link  139  is a Bluetooth link, then the transceivers  142  and  143  would be Bluetooth transceivers. In some cases the transceivers  142  and  143  may be capable of supporting more than one type of communication link. 
     The main audio signal generated by the main microphone  136  can be amplified by an amplifier  122  and digitized by an analog to digital converter  124  before being fed to the DSP  125 . The digitized main microphone signal is labeled as y 1 [n]. The reference audio signal generated by the reference microphone  141  can be amplified by an amplifier  121  and digitized by an analog to digital converter  123 . The transceiver  142  transmits the digitized reference microphone signal to the transceiver  143  in the main device  135  over communication link  139 . The output of the transceiver  143 , which output is the digitized reference microphone signal y 2 [n], can be fed to the DSP  125 . The DSP  125  can use the reference microphone signal y 2 [n] to suppress far field noise in the main microphone signal y 1 [n]. 
     The reference microphone signal and the main microphone signal may experience unequal delay paths to the input of the DSP  125 . For example, the reference microphone signal may experience more delay compared to the main microphone signal because the reference microphone signal has to additionally pass through the transceivers  142  and  143  before being fed to the DSP  125 . Such an exemplary scenario is illustrated in  FIG. 8 . Dotted line  401  can represent a timeline for capturing audio signals. For the sake of illustration, the timeline can be divided into a number of frames. One such frame, Frame-n, can begin at time t. Lines  402  and  403  can represent the timelines at the main microphone input and the reference microphone input of the DSP  125 . In other words, lines  402  and  403  aid in illustrating the amount of time it takes for the main microphone signal and the reference microphone signal to get from the outputs of their respective microphones to the DSP  125 . 
     Referring again to  FIG. 7 , the main microphone signal can be processed by the amplifier  122 , and the A/D converter  124  before being fed as y 1 [n] at an input of the DSP  125 . The delay associated with this processing is represented by tmain in  FIG. 8 . The reference microphone can also incur a similar delay due to processing by the amplifier  121  and the A/D converter  123 . However, the reference microphone signal can incur additional delay due to processing by the transceiver  142 , due to a transmission delay over link  139 , and due to processing by transceiver  143  before reaching the DSP  125 . The total delay of the reference microphone signal is represented by tref in  FIG. 8 . As evident from  FIG. 8 , Frame-n of the main microphone signal can arrive at the DSP  125  earlier than the corresponding Frame-n of the reference microphone signal. 
     The difference between the time of arrival of Frame-n associated with the main microphone signal and the reference microphone signal can be disadvantageous in some instances. This is because while the DSP  125  can receive Frame-n of the main microphone signal after time tmain, the corresponding Frame-n of the reference microphone signal still may not arrive. The only frames of the reference microphone signal that the DSP  125  may have already received are one or more frames previous to the Frame-n. Thus, if a far field noise event was confined during Frame-n, the mismatch of frames would not allow the DSP  125  to use the appropriate frame of the reference microphone signal to suppress far field noise in the main microphone signal. For all practical purposes, the algorithm can tolerate a small difference between the time of arrival of the main and reference microphones, as long as the difference is within 3 msec. 
       FIG. 9  shows one way for aligning and synchronizing frames of the main and reference microphone signals. DSP  125  can include delay blocks  410  and  411  for delaying input signals y 2 [n] and y 1 [n] by predetermined duration. Delay blocks  410  and  411  can be separate hardware delay blocks or can be part of pre-existing buffers within the DSP  125 . As such, the delay can be implemented even in software by way of a memory and a timer, which implementation is well known in the art. After being delayed, the main microphone signal y 1 [n] and reference microphone signal y 2 [n] are inputted to the noise suppression module  412  of the DSP  125 . 
       FIG. 10  illustrates an example in which the main microphone signal is delayed such that a frame from the main microphone signal aligns with the corresponding frame from the reference microphone signal. Main microphone signal can be delayed by delay block  411  by a duration td such that the sum of tmain and td are substantially equal to the delay tref incurred by the reference microphone signal. Thus, when the delayed main microphone signal is fed to the noise suppression module  412 , Frame-n of the main microphone signal can be aligned with Frame-n of the reference microphone signal. Of course, other frames of the main microphone and reference microphone signals will also be aligned. Although, the example of  FIG. 10  does not show the reference microphone signal being delayed, in some cases delaying the reference microphone may also be needed to achieve appropriate alignment of the frames of the main and reference microphone signals. 
     The delay inserted by the delay blocks  410  and  411  can be constant or programmable. In one example, the values of tmain and tref can be experimentally determined. If these values are determined to be relatively constant, then the delay blocks  410  and  411  can also be set to constant values. In another example, the DSP  125  may, at repeated intervals, determine the round trip time for packets between the two transceivers  142  and  143  (e.g., by sending ping packets). A change in the round trip time can indicate a change in total delay (tref) incurred by the reference microphone signal. Accordingly, the DSP can re-program the delay of delay blocks  410  and  411  to account for this change. 
     While delaying the main microphone signal aids in frame alignment, such delaying may add to a latency of the main microphone signal to reach the far end. Latency of the main microphone can be the sum of delays in the path of the main microphone signal from the microphone of the near end device to the loudspeaker of the far end device. In other words, the latency can be determined by the sum of delays at the near end, the transmission network, and the far end. For example, referring to  FIGS. 7 and 9 , the near end latency of the main microphone signal can be determined by summing individual latencies of the amplifier  122 , the A/D converter  124 , the delay block  411 , the noise suppression module  412 , additional processing module  413 , and the communication module  126 . The transmission network latency can be the latency of network  127 . The far end latency can be the latency of a far end communication module and of any pre-processing circuitry before the main microphone signal is delivered to a far end loudspeaker. A large latency can make two-way conversation between participants at the near end and the far end uncomfortable. Typically, the latency of the main microphone signal is preferably kept below 200 ms for a two way conversation. 
