Abstract:
At a microphone, voice activity is detected in a data stream while simultaneously buffering audio data from the data stream to create buffered data. A signal is sent to a host indicating the positive detection of voice activity in the data stream. When an external clock signal is received from the host, the internal operation of the microphone is synchronized with the external clock signal. Buffered data stream is selectively sent through a first path, the first path including a buffer having a buffer delay time representing the time the first data stream takes to move through the buffer. The data stream is continuously sent through a second path as a real-time data stream, the second path not including the buffer, the real-time data stream beginning with the extended buffer data at a given instant in time. The buffered data stream and the real-time data stream are multiplexed onto a single data line and transmitting the multiplexed data stream to the host.

Description:
TECHNICAL FIELD 
     This application relates to acoustic systems, and, more specifically to processing data in these audio systems. 
     BACKGROUND 
     Different types of acoustic devices have been used through the years. One type of device is a microphone and one type of microphone is a microelectromechanical system (MEMS) microphone, including a MEMS die having a diaphragm and a back plate. The MEMS die is supported by a substrate and enclosed by a housing (e.g., a cup or cover with walls). A port may extend through the substrate (for a bottom port device) or through the top of the housing (for a top port device). In any case, sound energy traverses the port, moves the diaphragm and creates a changing potential of the back plate, which creates an electrical signal. Microphones are deployed in various types of devices such as personal computers or cellular phones. 
     Digital microphones now exist that convert the analog data produced by the sensor into digital data. The digital data is utilized by different processing elements in the microphone to perform different sets of functions such as acoustic activity detection. Acoustic activity detection requires time to be performed in a reliable manner. Unfortunately, this time delay in detection incurs latency, which allows real-time data to pile or back-up thereby reducing the efficiency and performance of the system. The latency further requires use of a buffer to store audio data, while the acoustic activity detection is made. 
     The problems of previous approaches have resulted in some user dissatisfaction with these previous approaches, specially the latency that is incurred and that stays in the audio path impacting user experience in voice recognition tasks. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       For a more complete understanding of the disclosure, reference should be made to the following detailed description and accompanying drawings wherein: 
         FIG. 1  is a block diagram of a microphone; 
         FIG. 2  is a block diagram of a system of two microphones and a host; 
         FIG. 3  is a block diagram of a host; 
         FIGS. 4A and 4B  illustrate a timing diagram of the operation of the systems described herein according to various embodiments of the present invention; 
         FIG. 5  is a flow chart of the operation of the systems described herein; 
         FIG. 6  is a diagram showing one example of stitching; 
         FIG. 7  is a flow chart showing a stitching approach; 
         FIG. 8  is a time line and data diagram showing the stitching approach of  FIG. 7 ; 
         FIG. 9  is a flowchart of another stitching approach; 
         FIG. 10  is a time line and data diagram showing the stitching approach of  FIG. 9 . 
     
    
    
     Skilled artisans will appreciate that elements in the figures are illustrated for simplicity and clarity. It will be appreciated further 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 
     The present approaches allow a first microphone to be operated in a mode having a real-time data path and a path that includes buffered data. The present approaches utilize a host processing device that enables the buffered audio data of the first microphone to catch up or recover the latency as compared to the real-time or live audio data capture. Among other things, this allows the use of a second microphone where the second microphone does not have a buffer. Consequently, any latency issues associated with the first microphone are traversed. 
     In many of these embodiments and at a host processing device, buffered pulse density modulation (PDM) data and real-time PDM data that has not been buffered is received from a first microphone. The buffered PDM data and the real-time PDM data have the same data content but are discontinuous with respect to the other when received at the host processing device. The buffered PDM data is processed over a first time interval and the real-time PDM data is processed over a second time interval. The host processing device is operated so that the second time interval is less than the first time interval. The real-time PDM data is stitched to an end of the buffered PDM data. The stitching is effective to time align the buffered PDM data with respect to the real-time PDM data to create an output data stream that is sequentially ordered in time. This allows the latency that is otherwise always present in this class of acoustic activity detection MEMS microphones to be transferred to the host device, where it can be easily recovered by faster than real-time processing. 
     In other aspects, second real-time data is received from a second microphone, the second microphone not having a buffer. In some examples, the second real-time data is inserted into the output stream after the conclusion of the latency recovery mechanism described herein. 
