Patent Publication Number: US-10332538-B1

Title: Method and system for speech enhancement using a remote microphone

Description:
An aspect of the disclosure here relates to acoustic signal processing. Other aspects are also described. 
     BACKGROUND 
     Hearing aids have a microphone, amplifier and a speaker. They pick up sound through the microphone, amplify the resulting acoustic signal and produce sound from the speaker, so that a hearing-impaired listener can have improved hearing. Even so, the wearer of a hearing aid may have trouble hearing speech from a distant talker. A separate wired or wireless microphone may be placed closer to the talker and can therefore more strongly pick up the talker&#39;s speech, but ambient noise may obscure the picked up speech or can make speech comprehension challenging. 
     SUMMARY 
     There exists a need for improvement in audio systems that are for listening to speech, when the listener is at a distance from the talker. A speech enhancement system for a remote microphone improves the listening experience for a listener when the talker is at a distance. One version of such a speech enhancement system has a wireless communications receiver, a first delay buffer, and a noise suppressor. The wireless communications receiver receives a wireless signal from a remote device that contains a microphone signal. The remote device has a first microphone that produces the first microphone signal. 
     The first delay buffer receives a second microphone signal from a second microphone. The second microphone is contained within a headset housing, e.g., an earbud housing or a hearing aid housing or other personal sound amplification product housing that may be worn near or against one or both ears of the listener.) The first delay buffer delays the second microphone signal by an adjustable delay. The adjustable delay is based on a difference between a wireless delay and an acoustic delay. 
     The noise suppressor produces an output audio signal for driving an earpiece speaker in the headset housing. The output audio signal produced by the noise suppressor is based on the first microphone signal and the adjustable delayed second microphone signal. 
     A method for speech enhancement is performed in a listening system, using a remote microphone, as follows. A wireless communications signal is received into a listening device. The wireless communications signal contains a microphone signal from a remote device having a first microphone. A second microphone signal, from a second microphone in the listening device, is received into a first delay buffer of the listening device. The second microphone signal is delayed through the first delay buffer whose delay is adjustable and is set based on a difference between a wireless delay and an acoustic delay. The first microphone signal, the adjustable delayed second microphone signal, or both are processed and modified through a noise suppressor. The noise suppressor produces an output audio signal for driving an earpiece speaker of the listening device. 
     The above summary does not include an exhaustive list of all aspects of the present invention. It is contemplated that the invention includes all systems and methods that can be practiced from all suitable combinations of the various aspects summarized above, as well as those disclosed in the Detailed Description below and particularly pointed out in the claims filed with the application. Such combinations have particular advantages not specifically recited in the above summary. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       Several aspects of the disclosure here are illustrated by way of example and not by way of limitation in the figures of the accompanying drawings in which like references indicate similar elements. It should be noted that references to “an” or “one” aspect in this disclosure are not necessarily to the same aspect, and they mean at least one. Also, in the interest of conciseness and reducing the total number of figures, a given figure may be used to illustrate the features of more than one aspect of the disclosure, and not all elements in the figure may be required for a given aspect. 
         FIG. 1  depicts remote listening by a wearer of a headset, through a remote microphone in an acoustic environment with noises. 
         FIG. 2  depicts remote listening using a listening device that adjusts two microphone signals on two paths to match their relative delay and signal strength for noise suppression. 
         FIG. 3  is a block diagram of an example remote listening system. 
         FIG. 4  depicts timing of events in an example operation of a remote listening system. 
         FIG. 5  depicts memory and computation efficient cross-correlation of audio signals, for use in delay estimation in the remote listening system. 
     
    
    
     DETAILED DESCRIPTION 
     Several aspects of the disclosure with reference to the appended drawings are now explained. Whenever the shapes, relative positions and other aspects of the parts described are not explicitly defined, the scope of the invention is not limited only to the parts shown, which are meant merely for the purpose of illustration. Also, while numerous details are set forth, it is understood that some aspects of the disclosure may be practiced without these details. In other instances, well-known circuits, structures, and techniques have not been shown in detail so as not to obscure the understanding of this description. 