     If the delay due to the delay block  411  and the noise suppression module  412  pushes the latency of the main microphone signal over the preferable limit, the DSP  125  can choose not to carry out noise suppression of the main microphone signal. But, even two-way conversations can include durations during which one end can be said to be in a state of monologue. Generally, a monologue can be considered to be a substantially uninterrupted speech by the near end speaker for a given duration (e.g., few seconds to few minutes). Due to the one-way nature of a monologue, large latencies exceeding the preferable limit can go un-noticed at the far end. Therefore, DSP  215  can go back to carrying out noise suppression (including delaying the main microphone signal via delay block  411 ) when the near end is in a state of monologue. 
       FIG. 11  shows the DSP  125  having a monologue detector  422 . The DSP  125  can activate the monologue detector  422  only when it has been determined that the near end latency of the main microphone signals is above acceptable limits. Once activated, the monologue detector  422  can monitor both the main microphone signal y 1 [n] and one or more far end microphone signals  424  to detect whether the near end speaker has entered a monologue. If no monologue is detected, then the monologue detector can output a signal  423  to a multiplexer  421 , which selects the main microphone signal y 1 [n] (bypassing delay block  411  and noise suppression module  412 ) that has not been noise suppressed. On the other hand, if the monologue detector  422  detects a monologue, then it can output a signal  423  to the multiplexer  421 , which selects the noise suppressed main microphone signal s 1 [n]. If the DSP  125  determines that the near end latency is within acceptable limits, the monologue module  422  can be disabled and the control signal  423  can be set to a value such that the multiplexer selects noise suppressed main microphone signal s 1 [n] as its output. 
     Referring again to  FIG. 5 , one can see that it was the reference microphone signal that was transmitted from the reference device  140  to the main device  135 . This was done because it&#39;s the main device  135 , and not the reference device  140 , that is connected to the far end. However, the opposite scenario is also possible in which the reference device  140 , instead of main device  135 , is connected to the far end. In such a scenario, as shown in  FIG. 12 , the main microphone signal would have to be transmitted from the main device  135  to the reference device  140  over communication link  139 . The main microphone signal would be subsequently processed at the reference device  140  to carry out noise suppression and transmitted to the far end via network  127 . In one example, both the main device and the reference device, when not communicating with each other over the communication link  139 , can capture near end audio and separately and independently transmit the near end audio to one or more far end devices. 
     Another example, in which the main microphone signal is transmitted from the main device to the reference device, is depicted in  FIG. 13 .  FIG. 13  shows a wireless mobile phone  140  that can wirelessly communicate with a wireless headset  150  over wireless link  139 . Mobile phone  140  is typically used by the user for receiving and initiating phone calls with the far end over a mobile/cellular phone network (not shown). The user can listen to the received far end audio via speaker  148 , and speak into microphone  141 , which generates a near end audio signal to be sent to the far end. This, however, requires the user to use one hand to hold the mobile phone&#39;s  140  speaker  148  against the user&#39;s ear. The headset  150  alleviates the necessity of using the hand to hold the phone. Instead, the headset  150  fits around the users ear such that the speaker  151  is near the user&#39;s ear-canal and the microphone  152  extends out towards the user&#39;s mouth. The mobile phone  140  can conveniently stay in the user&#39;s pocket, on the desk, etc. Audio signals received by the mobile device  140 , which audio signals were previously sent to the speaker  148 , are now transmitted to the headset  150  over the communication link  139  for reproduction at speaker  151 . Similarly, user&#39;s voice is captured by the headset microphone  152  and transmitted to the mobile phone  140  over communication link  139 . Typically, when the headset  150  is operational, the mobile phone  140  disables its own speaker  148  and microphone  141 . 
     For noise suppression, the mobile phone  140  can activate its microphone  141  even when the headset  150  is operational. In such a scenario, the headset microphone  152  can capture the user&#39;s voice signal, and the mobile phone&#39;s  140  microphone  141  can be used to capture the far field noise. In other words, the headset microphone  152  can provide the main microphone signal and the mobile phone&#39;s microphone  141  can provide the reference microphone signal. 
     In normal operation, the main microphone signal from the headset  150  would be transmitted to the mobile phone  140 , which, in turn, would process the signal and transmit the processed signal to the far end. During noise suppression, however, the mobile phone  140  can use the reference microphone signal generated by the microphone  141  to suppress far field noise appearing in the main microphone signal received from the headset  150 . 
       FIG. 14  shows yet another example, in which the main microphone signal is captured by a cordless handset  149  and transmitted to the base station  144  over wireless link  139 . The base station  144  includes a reference microphone  145 , which captures a reference microphone signal. Thus, the cordless handset  149  can be considered to be the main device while the base station  144  can be considered to be the reference device. Under normal operation, the near end user may lift the cordless handset  149  from the cradle  146  of the base station  144 , and initiate communication with the far end. The main microphone signals captured by the main microphone  147 , as well as any user input on a user interface of the cordless handset  149 , can be wirelessly transmitted over wireless link  139  to the base station  144 . The wireless link  139  can be based on, for example, the Digital Enhanced Cordless Telecommunication (DECT) 6.0 standard. Once the main microphone signals is received at the base station  144 , the base station  144  carries out noise suppression using the reference microphone signal, and the noise suppressed main microphone signal can be sent to the far end. 
     One distinguishing feature between the scenarios of  FIGS. 13 and 14  and the scenario of  FIG. 12  is that while both the main and reference device of  FIG. 12  are capable of independently establishing communication to far end devices over network  127 , the headset  150  of  FIG. 13  and the cordless handset  149  of  FIG. 14  lack the ability to independently establish communication with a far end device. Instead they rely on the reference device (e.g., the mobile phone  140  and the base station  144 ) to establish such communication. Therefore, the far field noise suppression disclosed herein can be used where at least one of or both the main device and the reference device is or are able to independently communicate with a far end device. 
