Abstract:
A first microphone is operated in a low power sensing mode, and a buffer at the first microphone is used to temporarily store at least some of the phrase. Subsequently the first microphone is deactivated, then the first microphone is re-activated to operate in normal operating mode where the buffer is no longer used to store the phrase. The first microphone forms first data that does not include the entire phrase. A second microphone is maintained in a deactivated mode until the trigger portion is detected in the first data, and when the trigger portion is detected, the second microphone is caused to operate in normal operating mode where no buffer is used. The second microphone forms second data that does not include the entire phrase. A first electronic representation of the phrase as received at the first microphone and a second electronic representation of the phrase as received at the second microphone are formed from selected portions of the first data and the second data.

Description:
CROSS-REFERENCE TO RELATED APPLICATION 
       [0001]    This patent claims benefit under 35 U.S.C. §119(e) to U.S. Provisional Application No. 62/115,898 entitled “Audio Buffer Catch-up Apparatus and Method with Two Microphones” filed Feb. 13, 2015, the content of which is incorporated herein by reference in its entirety. 
     
    
     TECHNICAL FIELD 
       [0002]    This application relates to microphones and, more specifically, to approaches for operating these microphones. 
       BACKGROUND 
       [0003]    Microphones are used to obtain a voice signal from a speaker. Once obtained, the signal can be processed in a number of different ways. A wide variety of functions can be provided by today&#39;s microphones and they can interface with and utilize a variety of different algorithms. 
         [0004]    Voice triggering, for example, as used in mobile systems is an increasingly popular feature that customers wish to use. For example, a user may wish to speak commands into a mobile device and have the device react in response to the commands. In these cases, a voice activity detector may first detect whether there is voice in an audio signal captured by a microphone, and then, subsequently, analysis is performed on the signal to predict what the spoken word was in the received audio signal. Various voice activity detection (VAD) approaches have been developed and deployed in various types of devices such as cellular phones and personal computers. 
         [0005]    Microphones that are always on are often equipped with internal oscillators and operate at very low power. Low power microphones are used in various applications and sometimes two or more microphones are used when the device is brought out of the low power mode. Although the low power aspect allows some of the microphones to be on all the time in a low power listening mode, the microphones may also use buffers to aid in voice activity detection, which introduce processing delays. The processing delays may cause problems at the far end of the system where the signals frequently need to be processed as quickly as possible. 
         [0006]    The problems of previous approaches have resulted in some user dissatisfaction with these previous approaches. 
     
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         [0007]    For a more complete understanding of the disclosure, reference should be made to the following detailed description and accompanying drawings wherein: 
           [0008]      FIG. 1  comprises a system using two microphones that implements an audio buffer catch-up approach; 
           [0009]      FIG. 2  comprises a flowchart showing an audio buffer catch-up approach; 
           [0010]      FIG. 3  comprises a time line and flowchart showing one example of an audio buffer catch-up approach; 
           [0011]      FIG. 4  comprises the time line showing an example of an audio buffer catch-up approach. 
       
    
    