     In other examples, the processing of the buffered PDM data comprises determining an existence of a trigger word or phrase in the buffered PDM data. In yet other examples, the buffered PDM data and the real-time PDM data are decimated. In some examples, the buffered PDM data and the real-time PDM data are received in a multiplexed format. 
     In others of these embodiments, a host processing device includes an interface and a processor. The interface has an input and output, and is configured to receive buffered pulse density modulation (PDM) data and real-time PDM data that has not been buffered from a first microphone at the input. The buffered PDM data and the real-time PDM data have the same data content but having a latency and being discontinuous with respect to the other when received at the host processing device. The processor is coupled to the interface, and the processor is configured to process the buffered PDM data over a first time interval and process the real-time PDM data over a second time interval. The processor is operated so that the second time interval is less than the first time interval. The processor is configured to stitch the real-time PDM data to an end of the buffered PDM data. The stitching is effective to synchronize the buffered PDM data with respect to the real-time PDM data and to create an output data stream at the output. 
     Referring now to  FIG. 1 , a low power acoustic activity detection (AAD) microphone  100  is described. The microphone  100  includes a charge pump  102 , a transducer  104  (including a back plate and diaphragm), an input buffer  106  (with adjustable gain), a sigma delta modulator  108 , a decimator  110 , an Acoustic Activity Detection (AAD) module  112 , a circular buffer  114 , a first up converter  116 , a second up converter  118 , a control block (processor)  120 , an internal oscillator  122 , and a clock detector  124 . 
     The microphone  100  provides Voice Activity Detection (VAD) capabilities at ultra-low power. The AAD module  112  (including a (VAD) gain block) detects voice and voice-like activity. The circular buffer  114  receives data in real-time. In one aspect, the buffer may of sufficient size to hold 256 msec of audio. In another aspect, the buffer size may be trimable to sizes other than 256 msec. The charge pump  102  provides charge or energy to the transducer  104 , and the transducer  104  converts an acoustic signal into an analog signal, which is stored in the input buffer  106 . The sigma delta modulator  108  converts the analog signal into a pulse density modulation (PDM) signal, and the decimator  110  converts the PDM signal into a pulse code modulation (PCM) signal. PCM data has two paths: a first path through the circular buffer  114  to up-converter  118 , and a second path for real-time data that flows directly through up-converter  116 . 
     The first up converter  116  and second up converter  118  convert PCM data into PDM data. The control block (processor)  120  determines when transmissions are made to a host. The internal oscillator  122  supplies a clock signal and the clock detector  124  determines whether an external clock has been received from an external host via pin  134 . 
     The AAD module  112  detects acoustic activity in a low power operating mode of the microphone. The sensitivity of this block is partially controlled through the input gain of this block. The VAD gain portion of the AAD module  112  in one aspect has a trimable gain. The AAD module  112  monitors the incoming acoustic signals looking for voice-like signature, without the need for an external clock on clock pin  134  and this operation occurs in the aforementioned low power sensing mode. Upon detection of acoustic activity that meets the trigger requirements, the microphone  100  asserts a SEL/STAT pin  130  to wake up the rest of the system in the signal chain. Further, the microphone  100  provides real-time PDM data on DATA line  132  when a clock is made available on the CLOCK line provided by the system after it wakes up. The buffer  114  stores a previous amount of data (e.g., the previous 256 msec of data or a pre-set trimmed amount which may be different from 256 msec) generated prior to the activity detection. Once a clock signal has been detected on pin  134 , the microphone  100  transmits the buffered data to a host via DATA line  132 . Data output may start at the same time as the SEL/STAT line  130  indicates detection of voice. Alternatively, data output may start after receiving an external clock via pin  134 . 
     Referring now to  FIG. 2 , another example of a system with a catch-up buffer is described. The system includes a first microphone  202 , a second microphone  204 , and a host  206 . 
     The first microphone  202  includes a transducer  222  (including, for example, a diaphragm and back plate), a sigma delta converter  224 , a decimator  226 , a buffer  228 , a first up-converter  230 , a second up-converter  231 , a transmitter  232 , a buffer control module  234 , a control module  236 , an Acoustic Activity Detection (AAD) module  238 , and an internal clock  240 . 