       FIG. 15  shows a generalized block diagram of the devices shown in  FIGS. 12 ,  13  and  14 . As discussed previously, the main microphone signal is captured at the main device  135  by main microphone  136 . The main microphone signal is amplified by amplifier  122 , digitized by A/D converter  124 , and transmitted to the reference device  140  via transceiver  143 , communication link  139 , and transceiver  142 . The reference microphone signal is captured at the reference device  140  by reference microphone  141 . The reference microphone signal is amplified by amplifier  121 , digitized by A/D converter  123  and fed to an input of the DSP  125 . Thus, the DSP  125  receives the main microphone signal y 1 [n] and the reference microphone signal y 2 [n] at its inputs. 
     The DSP  125  of  FIG. 15  can align frames from the main microphone signal and the reference microphone signal by delaying the reference microphone signal. This is because, unlike  FIG. 5 , in which the reference microphone signal lagged behind the main microphone signal, in  FIG. 15  it&#39;s actually the main microphone signal that lags behind the reference microphone signal. In other words, tmain would be greater than tref. As such, the configuration of the DSP  125  shown in  FIG. 9  can stay the same, except that the delay provided by the delay block  410  to the reference microphone signal is greater than that provided by delay block  411  to the main microphone signal. Specifically, the delay block  410  can delay the reference microphone signal by time td such that the sum of tref and td is equal to the time tmain—the delay associated with the main microphone signal. 
     The monologue detector  422  can also operate in a manner similar to that described previously with respect to  FIG. 5 , in that the monologue detector  422  can be enabled when it is determined that the latency of the main microphone signal is greater than an acceptable limit. However, for the scenario depicted in  FIG. 15 , the resulting improvement in latency of the main microphone signal would be limited to avoiding delays due to delay block  411  and the noise suppression module  412 . 
     Having discussed various configurations for main and reference microphones and the devices where they can be located, the discussion now turns to describing the far field noise suppression methods used by the DSP  125 . As was mentioned above, the DSP  125  can utilize at least two separate methods. In one method, the noise suppression is carried out by subtracting an estimate of the far field noise from the main microphone signal, which estimate is derived from the reference microphone signal. In the second method, the noise suppression is carried out by preventing transmission of the main microphone signal whenever the near end talker is not talking. 
       FIG. 16  shows a signal flow diagram of the first far field noise suppression scheme employed by the DSP  125 . As a first step, the main microphone signal y 1 [n]  201  and the reference microphone signal y 2 [n]  202  are passed through subband analysis filters  203  and  204  respectively. Subband analysis filter banks  203  and  204  can each include a set of filters arranged in parallel that split the input signal into a number of frequency subbands. The number of subbands is determined by the number of filters used in parallel. Preferably, the subband analysis filters  203  and  204  use a large number of filter banks to split the signals y 1 [n]  201  and y 2 [n]  202  into correspondingly large number of subbands. For example, subband analysis filters  203  and  204  can include 480 or more filters to produce 480 or more subbands. One advantage of having a large number of subbands is the simplification of downstream filters by way of requiring smaller number of taps. In other words, as the number of subbands increases, the number of taps in filters used downstream for processing signals within each subband decreases. Having a smaller number of taps results in faster filters. Again as an example, if the main microphone signal y 1 [n]  201 , which is sampled at 48 Ksamples/s with a 10 ms frame rate, is passed through the subband analysis filter  203  having 480 subbands, the signal y 1 [n]  201  would be split into 480 subbands with each subband having a single frequency domain coefficient. As a result, filters (e.g. finite impulse response or FIR filters) used downstream for processing within each subband would need to have only a single tap—resulting in very simple and fast processing. 
       FIG. 16  shows the output of the subband analysis filter  203  resulting in k subband signals Y 1 (1,m)  205  to Y 1 (k,m)  207 , where k is the subband index and m is the time index (or the frame index). This means that the main microphone signal y 1 [n]  201  is split into subband signals Y 1 (1,m)  205  to Y 1 (k,m)  207 . Similarly, subband analysis filter  204  splits reference microphone signal y 2 [n]  202  into Y 2 (1,m)  208  to Y 2 (k,m)  210  subband signals. While the output subband signals of subband analysis filters  203  and  204  are shown to be in the frequency domain, it is understood that, if necessary, the subband signals can be easily converted and operated in the time domain. 
     Subband signal outputs from both the subband analysis filters  203  and  204  are sent to the weight computation block  211 . Before getting into details of the weight computation block  211  (which are discussed below with reference to  FIG. 17 ), one can see that the outputs of the weight computation block  211 , weight subband signals W(1,m)  212  to W(k,m)  214 , are multiplied with corresponding main microphone subband signals Y 1 (1,m)  205  to Y 1 (k,m). The weight subband signals are a result of determination of an estimate of the noise signal present in the reference microphone subband signals (the reference noise estimate) and the determination of the estimate of coupling between the main microphone and the reference microphone subband signals. When multiplied with the main microphone subband signals, the weight subband signals suppress the far field noise signals estimated to be present in the main microphone subband signal. The resultant main microphone subband signals are noise suppressed. These noise suppressed main subband signals are then input to a subband synthesis block  218 , which combines the k subbands to produce a noise suppressed main microphone signal s 1 [n]  219 . 
     The generation of weight subband signals W(1,m)  212  to W(k,m)  214  is shown in further detail in  FIG. 17 . In step  301 , the DSP  125  estimates the magnitude of noise signal present in the main microphone signals Y 1 (1,m)  205  to Y 1 (k,m)  207  and in the reference microphone signals Y 2 (1,m)  208  to Y 2 (k,m)  210 . Noise estimate for a subband is typically determined by averaging the magnitude of the signal over a few frames when no voice signal is present. Noise estimates N 1 (k,m) and N 2 (k,m) are expressed by the following equations:
 