     A remote listening system described herein provides improved hearing in a noisy acoustic environment and solves a technological problem of how to more accurately receive speech or other sound from a distance, using a remote device with a remote microphone, and a local listening device with a speaker. The remote device with remote microphone, which could be a smart phone, wireless microphone, wireless communication device or other wireless device in various versions, is placed close to a sound source such as a person talking, and transmits its microphone signal in real-time as a wireless audio signal (e.g., a radio frequency, RF, communications signal.) A listening device, which could for example be a wireless headset (e.g., a wireless earbud, or a wireless hearing aid), receives the wireless audio signal from the remote microphone, producing a localized remote microphone signal in the listening device. The listening device also has a local microphone, which receives an acoustic signal from the person talking or other sound source, producing a local microphone signal. 
     Delay matching is performed in the listening device, to align the localized remote microphone signal, which has a wireless delay due to transmission of the remote microphone signal over an RF communications link with the remote device, and the local microphone signal, which has an acoustic delay that is due to the acoustic path traveled by sound from the sound source. The time-aligned signals may then be matched for signal strength, through a gain adjustment that is determined when neither the distant talker nor the listener are speaking, and fed into a two channel noise suppressor. Output of the noise suppressor drives a speaker of the headset for the listener to hear the noise-reduced (or enhanced) speech of the talker (or a noise-reduced version of other sound that was picked up by the remote and local microphones.) 
     Some versions of the listening device use a memory and computation efficient cross-correlation and delay estimation technique, in which time slices of two audio signals are combined to reduce sample size in memory. In turn, this reduces the number of multiplications and amount of computation time for cross-correlation of the audio signals, thereby reducing the time needed for delay estimation and alignment of the audio signals, e.g., the local and remote microphone signals, prior to the noise reduction. 
       FIG. 1  depicts remote listening through a remote microphone  102  that is communicatively coupled to a headset  112 , in an acoustic environment with noises  104 ,  106 . By placing the remote microphone  102  close to the talker  108 , the signal-to-noise ratio (SNR) in the signal from that microphone  102  is increased, compared with the SNR at the ear of the user or listener  110 . In the scenario depicted in  FIG. 1 , the user of the headset  112  (the listener  110 ) and the audio source (the talker  108 ) are in the same acoustic environment (e.g., a room) at a distance D+d from each other, for example between 1 m and 10 m (within Bluetooth or other wireless communications connection range). The remote microphone  102  (and remote device  114 ) are placed in front of the talker  108  at a distance d much shorter than the distance D from the remote microphone  102  to the headset  112  user and listener  110 . One simplifying assumption that can be used in some versions is that d&lt;&lt;D. 
     If the environment contains ambient noise, for example Noise 1  106  and Noise 2  104 , e.g., sufficiently high levels of reverberation, the user, listener  110 , has difficulty in hearing the speech of the distant talker  108  when listening directly (i.e., without an electronic device). The remote microphone  102 , Mic1, the headset  112  or other listening device, and various aspects of the listening system described herein, overcome these difficulties. The remote microphone  102  is located in the remote device  114  such as a smart phone or other portable device equipped with wireless connectivity, which is capable of transmitting the microphone signal for example to a paired device such as the headset  112  using a Bluetooth connection or other radiofrequency (RF) connection. 
     Because of the Bluetooth or other wireless communications technology encoding and decoding process and transmission protocol, the received or “listening device version” of the audio signal from the remote microphone  102 , Mic1 will have a delay, herein referred to as wireless delay, relative to the original or “remote device version.” This wireless delay may be greater than the acoustic delay, which may be viewed as the time interval needed for the sound (represented or picked up in the local microphone signal from the local microphone  116 , Mic2 in the headset  112 ) to travel through an acoustic path from its acoustic source to the local microphone  116 . 