     The second microphone  204  includes a transducer, but does not include a buffer. In these regards, the second microphone  204  may be a micro electro mechanical system (MEMS) device that converts sound energy into an electrical signal. The second microphone  204  may include a back plate and a diaphragm. Other examples of microphones are possible. 
     The host  206  is, in one example, a processing element such as a codec or digital signal processor. The structure of the host  206  is described with respect to  FIG. 3 . The host  206  receives data streams (that may be multiplexed over a PDM data line  280 ). The first data stream is from the buffer  228  and the second data stream is un-buffered data. The buffer  228  introduces latency (delay), but is needed because the first microphone  202  needs time for the AAD module  238  to determine whether there is voice (or other acoustic) activity. Additionally, the host processor requires time to wake up from a low power mode and be ready to receive data. The buffer also provides important contextual information to a speech trigger recognition engine to allow it to perform better in noisy conditions. Because of the delay and latency, the two data streams (of the same data content) will be discontinuous and time delayed with respect to each other. The host  206  operates to synchronize the two data streams at its output, and eliminates any discontinuous aspects with respect to each other. In other words, the host guarantees that at some point in time, input data that it is receiving (from one or both of the first or second microphones) is the same data that it is outputting. 
     The transducer  222  (which may be a micro electro mechanical system (MEMS) device) converts sound energy into an analog electrical signal. The sigma delta converter  224  converts the analog electrical signal into a pulse density modulation (PDM signal). The decimator  226  converts the PDM signal into a pulse code modulation (PCM) signal. The buffer  228  stores the PCM signals. The up-converter  230  converts PCM signals into PDM signals. The transmitter  232  transmits a multiplexed signal (of the first and second data streams) over the data line  280 . The transmission is initiated with the receipt of the external clock on line  284 . The buffer contents are monitored by the buffer control module  234 . When the buffer has transmitted the pre-determined amount of data, for example 256 msec and some additional extension data (by “extension data” it is meant as data beyond the buffer length), the buffer control module  234  sends a buffer empty (bempty) signal  285  to the control module  236 , which causes the transmitter  232  to stop multiplexing the contents of the buffer  228 . The AAD module  238  detects whether there is voice or other acoustic signals and sends a SEL/STAT signal  282  when acoustic activity is detected. The host  206  responds with a clock signal  284 , which is sent to the first and second microphones  202  and  204 . The second microphone  204  is also controlled via the GPIO  286  which keeps microphone  204  disabled. The effect of the clock signal  284  is to cause microphone  202  to transmit data. A GPIO  286  is used to control power to the second microphone  204  and to select the second microphone  204 . The GPIO  286  is asserted only after stitching is completed at the host. The term “stitching,” means combining the real-time data stream at the end of the buffered data stream in the host, such that a continuous data stream is presented to the application. 
     In one example of the operation of the system of  FIG. 2 , the first microphone  202  stores or buffers data in the buffer  228  in order for acoustic activity detection to be performed by AAD module  238  on the buffered data. The host  206  is awaken by the SEL/STAT signal  282  and responsively sends the clock signal  284  to the first microphone  202 . Receipt of the clock signal allows the first microphone  202  to clock data out over data line  280 . 
     The first microphone  202  sends multiplexed data (of the first and second streams) to the host  206 . This multiplexed data will include real-time and buffered data of length X time units (e.g., 256 ms). 
     The host  206  processes the X units of buffer data until the processing is complete. X units of real-time data is also waiting for processing by the host  206 . The host  206  processes the real-time data over a second time period that is much, much less than the first time period. The host  206  may be operated faster to accomplish this function. The host  206  stitches the real-time data to the end of the buffered data. The goal is that the data being input into the host  206  is being output from the host  206  in real-time. 
     In order to support low power applications that require or prefer to reduce the signal latency due to the buffer  228 , a burst mode is provided in the system of  FIG. 2 . Burst mode provides the capability for faster than real-time data transfer. Burst mode implements two data paths, one for the real-time data and the other for the buffered data, both of which go through the decimation and interpolation functions needed to run the AAD module  238 , for example, at 16 kHz/16 bits PCM. In one aspect, the burst mode utilizes two interpolators to ensure that the signal paths for both signals have the same phase response, excluding any coding and decoding associated with the buffering operation. 