 N   1 ( k,m )= E{|Y   1 ( k,m )|}  (1)
 
 N   2 ( k,m )= E{|Y   2 ( k,m )|}  (2)
 
where E{ } is the average magniture of a signal over a few frames when no voice signal is active.
 
     Once the noise estimates for each subband for the main and reference microphone signals is determined, step  302  determines the signal to noise ratio (SNR) for each subband. The equations of SNR for each subband of the main and reference microphone signals is given below: 
     
       
         
           
             
               
                 
                   
                     
                       SNR 
                       1 
                     
                     ⁡ 
                     
                       ( 
                       
                         k 
                         , 
                         m 
                       
                       ) 
                     
                   
                   = 
                   
                     
                        
                       
                         
                           Y 
                           1 
                         
                         ⁡ 
                         
                           ( 
                           
                             k 
                             , 
                             m 
                           
                           ) 
                         
                       
                        
                     
                     
                       
                         N 
                         1 
                       
                       ⁡ 
                       
                         ( 
                         
                           k 
                           , 
                           m 
                         
                         ) 
                       
                     
                   
                 
               
               
                 
                   ( 
                   3 
                   ) 
                 
               
             
             
               
                 
                   
                     
                       SNR 
                       2 
                     
                     ⁡ 
                     
                       ( 
                       
                         k 
                         , 
                         m 
                       
                       ) 
                     
                   
                   = 
                   
                     
                        
                       
                         
                           Y 
                           2 
                         
                         ⁡ 
                         
                           ( 
                           
                             k 
                             , 
                             m 
                           
                           ) 
                         
                       
                        
                     
                     
                       
                         N 
                         2 
                       
                       ⁡ 
                       
                         ( 
                         
                           k 
                           , 
                           m 
                         
                         ) 
                       
                     
                   
                 
               
               
                 
                   ( 
                   4 
                   ) 
                 
               
             
           
         
       
     
     In the next step  303 , the DSP  125  determines a coupling estimate between the main microphone signal and the reference microphone signal in each subband. The coupling estimate α(k,m) is expressed by the following equation: 
     
       
         
           
             
               
                 
                   
                     α 
                     ⁡ 
                     
                       ( 
                       
                         k 
                         , 
                         m 
                       
                       ) 
                     
                   
                   = 
                   
                     
                       α 
                       ⁡ 
                       
                         ( 
                         
                           k 
                           , 
                           
                             m 
                             - 
                             1 
                           
                         
                         ) 
                       
                     
                     + 
                     
                       
                         β 
                         ⁡ 
                         
                           ( 
                           
                             k 
                             , 
                             m 
                           
                           ) 
                         
                       
                       ⁢ 
                       
                         
                           ( 
                           
                             
                                
                               
                                 
                                   Y 
                                   1 
                                 
                                 ⁡ 
                                 
                                   ( 
                                   
                                     k 
                                     , 
                                     m 
                                   
                                   ) 
                                 
                               
                                