       FIG. 2  depicts remote listening through the use of the remote device  114 , and a listening device  222  that adjusts audio signals on two paths, to match delays and signal strength for noise suppression. The listener  110  hears speech (or other sounds) from a talker  108  at a distance, through the speaker  216  of the listening device  222 . Speech from the talker  108  is picked up by the microphone  102  of the remote device  114 , and transmitted through a wireless communications module  204  and antenna  202  of the remote device  114 . The transmitted microphone signal is then received through an antenna  224  and a wireless communications module  208  of the listening device  222 , with a wireless delay  220  that accounts for the encoding, transmitting, receiving, decoding, protocol and other delays in wireless communication until the speech appears for example at the output of the wireless module  208 . That same speech from the talker  108  is also picked up by the local microphone  116  of the listening device  222  and amplified through the audio codec  206 , experiencing an acoustic delay  218  that accounts for the distance and sound propagation from the talker  108  to for example the output of the audio codec  206  in the listening device  222 . 
     Speech signals from these two paths, also referred to here as acoustic and wireless paths, are presented to a delay match process  210  that determines a delay of one signal relative to the other, and adjusts signal timing until the two signals are matched in time. For example, the later arriving signal, which is on the wireless path as it is delayed as a result of the wireless delay  220 , may simply pass through the delay match process  210  with no further (deliberate) delays. The earlier arriving signal, which is on the acoustic path and is delayed as a result of acoustic delay  218 , is then adjustably delayed, for example through a delay buffer, to align in time with the later arriving signal, based on the determined difference between the wireless delay and the acoustic delay. 
     Now that the speech signals on the two paths are aligned in time, these signals are presented to a signal strength match process  212 , which performs gain adjustment to match the two signals in strength when neither the distant talker or the user/listener is speaking. For example, the gain of the adjustable delayed earlier arriving signal could be adjusted, or the gain of the later arriving signal could be adjusted, or both, until signal levels, signal power or other measurement of signal strengths match (while neither the distant talker nor the user/listener is speaking.) With the speech signals on the two paths aligned in both time and strength, the speech signals are presented to the input channels, respectively, of a two channel noise suppressor  214 . The noise suppressor  214  could adjust gain on one channel relative to the other, switch between channels, combine channels, subtract noise detected on one channel from the other channel and/or vice versa, reduce gain when no speech is detected, and/or perform other forms of noise suppression based on commonality or differences between the two channels, frequency domain analysis of signals, etc. in order to produce a single, noise reduced audio signal as its output. Output of the noise suppressor  214  is converted to sound through the speaker  216 , for the listener  110 , who as a result hears sound of the talker with speech enhancement, courtesy of the remote microphone  102  and listening device  222 . Various features of the listening device  222  of  FIG. 2  are implemented in the remote listening system shown in  FIG. 3 , and variations thereof. 
       FIG. 3  is a block diagram of an example remote listening system having speech enhancement blocks that perform a speech enhancement process as introduced above, to improve a listener&#39;s experience. In this version, the headset  112  (see  FIG. 1 ) is equipped with not just the local microphone  116 , Mic2, but also an accelerometer  118 , Acc that can contemporaneously pick up vibrations due to the talker&#39;s speech (e.g., via bone conduction.) It should also be noted that while  FIG. 3  shows all of the speech enhancement blocks or operations as being outside of the remote device  114  containing the remote microphone  102 , Mic1, an alternative is to configure a digital processor in the remote device  114  to perform some or all of those operations. Performing the speech enhancement signal processing on the headset  112  has the advantage of a lower delay due to not having to deliver the signal from the headset microphone (Mic2) to the remote device  114  (one-way only). In the version shown in  FIG. 3 , the speech enhancement digital signal processing takes place on the headset  112  (which also contains the headset Bluetooth decoder  304 , the finger tap detector  312 , the local microphone  116 , the accelerometer  118 , the volume  332 , the DRC  334  and the speaker  216 . 