     The burst mode operates as follows. The SEL/STAT line  282  is used for signaling the state of the microphone  202  to the host  206 . The microphone  202  is normally in sense mode with no activity on the data line  280  and SEL/STAT line  282 , when there is no voice and the microphone AAD module  238  has converged to the ambient noise. 
     When the AAD module  238  detects acoustic activity and asserts the SEL/STAT line  282 , the host  206  enters the wake-up mode. This action wakes up the host  206  with some latency. The host  206  in one aspect provides a 768 kHz signal to the clock line  284 . 
     The reception of the clock signal  284  by the first microphone  202  along with acoustic detection puts the first microphone  202  into burst mode. In one example, the first microphone  202  enters burst mode within 10 clock cycles of receiving the external clock at 768 kHz. The burst mode uses a first PDM channel to send the buffer data and a second PDM channel to send real-time data to the host. 
     In some aspects, the real-time PDM channel may be the default channel, so that the real-time data is valid and may be latched during the rising edge of the clock. Buffered data is valid and may be latched during the falling edge of the clock. The data transfer rate in burst mode is in one example double the normal data rate at 768 kHz. When in the burst mode and in one example, the first microphone  202  will toggle the SEL/STAT pin  282  at 8 kHz, synchronous to the 768 kHz CLOCK edges. When the buffer  228  is emptied via the burst mode, the SEL/STAT pin  282  is held high so the host  206  is signaled that the first microphone  202  is now caught up with real-time data. The host  206  may also use a count of the toggle to verify the amount of data collected to aid in “stitching” the buffered and real-time data. Slower toggle rates will cause lower overhead on host systems. In one aspect, the use of an 8 kHz toggle rate will allow the time between each transition to be the duration of 1 PCM sample. 
     The signal processing algorithms for decimation may cause pops or clicks at the stitch point of the real-time and buffered audio. By a “pop” or “click,” it is meant that unnatural discontinuities in the audio samples will cause distortions in the output audio signal that resemble a “pop” or “click” sound. Some overlap is expected to be required between the buffered and real-time data to eliminate these pops or clicks. The buffered data will be extended beyond the 256 msec or the specific trimmed size to provide this overlap. During the extended buffer state, the SEL/STAT line  282  is held high. The end of the extended buffer period is signaled by toggling SEL/STAT pin  282  at 16 kHz to allow distinction from the burst mode state. 
     At the end of the extended buffer period or state, the first microphone  202  enters the Real-Time low power mode. When in Real-Time low power mode, the first microphone  202  only uses one of the PDM channels. Data is valid during the rising edge. This permits the use of the second PDM microphone  204  on the PDM port. The second PDM microphone  204  has to be off during the combined time for burst mode output and extended buffer output durations. The SEL/STAT toggle on line  282  may be used as a signal to determine when the second microphone  204  can be powered on. The SEL/STAT pin  282  will keep toggling until the end of detected voice activity. Thus, the activity of the SEL/STAT pin  282 , either high or toggling is an indicator of voice activity. If the host  206  uses internal timers available to it, exact grabbing of the extension buffer may not be necessary, but may be self-regulated by the host  206 . 
     Only after the cessation of voice activity and the external clock  284  from the host  206  will the first microphone  202  re-enter sense mode. 
     Referring now to  FIG. 3 , one example of a host  300  (e.g., host  206  from  FIG. 2 ) is described. The host  300  includes a stereo decimator  302  (acting as an interface) and a processor  304 . The decimator  302  converts PDM data into PCM data. The processor  304  implements or executes stitching approaches (any combination of hardware and software) that append real-time data to the buffered data. The processor  304  includes a buffer for real-time data. 
     Data discontinuity exists at the start of a burst when the microphone (e.g., microphone  202 ) is operated in burst mode. Discontinuity can be represented as x(m)-x(n) and is approximately equal to 256 ms where 256 ms is the buffer length of the first microphone (e.g., microphone  202 ). A voice trigger algorithm starts recognition on the buffered data, x(m) over a first processing interval, while the real-time data x(n) is saved in a buffer on the host  300  and will be processed by voice trigger algorithm over a second processing interval. Data is stitched by the host  300  (e.g., host  206 ) after the entire buffer (256 ms) is drained and latency is consequently recovered. Buffer data of the buffer in the first microphone (e.g., buffer  228  in first microphone  202 ) is extended (e.g., by a length less than 256 ms) to allow the stitch algorithm operated by the processor  304  to synchronize x(m) and x(n) and eliminate signal discontinuity. 