       [0012]    Skilled artisans will appreciate that elements in the figures are illustrated for simplicity and clarity. It will further be appreciated that certain actions and/or steps may be described or depicted in a particular order of occurrence while those skilled in the art will understand that such specificity with respect to sequence is not actually required. It will also be understood that the terms and expressions used herein have the ordinary meaning as is accorded to such terms and expressions with respect to their corresponding respective areas of inquiry and study except where specific meanings have otherwise been set forth herein. 
       DETAILED DESCRIPTION 
       [0013]    The present approaches utilize two (or potentially more) microphones to obtain speech phrases from an utterance of a speaker. The effects of delays caused by buffers in one of the microphones are significantly reduced or eliminated. The approaches described herein are easy to implement and eliminate problems and limitations associated with prior approaches. 
         [0014]    Referring now to  FIG. 1 , one example of a system  100  for audio buffer catch-up is described. The system  100  includes a first microphone  102  (a microphone with a built in voice activity detector, manufactured by Knowles, Inc., in one example), a second microphone  104  (a pulse density modulation (PDM) microphone, in one example), and a processor  106 . The processor  106  may include a codec  108  and an application processor  110 . 
         [0015]    The first microphone  102  and second microphone  104  may be micro electro mechanical system (MEMS) microphones. In one example, these microphones are assemblies including a sensing element (diaphragm and back plate) and an application specific integrated circuit (which includes a buffer in the case of microphone  102  and potentially performs other processing functions). Sound energy is received by the microphones, moves the diaphragms and produces an electrical signal (which may or may not be buffered). 
         [0016]    The processing device  106  may include a codec  108  and an application processor  110 . The codec  108  in this example may supply the clock signals to the microphones  102  and  104 , and may perform other signal processing functions. The application processor  110  may also perform processing related to the device in which the microphones  102  and  104  are deployed. For example, if the microphones  102  and  104  are deployed in a cellular phone, the application processor  110  may perform processing associated with the cellular phone. Although both a codec  108  and an application processor  110  are shown here, it will be appreciated that these devices can be merged together into a single processing device. 
         [0017]    A clock signal  112  is applied to the microphones. Application of the clock signal, when applying the power signal to the microphone, causes the first microphone  102  to operate in a normal operating mode where incoming data to the ASIC is not buffered, but passed through to the output of the microphone  102 . Non-application of the clock signal after power has been applied to the microphone, causes the first microphone  102  to operate in a low power operating mode. In this mode, incoming data to the ASIC is buffered and not directly passed through to the output of the microphone  102 , thereby introducing a buffering delay, in one example of 256 milliseconds. The clock signal may be applied at one frequency when the microphone is in low power mode after acoustic activity has been detected and may be applied at the same or different frequency in the normal operating mode. 
         [0018]    In one example of the operation of the system of  FIG. 1 , the first microphone  102  is on and uses its buffer (i.e., incoming audio data must pass through the buffer). The second microphone  104  is off. Incomplete segments of a phrase (e.g., OK GOOGLE NOW, WHAT IS THE WEATHER TODAY?) are received by at least one, and in some embodiments both, microphones  102  and  104 . When the processing device  106  detects speech in data provided by the first microphone, the second microphone  104  is activated and begins to provide unbuffered (real time) data to the processing device. The second microphone may be activated by applying the clock signal  112  to the second microphone  104 . 
         [0019]    In one embodiment, the processing device determines whether the phrase includes a trigger phrase (e.g., OK GOOGLE NOW) within a phrase segment received at the first microphone  102 . By “trigger phrase”, it is meant any phrase that signifies that a command is immediately present after the trigger phrase. The second microphone  104  is turned on by the processor  106  as a result of the trigger phrase having been detected at the first microphone; the second microphone  104  after activation captures voice data in real time. 
         [0020]    In one embodiment, the first microphone  102  is turned off by the processor  106  after a time period of (delay+x) where delay is the buffering delay of the first microphone  102  and x is the period of common speech information that has been received at each of the microphones  102  and  104 . In one example, x can be determined by the algorithm required to calibrate the two microphones. This calibration may include determination of and compensation for the acoustic delay and gain difference between microphones  102  and  104 . In embodiments where the first microphone is turned off, the first microphone is quickly turned back on after being turned off (e.g., within approximately 20 milliseconds) and placed in a normal mode of operation by receiving a clock signal  112  from the processor  106 . 
         [0021]    As will be more fully apparent from the discussion below, at least one and in some cases both microphones do not detect the entire uttered phase (e.g., OK GOOGLE NOW, WHAT IS THE WEATHER TODAY?) and thus the one or more microphones do not provide data derived from or corresponding to the entire phrase to the processor  106  for further processing. At the processor  106 , the entire phrase (e.g., OK GOOGLE NOW, WHAT IS THE WEATHER TODAY?) is stitched together for each microphone  102  and  104  based upon information received from both microphones. An output  114  from processor  106  includes assembled phrases  116  and  118  (e.g., each being OK GOOGLE NOW, WHAT IS THE WEATHER TODAY?) with the first phrase  116  being associated with the first microphone  102  and the second phrase  118  being associated with the second microphone  104 . It will be appreciated that this various processing described above can occur at either the codec  108  or the application processor  110 , or at other processing devices (not shown in  FIG. 1 ). 
         [0022]    Referring now to  FIG. 2 , one example of an approach for audio buffer catch-up is described. This processing occurs at the processor  106  (and in one specific example at the codec  108  although this functionality can be moved to or shared with the application processor  110 ). 
         [0023]    At step  202 , the first microphone is on and uses the buffer. The second microphone is off. 
         [0024]    At step  204 , the trigger phrase is detected from data received from the first microphone. At step  206 , the second microphone is turned on as a result of the trigger phrase having been detected. At step  208 , the second microphone captures voice data in real time. 