     The speech enhancement signal processing in the listening device  222  (e.g., headset  112 ) in this particular example contains three inputs: 
     a) The remote microphone  102  input (Mic1). This signal contains an inherent relatively large delay, referred to herein as the wireless delay  220  (e.g., 50 ms) due to the wireless encoding, by for example a remote Bluetooth encoder  302 , decoding by the headset Bluetooth decoder  304 , and over the air transmission of the microphone signal to the headset  112 . Of course, wireless modules other than Bluetooth modules could be used.
 
b) The local microphone  116  input (Mic2), also referred to here as a headset microphone input. When the distant talker  108  speaks, the acoustic signal reaches the listener  110  (or the headset microphone worn by the listener) with an acoustic delay  218  (˜3 ms for 1 m to ˜30 ms for 10 m) due to the distance between the talker and the listener or user.
 
c) The accelerometer  118  input (Acc), also referred to here as a headset accelerometer input. This signal is active when the user speaks or taps the headset  112  (e.g., to indicate for example a request to calibrate the speech enhancement process or initiate the speech enhancement process described here).
 
     In some instances, the listening device  222  has a calibration phase in which
         the actual delay between the inputs a) and b) is estimated (before the delay match process), and   the noise level difference between the inputs a) and b) (when neither the distant talker nor the user is speaking, e.g., see  FIG. 4 , between t 5  and t 6 ) is estimated and compensated for (before the signal strength match process).       

     Components, timing, functionality and operation of the remote listening system of  FIG. 3  are further described below with reference to timing diagram  FIG. 4 . 
       FIG. 4  depicts timing of events in an example operation of a remote listening system. With reference also to  FIGS. 1-3 , the user of the headset  112  (listener  110 ) places the remote microphone  102  (e.g., placing remote device  114 ) in front of the distant talker  108  at time t 0  and moves back to a listening location. The listener  110  then initiates a calibration process, for example by double tapping the headset  112  at time t 1 . Once the double tap is detected by the accelerometer  118  and the Finger Tap Detector  312 , which monitors the accelerometer signal, the calibration process for delay estimation is enabled but does not actually start to process input data until the distant talker starts speaking while the user (listener) is not speaking—here star ling at t 4 . The timing diagram of  FIG. 4  shows that before t 4 , the user/listener begins speaking at t 2  (e.g., the user confirms by saying that he is ready for listening) followed by the distant talker joining the conversation at t 3 . Note however that the user&#39;s confirmation that he or she is ready to listen is optional (e.g., the user may simply wait until the distant talker begins the conversation.) 
     As seen in  FIG. 4 , the accelerometer signal (containing speech of the user/listener only) is passed to an energy-based voice activity detector  306 , VAD1, followed by a delay buffer  320 , Delay Buffer 2, set to the delay value Dt of the known wireless delay  220  between remote microphone  102  Mic 1 and headset  112 . This VAD1 output can be extended in time by a hangover window (e.g., 100-150 ms) to account for unvoiced consonants, in some versions. Then the output of VAD1 is inverted and applied to one of the inputs of a logic AND  322  block, AND1. 
     A second voice activity detector  308 , VAD2, detects when the distant talker speaks. This VAD2 has two inputs, the remote microphone signal from the headset Bluetooth decoder  304  or other wireless module  208  (see  FIG. 2 ), and the Mic2 signal delayed by a delay buffer  318 , Delay Buffer 1, set to the value Dt which is the known wireless delay  220  (e.g., including encoding, decoding and protocol) between remote microphone  102  and headset  112  (e.g., 50 ms). The VAD2 compares the energies/powers in the two channels in each small frame or block of, for example, 1 ms like the first VAD1. When the energy in input  1  is greater than in input  2  of the second voice activity detector  308 , it is assumed that the distant talker  108  speaks. 