                             
                             - 
                             
                               
                                 α 
                                 ⁡ 
                                 
                                   ( 
                                   
                                     k 
                                     , 
                                     
                                       m 
                                       - 
                                       1 
                                     
                                   
                                   ) 
                                 
                               
                               ⁢ 
                               
                                  
                                 
                                   
                                     Y 
                                     2 
                                   
                                   ⁡ 
                                   
                                     ( 
                                     
                                       k 
                                       , 
                                       m 
                                     
                                     ) 
                                   
                                 
                                  
                               
                             
                           
                           ) 
                         
                         
                           
                             
                               t 
                               1 
                             
                             ⁢ 
                             
                                
                               
                                 
                                   Y 
                                   2 
                                 
                                 ⁡ 
                                 
                                   ( 
                                   
                                     k 
                                     , 
                                     m 
                                   
                                   ) 
                                 
                               
                                
                             
                           
                           + 
                           
                             
                               ( 
                               
                                 
                                   t 
                                   1 
                                 
                                 - 
                                 1 
                               
                               ) 
                             
                             ⁢ 
                             
                                
                               
                                 
                                   Y 
                                   2 
                                 
                                 ⁡ 
                                 
                                   ( 
                                   
                                     k 
                                     , 
                                     
                                       m 
                                       - 
                                       1 
                                     
                                   
                                   ) 
                                 
                               
                                
                             
                           
                         
                       
                     
                   
                 
               
               
                 
                   ( 
                   5 
                   ) 
                 
               
             
             
               
                 
                   
                       
                   
                   ⁢ 
                   
                     where 
                     , 
                     
                       
 
                     
                     ⁢ 
                     
                         
                     
                     ⁢ 
                     
                       
                         β 
                         ⁡ 
                         
                           ( 
                           
                             k 
                             , 
                             m 
                           
                           ) 
                         
                       
                       = 
                       
                         min 
                         ⁡ 
                         
                           ( 
                           
                             1 
                             , 
                             
                               
                                 t 
                                 2 
                               
                               ⁢ 
                               
                                 
                                   
                                     SNR 
                                     2 
                                   
                                   ⁡ 
                                   
                                     ( 
                                     
                                       k 
                                       , 
                                       m 
                                     
                                     ) 
                                   
                                 
                                 
                                   
                                     SNR 
                                     1 
                                   
                                   ⁡ 
                                   
                                     ( 
                                     
                                       k 
                                       , 
                                       m 
                                     
                                     ) 
                                   
                                 
                               
                             
                           
                           ) 
                         
                       
                     
                   
                 
               
               
                 
                   ( 
                   6 
                   ) 
                 
               
             
           
         
       
     
     In Equation (5), α(k,m) denotes the coupling estimate between the main microphone signal Y 1 (k,m) and the reference microphone signal Y 2 (k,m) for each subband k. Generally, the coupling estimate can be a measure of the ratio of the magnitudes of the main microphone signal and the reference microphone signal. α(k,m) is adaptive, in that it modifies (if erroneous) the coupling estimate of the previous frame, denoted by α(k,m−1), to determine the coupling estimate for the current frame. For example, if the coupling estimate in the previous frame α(k,m−1) were precisely correct for the current frame, then the numerator (|Y 1 (k,m)|−α(k,m−1)|Y 2 (k,m)|) would be zero. As a result, the coupling estimate α(k,m) for the current frame would be equal to that determined for the previous frame. If however, the magnitudes of the main and reference microphone subband signal have sufficiently changed to alter the coupling, then the error (calculated by previously mentioned numerator) is normalized (by the denominator), multiplied by the step size denoted by β(k,m), and added or subtracted from the previous frame&#39;s coupling estimate. The step size β(k,m) represents the magnitude of correction being applied, and is typically less than or equal to 1. Larger the magnitude of correction, faster the coupling estimate adapts. β(k,m) is typically a function of the ratio of SNRs of the reference microphone subband and the main microphone subband signals. The DSP  125  can select a maximum possible step size by having t 2 =1, and minimum possible step size by having t 2 =0. 
     In step  304 , the DSP  125  determines the reference noise estimate for each of the reference microphone subband signals, Y 2 (k,m). The equation for the estimate, denoted by {tilde over (V)} 2 (k,m), is given below: 
     
       
         
           
             
               
                 
                   
                     
                       
                         V 
                         ~ 
                       
                       2 
                     
                     ⁡ 
                     
                       ( 
                       
                         k 
                         , 
                         m 
                       
                       ) 
                     
                   
                   = 
                   
                     
                       
                         ( 
                         
                           1 
                           - 
                           γ 
                         
                         ) 
                       
                       ⁢ 
                       
                         
                           
                             V 
                             ~ 
                           
                           2 
                         
                         ⁡ 
                         
                           ( 
                           
                             k 
                             , 
                             
                               m 
                               - 
                               1 
                             
                           
                           ) 
                         
                       
                     
                     + 
                     
                       γ 
                       ⁢ 
                       
                           
                       
                       ⁢ 
                       
                          
                         
                           
                             Y 
                             2 
                           
                           ⁡ 
                           
                             ( 
                             
                               k 
                               , 
                               m 
                             
                             ) 
                           