     After data discontinuity is resolved and synchronization is achieved, real-time data from the first and second microphones (e.g., microphones  202  and  204 ) can be multiplexed on the incoming data line and output in real-time. This may correspond to a low-power real-time mode. 
     Referring now to  FIG. 4 , a timeline showing the operation of the approaches described herein is described. The time line shows the occurrence of voice activity  402 . It will be appreciated that this timing diagram illustrates the operation of the systems described with respect to  FIGS. 1-3 . 
     Voice is detected causing the SEL/STAT line  404  to go high. SEL/STAT stays high until the clock (e.g., 768 kHz clock) is received from the host. The host sends clock signal  406  back to the first microphone. The first microphone detects the clock signal and sends data out on data line  408  at time  410 . SEL/STAT then toggles at a suitably chosen signaling frequency. An example frequency that may be used is 8 kHz. On the rising edge of the clock, real-time PDM data  440  is received over the data line. On the falling edge, buffer PDM data is received over the data line from the first microphone. This is the burst mode. 
     Then at time  412 , extension mode is entered. On the rising edge of the clock real-time PDM data is received over the data line and on the falling edge of the clock extension buffer data is received over the data line. This allows the host to stich the real-time data to the buffer data. The extension period may last a pre-determined time. In one example, this extension period is less than 128 ms and in other examples, this extension period could be 32 msec, 16 msec or 8 msec or another suitable time interval. SEL/STAT toggles at a suitably chosen signaling frequency different from the burst mode signaling frequency until AAD goes inactive. An example frequency could be 16 kHz. At this point, real-time PDM data alone is being received over the data line. Optionally, at time  414 , a second microphone (without a buffer) may be powered on the falling edge of the clock after the buffer extension period. On the rising edge of the clock real-time PDM data from first microphone is received over the data line and on the falling edge of the clock real-time PDM data from second microphone is received over the data line. 
     Referring now to  FIG. 5 , one example of the state transitions is described. It will be appreciated that this flow chart illustrates the operation of the systems described with respect to  FIGS. 1-4 . 
     At step  502 , the system is powered ON. At step  504 , determine if the SEL/STAT line is VDD or floated. 
     If at step  504  VDD and acoustic activity detection (AAD) is off, then at step  506  the external clock rate is determined. In one aspect of the invention, if the clock rate is 0-315 kHz, at step  508 , the microphone goes to sleep mode. If the clock rate is between 315 and 1400 kHz, at step  510 , the microphone is operated in low power mode. If the clock rate is between 1.4 to 4.8 MHz, the microphone goes to normal operating mode at step  512 . 
     If at step  504  the SEL/STAT is floated, then at step  514  it is determined if there is an external clock being received at the microphone. If the external clock is detected to be 1 to 4.8 MHz, execution continues with step  526  where the microphone is operated in normal operating mode. If the external clock is at 768 kHz, execution continues with step  524  at a low power real-time mode. If the answer at step  514  is negative, at step  516  the microphone enters PDM sensing mode. At step  518 , wake up is performed. If no external clock is being received at the microphone, execution continues with step  516 . If external clock is being received at the microphone, burst mode is entered at step  520 . At step  520 , burst mode is executed as has been described herein. If at step  524  or step  526 , the external clock is stopped, then the execution reverts to block  516  and the microphone enters the PDM sensing mode. 
     Referring now to  FIG. 6 , one example of stitching data from the buffer and real-time data is described. It will be appreciated that this example shows how data may be stitched together in the system of  FIGS. 1-5 . 
     A buffer (e.g., the buffer  228  in the first microphone of  FIG. 2 ) includes buffered data  602 . An audio phrase is received. “He” which (in this example) is the first part of the phrase “Hello VT.” Real-time data  604  also is received and this may be “llo VT” from the last part of the phrase “Hello VT.” The stitching algorithm in the host (e.g., host  206 ) receives these two data streams and stitches “llo VT” to the end of the buffered data to make stitched data  606  “Hello VT.” The processing of the buffer data must proceed at a real-time rate as it is received at a real-time rate with the latency determined by the buffer size in the microphone. The processing of the real-time data may be made much faster than real-time, because of the accumulated data in the host after the stitching process is completed. Thus, the stitched continuous data stream present at the output of the host recovers the latency and catches up to the live signal with significantly reduced latency. The buffered data  602  and the real-time data  604  are now ordered sequentially with each other and the host can process the data received from one or more microphones in real-time without needing to consider synchronization issues between the first and the second microphone. 