     In the example logic shown in  FIG. 4 , the VAD2 output is combined with the VAD1 delayed-and-inverted output in the AND  322  logic block AND1. When the output of AND1 becomes 1, the delay estimation process  402  starts at t 4  (and is represented in  FIG. 3  by the dashed-line block, Delay Estimation  316 .) This delay estimation process  402  is described below and takes place for only a short duration between t 4  and t 5  when the distant talker  108  but not the user speaks (e.g., set to 1 or 2 s). 
     Still referring to  FIG. 3 , once the Delay Estimation  316  successfully estimates the actual delay (part of the calibration phase), it applies an adjustment to the first delay buffer  318 , Delay Buffer 1, and the second delay buffer  320 , Delay Buffer 2, by taking the outputs from these buffers  318 ,  320  at earlier locations Da instead of the initial location Dt (total buffer delay). The adjusted Da output of the Delay Buffer 1 is then passed to the gain multiplication block  328  and after that to the second input of the 2-channel Noise Suppressor  330 . 
     The adjusted Da output of the Delay Buffer 2 is then passed to an inverter and then to a second AND  324  logic block, AND2. When the delay estimation process  402  is complete, the Delay Estimation  316  block signals to the Gain Estimation  314  block at t 5  that it should start estimating the noise in its two input channels when the second AND  324  logic block (AND2) sends a triggering signal to do that. This is part of gain estimation process  404  performed between t 5  and t 6 . 
     A third voice activity detector  310  block (VAD3) makes a more accurate detection of when the distant talker speaks. This VAD3 has two inputs which are processed on short frames/blocks of, e.g., 1 ms each: the remote microphone signal from the headset Bluetooth decoder  304  and the Mic2 signal delayed by the adjusted delay Da from the Delay Buffer 1 after the delay estimation  402  is complete. 
     The output of VAD3 is then inverted and sent to the second AND  324  block AND2. When the output of AND2 becomes 1 at t 5 , for a short period of time (for example 200-500 ms), the noise powers in the two channels of the Gain Estimation  314  block are computed. Then, a gain to be applied to the delayed Mic2 by Da is computed from these two powers (by the gain estimation process  404 ) in such a way that when this gain is applied to the Mic2 (delayed by Da) the noise in this channel equals the noise in the remote microphone Mic1 from the headset Bluetooth decoder  304 . Then this gain is stored in the Gain Store  326  block at time t 6 . 
     Once the delay estimation process  402  and gain estimation process  404  (or delay estimation and gain estimation phases) are complete, the calibration phase may be deemed complete at which point the user (listener  110 ) can listen and talk to the distant talker  108  using the remote microphone  102  enhanced by the 2-channel noise suppressor  330 . The two channels of the Noise Suppressor  330  are synchronized in time and the signals are matched in order to suppress ambient noise. Some versions of the two-channel noise suppressor  330  use principles of a 2-channel noise suppressor as used in a mobile telephone. 
     After the noise suppressor  330 , a volume  332  block can allow the user to adjust the overall volume, in some versions. This volume  332  block can be followed in some versions by a dynamic range compressor block, DRC  334 , which amplifies small sounds and attenuates loud sounds. 
     At any point in time the user may re-start the calibration process if the noise conditions change or if the distance between the user and the distant talker  108  changes. However, the enhanced remote listening presented in this disclosure can also occur without a calibration phase, in some versions, by processing default Mic2 values of Dt for delay and 0 dB for gain. In this case the quality of the enhanced speech generated by the system may not be as good as when the calibration is performed. 
     The calibration process (operations performed between t 1  and t 6 ) can be repeated at any time such as when ambient noise distribution changes or the distance between talker and user changes. In other instances, the delay estimation &amp; adjustment is not performed, or the gain estimation &amp; adjustment is not performed. Also other methods to trigger the calibration process can be employed, such as to trigger automatically after a set time of a few seconds from the start of the listening session at t 0 . 