                         
                          
                       
                     
                   
                 
               
               
                 
                   ( 
                   7 
                   ) 
                 
               
             
             
               
                 
                   where 
                   , 
                   
                     
 
                   
                   ⁢ 
                   
                     γ 
                     = 
                     
                       
                         α 
                         2 
                       
                       ⁢ 
                       
                         min 
                         ⁡ 
                         
                           ( 
                           
                             1 
                             , 
                             
                               
                                 
                                   SNR 
                                   2 
                                 
                                 ⁡ 
                                 
                                   ( 
                                   
                                     k 
                                     , 
                                     m 
                                   
                                   ) 
                                 
                               
                               
                                 
                                   SNR 
                                   1 
                                 
                                 ⁡ 
                                 
                                   ( 
                                   
                                     k 
                                     , 
                                     m 
                                   
                                   ) 
                                 
                               
                             
                           
                           ) 
                         
                       
                     
                   
                 
               
               
                 
                   ( 
                   8 
                   ) 
                 
               
             
           
         
       
     
     In Equation (8) above, 0&lt;α 2 &lt;1 is the smoothing constant. As discussed previously, the far field noise suppression is carried out by determining the reference noise estimate, combining this reference noise estimate with the coupling estimate between the main microphone and the reference microphone to estimate the far field noise component in the main microphone, and finally subtracting the far field noise estimate from the main microphone signal. Equation (7) estimates one of the aforementioned quantities: the reference noise estimate. Generally, the reference noise estimate can be made equal to the magnitude of the reference microphone signal Y 2 (k,m) itself. But, there may be instances when the reference microphone may also include the signal of interest, e.g., the voice of the speaker. In such instances, the reference microphone signal cannot be directly equated to the reference noise estimate, because doing so would result in subtracting the voice signal from the main microphone signal. 
     Equation (7) employs variable γ to address this issue. As shown in Equation (8), γ is a function of the ratio of the SNR 2 (k,m) and SNR 1 (k,m), which are the signal to noise ratios for the reference microphone and the main microphone signals for each subband, respectively. In scenarios where the speaker is talking, SNR 2 (k,m) would be considerably smaller than SNR 1 (k,m) because of the relative proximity of the main microphone to the speaker. As a result, the ratio of SNR 2 (k,m) over SNR 1 (k,m) would be considerably less than 1. Assuming that the smoothing constant is equal to 1, γ would have a value that is considerably less than 1. Referring again to Equation (7), we see that γ is multiplied with the magnitude of the reference microphone signal, |Y 2 (k,m)|, and (1−γ) is multiplied with the estimate from the previous frame, {tilde over (V)} 1 (k,m−1). Because γ is considerably less than 1, the reference noise estimate {tilde over (V)} 1 (k,m) for the current frame m would essentially assume the estimate of the previous frame, and the contribution of the reference microphone signal for the current frame, which includes the voice signal, is desirably kept very small. 
     In scenarios where the speaker is not talking, the signal captured by the reference microphone signal is a good estimate of the far field noise. In such cases, the ratio of SNR 2 (k,m) over SNR 1 (k,m) would be approximately equal to 1. Again assuming that the smoothing constant α 2 =1, γ would be approximately equal to 1. As a result, the reference noise estimate would be largely composed of the magnitude of the reference microphone signal for the current frame, |Y 2 (k,m)|. 
     Note that {tilde over (V)} 1 (k,m) is updated every frame, and, as a result, estimates near instantaneous far field noise. This is in contrast with the prior art technique of measuring stationary noise, which technique measures noise over a long period of time (typically 1-2 seconds). Thus short-lived far field noise phenomenon, which would be ignored by the stationary noise technique, would be included in the reference noise estimate {tilde over (V)} 1 (k,m). Also, in contrast with the stationary noise cancellation technique, {tilde over (V)} 1 (k,m) monitors the presence of the voice signal, and automatically adapts such that the contribution of the voice signal on the reference noise estimate is minimal. 
     Once the estimates of coupling between the main microphone and the reference microphone and reference noise estimate for each subband are known, the DSP  125 , in step  305 , can compute the weight signals W(k,m) for each subband, as described by the following equation: 
                     W   ⁡     (     k   ,   m     )       =     1   -       α   ⁡     (     k   ,   m     )       ⁢           V   ~     2     ⁡     (     k   ,   m     )                Y   1     ⁡     (     k   ,   m     )                          (   9   )               
Where the reference noise estimate {tilde over (V)} 1 (k,m) is divided by |Y 1 (k,m)| for normalization. Referring again to  FIG. 16 , the weight signal for each subband is multiplied with the corresponding main microphone subband signal by multipliers  215 - 217 . The output of these multipliers results in Y 1 (k,m)−Y 1 (k,m)α(k,m){tilde over (V)} 2 (k,m)/|Y 1 (k,m)|. In other words, the estimate of far field noise in each subband of the main microphone, represented by the term Y 1 (k,m)α(k,m){tilde over (V)} 2 (k,m)/|Y 1 (k,m)|, is subtracted from the main microphone subband signal Y 1 (k,m), resulting in a noise suppressed main microphone signal S 1 (k,m).
 