         [0025]    At step  210 , the first microphone is turned off after a time period of (delay+x) where the delay is the buffering delay of the first microphone (i.e., how long data takes to move through its buffer) and x is the period of common speech between the two microphones. In one example, x can be determined by the algorithm required to calibrate the two microphones. 
         [0026]    At step  212 , the first microphone is quickly turned on after being turned off (e.g., the microphone is activated approximately 20 milliseconds after being deactivated) and placed in a normal mode of operation (i.e., a non-buffering mode of operation as explained elsewhere herein). At step  214 , data derived from segments of the phrase received by the plural microphones are stitched together using suitable algorithms to form electronic representations of the entire phrase, one associated with each microphone. One example of assembling data from different microphones to form two separate electronic representations of complete phrases (one for first microphone and the other for the second microphone) is described below with respect to  FIGS. 3 and 4 . 
         [0027]    Referring now to  FIG. 3 , one example of an approach for audio catch-up using two microphones is described. It will be appreciated that this is one example of an approach using two microphones and that other examples are possible. 
         [0028]    At step  322 , a user utters a trigger phrase (e.g., OK GOOGLE NOW) seamlessly followed by a command (WHAT IS THE WEATHER TODAY?). In this example, the trigger phrase and command are collectively labeled as  302 . A first microphone  352  and a second microphone  354  (PDM microphone) detect parts of the complete phrase (e.g., the trigger and command in the example above). The first microphone  352  is in a low power sensing mode and all signals in this mode are buffered before being output, consequently introducing a buffer delay  324  (e.g., 256 milliseconds). During this delay time, a time period  304  exists at the output of the first microphone  352  where no audio is being supplied. At time  326 , the start of the audio output of the first microphone  352  occurs. As mentioned, the buffer delay is approximately 256 milliseconds and the 256 millisecond delayed output occurs at the output of the first microphone  352  during period  306 . 
         [0029]    Another delay  328  (in this case an approximately 100 millisecond delay) may be introduced by the trigger phrase recognition algorithm in the processor (e.g., codec, applications processor, or digital signal processor to mention a few examples). The trigger phrase recognition algorithm compares the received audio to a predefined trigger word or phrase, to determine whether the trigger word or phrase has been uttered. The delay  328  may occur after time  330 , which is the end of the delayed version of the trigger phrase. At time  332 , after an approximately 256 plus 100 millisecond delay, a beep (or other signal) is emitted or presented to the user signifying that the trigger phrase has been detected. In some examples, no beep may be used and in other examples the “beep” may be inaudible to humans. This signal may be a marker used in later processing and may be removed before stitching together the various speech segments. 
         [0030]    At time  334 , the second microphone  354  is turned on. Prior to the second microphone  354  being turned on, a time period  304  exists at its output where no audio is being produced. 
         [0031]    At time  336 , the first microphone  352  is turned off a predetermined time after turning on the second microphone  354 . The predetermined time may be 256 milliseconds (the buffer delay of the first microphone  352 ) plus x, where x is a time period  338  of overlapping speech segments that is used to determine the acoustic delay between microphones  352  and  354 . As shown here in this example, x relates to the phrase “E WEA” because “E WEA” is the audio that has been received at both of the microphones  352  and  354 . In these regards, the processor can first determine the common audio information received, and then use that to calibrate the microphone signals. This common time period (in this case, the length in time of the phrase “E WEA”) is the value of x. 
         [0032]    At step  340 , the first microphone  352  is quickly or immediately turned on after a very small (e.g., after an approximately 20 millisecond delay) to operate in a normal processing mode. By normal processing or operating mode, it is meant that the microphone  352  does not buffer the incoming signal, but passes data through without buffering the data. In one example, normal operating mode may be entered by applying a clock signal when applying the power signal, while in low power mode no clock signal is applied, till acoustic activity is detected. 
         [0033]    Referring now to  FIG. 4 , further explanation of assembling the phrases detected at two microphones  352  and  354  is provided.  FIG. 4  illustrates assembling the information together to form the complete phrases  430  and  432  for each of the microphones  352  and  354  respectively. When each of the phrases is assembled, each phrase “OK GOOGLE NOW, WHAT IS THE WEATHER TODAY?” will be complete, but the internal characteristics (e.g., amplitude, acoustic delay) within the phrase may be different for each microphone. These phrases may be used in further processing (e.g., transporting to the ‘cloud’ for speech recognition in a cellular phone, tablet, or personal computer). 
         [0034]    As shown, user uttered audio  402  is in this example OK GOOGLE NOW, WHAT IS THE WEATHER TODAY? There is a large missing segment of missing audio (at period  410 ) at the second microphone  354 . However, the first microphone  352  has obtained this audio at time period  404 . Thus, audio (for time period  404 ) can be used to obtain audio (for time period  410 ) for the second microphone, after appropriate calibration. 
         [0035]    Time period  406  is missing from the first microphone  352 . But, the second microphone  354  has obtained this audio and so this can be included in the audio for the first microphone  322 . 
         [0036]    The first microphone  352  has obtained audio for time period  408  in real time. Consequently, the complete audio phrase ( 430 ) (“OK GOOGLE NOW, WHAT IS THE WEATHER TODAY?”) has been assembled for the first microphone  352  since time periods  404 ,  406 , and  408  have been filled in. 
         [0037]    The second microphone  354  obtains real time audio for time period  412 . Consequently, the audio phrase  432  (“OK GOOGLE NOW, WHAT IS THE WEATHER TODAY?”) has been assembled for the second microphone  354  since time periods  410  and  412  have been filled in. 
         [0038]    In this way, the audio phrases  430  (from the first microphone  352 ) and  432  (for the second microphone  354 ) are assembled or stitched together. This processing may occur at a codec, application processor, or digital system processor to mention a few examples. The phrases  430  and  432  may be further processed by other processing devices as needed. 
         [0039]    Preferred embodiments of this disclosure are described herein, including the best mode known to the inventors. It should be understood that the illustrated embodiments are exemplary only, and should not be taken as limiting the scope of the appended claims.