       FIG. 5  depicts a block diagram of a process for memory and computation efficient digital signal processing of audio signals, for use in for example the delay estimation  316  in the remote listening system of  FIG. 3 . The block diagram shows how a processor is configured according to various routines stored in memory to perform the prescribed digital signal processing operations on data structures stored in the memory. Before describing that process however, consider a standard approach to delay estimation which relies on a cross-correlation technique whereby the delay between two similar signals corresponds to the offset of a peak in the cross-correlation function of the two. For example, in the case of  FIG. 3 , when VAD1=1, this indicates that there is a segment of a User voice present in Mic2 input, and an acoustically similar segment will appear in the second input (the remote microphone Mic1 coming from the wireless link). In order to have a strong and distinctive peak in the cross-correlation measure between the Mic2 and Mic1 inputs, the segments need to be of some minimum length in order to find a similar acoustic signature in both. Typical lengths of around 1-2 seconds, which corresponds to the length of a short utterance, would give satisfactory results even in some noisy conditions. 
     In a real-time implementation on embedded platforms with limited memory resources, a conventional delay estimation technique would require significant storage space in memory. For example, if the calibration signal is 1 [sec] long, then that would require memory storage of ˜128 [kB] (assuming typical sampling rate of 16 kHz): 
     M1 array=1 [sec]=16000 [samples]=32000 [Bytes]=˜32 [kB] 
     M2 array=˜32 [kB] 
     Cross-Correlation Array=˜64 [kB] 
     An efficient mechanism and method are described in connection with  FIG. 5 , which reduce memory requirements for cross-correlation based delay measurement. In this method, it is not required to contiguously store the two signals of interest, before performing cross-correlation on them. Instead, each signal is partitioned into a sequence of short-segments, for example 100 msec long each, which are stacked on top of each other through a respective combiner  502   a ,  502   b , i.e. the stacking process  504  accumulates a sum of short segments in a small array  512 ,  514  (in this example, 3200 Bytes which corresponds to 100 msec @ 16 kHz @ 2 bytes/sample). When each audio signal, e.g., having a total length of 1 second (see  FIG. 5 ) is compacted in this manner into its own 100 msec array  512 ,  514 , then the cross-correlation is performed upon these two arrays  512 ,  514  of 100 msec, instead of upon the original 1 sec arrays. This corresponds to a 10 times reduction in the size of arrays. Then, the delay measurement is performed by the Delay Estimator  510 , also on the reduced arrays of, e.g., 100 msec, using known techniques such as cross-correlation or normalized cross-correlation (for example using a Generalized Cross-Correlation Phase Transform). 
     Using this stacking technique, the total memory requirement in the above example could be reduced from 128 kB to ˜12 kB, while at the same time maintaining the quality of having a strong and distinctive peak in the cross-correlation measure. A sequence of ten individual 100 msec sub-segments of the calibration utterance are added on top of each other into an input buffer of the Cross-Correlator  506 . The same process is done in parallel for both Mic1 and Mic2 signals. In this scenario, as depicted in  FIG. 5 , the audio signal  516  from the wireless remote microphone  102 , in the listening device  222 , is divided into 100 ms sub-segments and input into a combiner  502   a . Through the stacking process  504 , the sub-segments are combined, for example by adding signal values (e.g., amplitudes) at respective offsets into the segments, into a combined or merged segment  520  that is stored in the M1 Array  512 . 
     Similarly, the audio signal  518  from the local microphone  116  in the listening device  222  is divided into 100 ms sub-segments and input into a combiner  502   b . Again through the stacking process  504 , the sub-segments are combined into a combined or merged segment  522  that is stored in the M2 Array  514 . Other combination or merging processes are readily devised to produce combined segments that are smaller than the size of the original audio signal samples (sequence), and accordingly take less memory to store, and these can be implemented in other variations of the efficient cross-correlation process described here. 