     The noise suppressed main microphone subband signals S 1 (1,m)  220 , S 1 (2,m)  221  . . . , S 1 (k,m)  222  are subband synthesized by the subband synthesis filter  218 . Output of the subband synthesis filter  218  is a time domain, noise suppressed, main microphone signal s 1 [n]  219 . 
     The second method of suppressing far field noise in the main microphone signal is now explained. As mentioned above, the second method—local talk detection—suppresses far field noise in the main microphone signal by suppressing the main microphone signal itself when no local talk is detected. At other times, i.e., when the local participant is talking, no suppression is carried out. During the time when no suppression is carried out, the DSP  125  may continue to process the main microphone signal as it normally would for transmission to the far end. 
     The local talk detection can be carried out using both the main microphone signal and the reference microphone signal. One such way of local talk detection is shown in  FIGS. 18A and 18B , which show spectral levels of the main and reference microphone signals when the local talker is inactive and active, respectively. Plot  451  shows the spectral characteristics of the main  453  and reference  454  microphone signals that have been collected simultaneously for one time frame (typically 10 ms). The vertical axis of the plot  451  can represent amplitude, power, energy, etc. of the main  453  and reference  454  microphone signals for various frequencies. The spectral levels of the main  453  and reference  454  microphone signals remain relatively same for majority of the spectrum when the local talker is inactive. 
     But when the local talker is active, the relative spectral levels of the main and reference microphone signals are no longer the same. This can be seen in  FIG. 18B , in which plot  452  shows that there is appreciable difference between the spectral levels of the main microphone signal  455  and the reference microphone signal  456 . Measurements have shown that the difference is in the range of 6 dB-10 dB across the spectrum. This difference is primarily due to the closer proximity of the local talker&#39;s mouth to the main microphone than to the reference microphone. As will be discussed below, this difference can be used to determine a threshold value over which the DSP  125  can be certain that the local talker is active. 
     It will be appreciated that comparing relative spectral levels of the main and reference microphone signals is only one exemplary approach to local talk detection, and that other methods, such as comparing time domain amplitude levels, comparing transformed (e.g., Fourier transform) main and reference microphone levels, etc., may also be used. 
     As previously discussed with respect to  FIGS. 18A and 18B , there is appreciable difference in spectral levels of the main and reference microphone signals when the local talker is active.  FIG. 19  illustrates one approach in quantifying this difference. The main microphone signal y 1 [n]  201  and the reference microphone signal y 2 [n] are fed to subband analysis blocks  203  and  204 , respectively. The subband analysis blocks  203  and  204  have been previously discussed with reference to  FIG. 16 , and are therefore not discussed further. It is understood that the subband analysis blocks  203  and  204  can split their respective input signals into k subbands. For example, the subband analysis block  203  splits the main microphone signal y 1 [n]  201  into subband signals Y 1 (1,m)  205  to Y 1 (k−1,m)  230  and Y 1 (k,m)  207 . Similarly, the subband analysis block  204  splits the reference microphone signal y 2 [n] into subband signals Y 2 (1,m)  208  to Y 2 (k−1,m)  209  and Y 1 (k,m)  210 . As mentioned before, k is the subband index while m is the time index (or the frame index). 
     The example of  FIG. 19  quantifies the difference between the main and reference microphone signals by summing the ratios of their amplitude (or energy, power, etc.) levels within each subband. This sum, S(m), is determined by the sum block  231 . As shown in  FIG. 19 , the sum S(m) is expressed as: 
     
       
         
           
             
               
                 
                   
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     It will be appreciated that other approaches to quantifying the difference between the main and the reference microphone signals can also be taken. For example, instead of taking the ratio of the levels of the main and the reference microphone signal for each subband, one can take the difference between them. In another approach, instead of summing the ratios across all subbands, one can take an average of all the ratios. 
     In some instances a double talk detector can be adapted to determine the sum S(m). Double talk detectors are commonly used in telecommunication signal processing to aid echo cancellation. The double talk detector detects simultaneous near end and far end speech. One exemplary approach to determining double talk is shown in equation (11) below: 
                     D   ⁡     (   m   )       =       ∑     i   =   1     k     ⁢           ⁢              Y   mic     ⁡     (     i   ,   m     )                       Y   spk     ⁡     (     i   ,   m     )            ⁢     ERL   ⁡     (   i   )                     (   11   )               
Where |Y mic (i,m)| and |Y spk (i,m)| are the energy or amplitude levels of the main microphone signal and the speaker signal in the ith subband and the mth audio frame. ERL(i) is the echo return loss estimate of the ith subband. The double talk detector declares a double talk state when the value of D(m) exceeds a predefined threshold. If the Y spk (i,m) were to be replaced by the reference microphone signal Y 2 (i,m), and the ERL(i) were to be equated to unity, D(m) would be the same as S(m) of equation (10). Therefore, hardware or program code or both for double talk detector already present in the DSP  125  can be reused to also determine the sum S(m).
 