     Cross-Correlator  506  cross-correlates contents of the M1 Array  512  and the M2 Array  514 , i.e., the combined or merged segment  520  and the combined or merged segment  522  are cross-correlated, and stores the resultant values in the cross-correlation array  508 . Because the stacking process  504  reduces the total number of values to be stored in the arrays  512 ,  514 , which in turn reduces the amount of data that is to be stored in the Cross-Correlation Array  508 , the overall cross-correlation process is more memory efficient than standard cross-correlation techniques. In addition, the Cross-Correlator  506  is computationally efficient, because a fewer number of data points is input into the cross-correlation algorithm, and a fewer number of multiplications are performed in the cross-correlation algorithm. 
     Based on the results of cross-correlation, a processor could determine whether two audio signals have a similar acoustic signature. If the cross-correlation method does not yield a strong and distinctive peak in its output, it would be an indication that the remote microphone (Mic1) is, for example, in a different room (different acoustic environment), and the system (e.g., Delay Estimator  510 ) may in that case decide to not perform the Delay Estimation/Adjustment operation in  FIG. 3 . 
     With ongoing reference to  FIGS. 3 and 4 , a Gain Estimation and Adjustment is performed after the Delay Estimation/Adjustment. After the delay between remote microphone Mic1 and the headset microphone Mic2 (where the latter was delayed by wireless delay Dt) is estimated and adjusted on Mic2, the estimation of the necessary gain to be applied on Mic2 can proceed at time t 5 . This condition will take place in a short set segment of say 200-500 ms when neither the user (listener) nor the distant talker speaks as controlled by the AND2 and VAD1 and VAD3 blocks. In such condition the Gain Estimation block measures the noise power levels in the two input signals ( 1  and  2 ). Then the difference between the two powers is converted in magnitude and it is stored in the Gain Store block and then applied to the Mic2 delayed by the adjusted delay Da. 
     This way the background noise levels in the two inputs ( 1  and  2 ) of the 2-channel Noise Suppressor are approximately the same and thus the Noise Suppressor is able to continuously estimate the noise and apply the necessary suppression on it. The Noise Suppressor applies suppression on both stationary and non-stationary ambient noises and during both periods of time when the distant talker is speaking or not speaking. 
     In an example scenario (see  FIGS. 1-4 ), the noise and speech spectra in Mic1 and Mic2 delayed by Da differ due to non-uniform distribution of ambient noise between the distant talker and the user/listener. A gain adjustment of, for example, −5 dB brings the two noise spectra closer to each other and thus allows the Noise Suppressor to suppress the ambient noises. 
     With reference to  FIGS. 1-5 , various components of the listening system described herein can be implemented in hardware, software executing on one or more processors, firmware, or combinations thereof. Two or more related components can be implemented as a single component with multiplexing, time-sharing, timeslicing or multithreaded operation, for example. Sections of logic can be implemented with equivalent logic or variations of logic as hardware, or in software or firmware, etc. 
     While certain aspects have been described and shown in the accompanying drawings, it is to be understood that such are merely illustrative of and not restrictive on the broad invention, and that the invention is not limited to the specific constructions and arrangements shown and described, since various other modifications may occur to those of ordinary skill in the art. For example, while  FIG. 4  depicts timing for a device in which double taps initiate calibration, it is also possible to have calibration initiated by recognition of a phrase, changes in other signals, etc. Also, one application of the memory and computation efficient cross-correlation technique is delay estimation, and where the delay of a delay buffer is set in accordance with the cross-correlation result; such an adjustment of timing or synching of the two audio signals is however but one specific system application of the technique. Other applications might not need such adjustment but rather would only need the delay estimation. The description is thus to be regarded as illustrative instead of limiting. 
     As described above, one aspect of the present technology may involve gathering and use of data available from various sources. The present disclosure contemplates that in some instances, this gathered data may include personal information data that uniquely identifies or can be used to contact or locate a specific person. Such personal information data can include demographic data, location-based data, telephone numbers, email addresses, twitter ID&#39;s, home addresses, data or records relating to a user&#39;s health or level of fitness (e.g., vital signs measurements, medication information, exercise information), date of birth, or any other identifying or personal information. 
     The present disclosure recognizes that the use of such personal information data, in the present technology, can be used to the benefit of users. For example, a user may wish to better hear personal information. Further, other uses for personal information data that benefit the user are also contemplated by the present disclosure. For instance, health and fitness data may be used to provide insights into a user&#39;s general wellness, or may be used as positive feedback to individuals using technology to pursue wellness goals. 
     The present disclosure contemplates that the entities responsible for the collection, analysis, disclosure, transfer, storage, or other use of such personal information data will comply with well-established privacy policies and/or privacy practices. In particular, such entities should implement and consistently use privacy policies and practices that are generally recognized as meeting or exceeding industry or governmental requirements for maintaining personal information data private and secure. Such policies should be easily accessible by users, and should be updated as the collection and/or use of data changes. Personal information from users should be collected for legitimate and reasonable uses of the entity and not shared or sold outside of those legitimate uses. Further, such collection/sharing should occur after receiving the informed consent of the users. Additionally, such entities should consider taking any needed steps for safeguarding and securing access to such personal information data and ensuring that others with access to the personal information data adhere to their privacy policies and procedures. Further, such entities can subject themselves to evaluation by third parties to certify their adherence to widely accepted privacy policies and practices. In addition, policies and practices should be adapted for the particular types of personal information data being collected and/or accessed and adapted to applicable laws and standards, including jurisdiction-specific considerations. For instance, in the US, collection of or access to certain health data may be governed by federal and/or state laws, such as the Health Insurance Portability and Accountability Act (HIPAA); whereas health data in other countries may be subject to other regulations and policies and should be handled accordingly. Hence different privacy practices should be maintained for different personal data types in each country. 
     Despite the foregoing, the present disclosure also contemplates instances in which users selectively block the use of, or access to, personal information data. That is, the present disclosure contemplates that hardware and/or software elements can be provided to prevent or block access to such personal information data. For example, the present technology can be configured to allow users to select to “opt in” or “opt out” of participation in the collection of personal information data during registration for services or anytime thereafter. In addition to providing “opt in” and “opt out” options, the present disclosure contemplates providing notifications relating to the access or use of personal information. For instance, a user may be notified upon downloading an app that their personal information data will be accessed and then reminded again just before personal information data is accessed by the app. 
     Moreover, it is the intent of the present disclosure that personal information data should be managed and handled in a way to minimize risks of unintentional or unauthorized access or use. Risk can be minimized by limiting the collection of data and deleting data once it is no longer needed. In addition, and when applicable, including in certain health related applications, data de-identification can be used to protect a user&#39;s privacy. De-identification may be facilitated, when appropriate, by removing specific identifiers (e.g., date of birth, etc.), controlling the amount or specificity of data stored (e.g., collecting location data a city level rather than at an address level), controlling how data is stored (e.g., aggregating data across users), and/or other methods. 
     Therefore, although the present disclosure broadly covers use of personal information data to implement one or more various disclosed aspects, the present disclosure also contemplates that the various embodiments can also be implemented without the need for accessing such personal information data. That is, the various aspects of the present technology are not rendered inoperable due to the lack of all or a portion of such personal information data. For example, live listening can take place based on non-personal information data or a bare minimum amount of personal information, such as the content being requested by the device associated with a user, other non-personal information available to the listener, or publicly available information. 
     To aid the Patent Office and any readers of any patent issued on this application in interpreting the claims appended hereto, Applicant wishes to note that they do not intend any of the appended claims or claim elements to invoke 35 U.S.C. 112(f) unless the words “means for” or “step for” are explicitly used in the particular claim.