       FIG. 20  shows a flowchart illustrating the steps performed by the DSP  125  in suppressing far field noise. The DSP  125  determines the sum S(m), described above, for each audio frame m (step  311 ). After determining the sum S(m), the DSP  125  compares the sum to a threshold value Th (step  312 ). If the sum S(m) exceeds this threshold value, then the local talker can be considered to be active; if however the sum S(m) is less than the threshold value Th, then the local talker can be considered to be inactive. The threshold value Th can be experimentally determined. For example, referring again to  FIG. 18B , it was observed that when the local talker is active, the difference between the main and the reference microphone signals is in the range of 6 dB-10 dB across the spectrum. Therefore, one exemplary threshold value Th can be 10 dB multiplied by the total number of subbands k. In this case, because the threshold value Th is being expressed in dB, the comparison step  312  can compare 10 log 10 S(m) to the threshold value of k×10 dB. 
     If the comparison step  312  determines that the sum S(m) is not greater than the threshold Th, then the local talker is considered to be inactive. Subsequently, in step  313 , the DSP  125  can mute the main microphone signal. As a result, any far field noise (stationary or non-stationary) captured by the main microphone would be prevented from being transmitted to the far end. It is understood that the DSP  125  can suppress or interrupt the transmitting of far field noise captured by the main microphone to the far end by using approaches other than simply muting the main microphone. For example, the DSP  125  may instead choose to keep the main microphone on, but attenuate the main microphone signal to such an extent that any far field noise is inaudible at the far end. In another example, the DSP  125  may transmit comfort noise or white noise to the far end in place of the main microphone signal or in addition to an already attenuated main microphone signal. 
     If, however, the comparison step  312  determines that the sum S(m) is greater than the threshold Th, then the local talker is considered to be active. Subsequently, in step  314 , the DSP  125  can un-mute the main microphone to allow the main microphone signal to be transmitted to the far end. 
     In some instances, due to the nature of the local talker&#39;s speech, the DSP  125  may inadvertently switch the state of the main microphone from un-muted to mute. To prevent inadvertent muting, and the resultant choppiness, the DSP  125  can introduce a period of time, say 200 ms, after it un-mutes the main microphone, during which period the DSP  125  will not mute the main microphone even if the comparison in step  312  indicates that local talker is inactive. Furthermore, to make the detection of either the active or inactive state more robust, the DSP  125  may require the result of the comparison step  312  to be the same for at least n number of consecutive frames (e.g., 3 consecutive frames) before determining to mute or un-mute the main microphone signal. 
     In some instances, the DSP  125  may not be able to detect, or detect too late, the onset of speech by the local talker. For example, some words such as “six” have weak or unvoiced first syllables, and if the speech begins with such words, the DSP  125  may not detect that the local talker is active until after later occurring stronger syllables have been voiced. This means that the far end would receive the local talker&#39;s speech with first few syllables missing. One reason the DSP may be unable to detect the weak syllables is because their spectral energy lies in high frequency bands (above 3 KHz), which are typically higher than the range of frequencies considered while determining the sum S(m). Of course, one way to account for these syllables can be to extend the subbands over which the sum S(m) is determined to higher frequencies. 
     For example, plot  461  of  FIG. 21  shows the main microphone spectrum  463  and the reference microphone spectrum  464  in response to the spoken word “six.” Appreciable energy difference between the main and reference microphone signal spectrums appear at frequencies greater than 2.5 KHz to 3 KHz. Therefore, including higher frequency bands in the determination of the sum S(m) may help in detecting the onset of weak syllables. However, as shown in plot  462 , percussive noise produces main and reference microphone signal spectra similar to the one shown in plot  461 . Therefore, by merely including higher frequency bands in the determination of sum S(m) may render the DSP  125  to falsely detect the local talker as active in the presence of percussive noise. 
     One solution to this problem is to modify the way the sum S(m) is calculated, such that levels of amplitude (or energy, power, etc.) are summed differently in different frequency ranges. Thus, even though the sum S(m) is determined over all available frequency bands, including the higher frequency bands, the contribution of each frequency band to the sum can be different based on the position of the frequency band within the spectrum. In one example, the sum S(m) is calculated as follows: 
     For the frequency region between 0-3 kHz, the ratios of main and reference microphone levels for each subband are summed to determine S(m). 
     For the frequency range between 3-4 kHz, the sum of two energy values of the main and reference microphone are divided and summed to determine S(m). 
     For the frequency range between 4-5.5 kHz, the sum of four energy values of the main and reference microphone are divided and summed to determine S(m). 
     For the frequency range between 5.5-8 kHz, the sum of eight energy values of the main and reference microphone are divided and summed to determine S(m). 
     With S(m) calculated as shown above, the DSP  125  can not only detect onset of weak syllables, but also distinguish the weak syllables from high frequency percussive noise. 
     While the signals produced by the subband analysis filter banks have been shown to be in the frequency domain, it is understood that the signals can also be converted into time domain and the processing carried out in time domain. The conversion from frequency domain to time domain is well known. 
     The above description is illustrative and not restrictive. Many variations of the invention will become apparent to those skilled in the art upon review of this disclosure. The scope of the invention should therefore be determined not with reference to the above description, but instead with reference to the appended claims along with their full scope of equivalents.