     Referring now to  FIG. 7  and  FIG. 8 , one example of a stitching approach is described. The discussion with respect to  FIG. 7  and  FIG. 8  assumes a microphone and host device, for example, as described previously above. 
     Transients occur whenever PDM data is fed into a decimation filter or when it is stopped. In some aspects, when buffered data is followed by the real-time data, the transients will occur in the middle of the combined audio streams. Using an extended buffer of length greater than the end transient of the buffered audio and the start transient of the real-time audio allows the skipping of these time intervals by calculation of the decimation filter characteristics. One stitching approach provides an extended buffer and skips these transients. Thus, first the buffered and real-time signal must be time aligned at the host. This is possible because both streams start simultaneously only after the host clock is received. 
     The lengths of the buffer and the extended buffer are pre-determined and may be based upon various factors. 256 ms and 16 ms are examples of lengths for the buffer and extended buffer, respectively. 
     The output data is taken from the buffered audio until it is past the point where the start transient of the real-time audio has damped out. The output data is then switched to the corresponding real-time stream, so that the transient at the end of the extended buffer data may be skipped. This audio stream does not have any transient in the middle of the stream with this stitching strategy. 
     At step  702 , the host is asleep. At step  704 , the microphone wakes up the host, for instance, as has been described above. 
     At step  706 , various parameters or variables are initialized. More specifically, Bufl is the length of the buffer and this is initialized to a value, for example, in milliseconds (e.g., 256 ms). Bufl is shown as element  802  in  FIG. 8 . 
     Stpt is the stich point and is a time value as measured from the end of BUFFERPCM. It is also the same time value when measured from the beginning of the RT_BUF, the real-time buffer on the host. Stpt is represented as element  804  in  FIG. 8 . Extl is the length of the extension buffer in the microphone and is represented by element  803 . 
     Rt_Buf[BufL+StPt] is an amount of allocation of memory space for real-time data in the host. Real-time data will be stored in a real-time buffer in the host. In one example, the real-time buffer Rt_buf could be set to 256 ms+8 ms if 8 ms is the stitch point. pWR and pRD are write and read pointers and these are initialized to zero. 
     At step  708 , a check is made to determine if line  130  (of  FIG. 1 ) is active. If it is not, return to step  702 . 
     If the line is active at step  710 , the host inputs the 2 channels (stereo) of data. The host decimates the data from PDM format to PCM format. 
     At step  712 , store the real-time PCM data in a real-time buffer using the pWR pointer to point to the correct place in the buffer to write the data. 
     At step  714 , a check is made to determine if the pWR pointer has gone past the stitch point. If it has not, at step  716  output the buffered data stream (buffered PCM data) so that it can be further processed. At step  718 , the pWR pointer is incremented. 
     If at step  714 , the pWR pointer has gone beyond the stitch point, control continues to step  720 . A check is made to see if the pRD flag (used as a position pointer in the real-time data buffer in the host) has reached the stitch point. If it has, output real-time data at step  726 . If it has not reached the stitch point, real-time buffer data [pRD+StPt] is output. Then, the pRD pointer is incremented at step  724 . 
     It can be seen in  FIG. 8  that the output of this approach will have region  830  (from buffered PCM), region  832  (from extended buffer), region  834  (from extended buffer from RT buffer), region  836  (from RT buffer), and region  838  (not from RT buffer), as the data comes in to the host. It is apparent that the transient regions are avoided because the regions that include the transients are not used (as from a particular buffer). 
     Referring now to  FIG. 9  and  FIG. 10 , another example of a stitching approach is described. The discussion with respect to  FIG. 9  and  FIG. 10  assumes a microphone and host device, for example, as described previously above. At step  902 , the host is asleep. At step  904 , the microphone wakes up the host. 
     At step  906 , various parameters or variables are initialized. Bufl is the length of the buffer and this is initialized to a value, for example, in milliseconds. Bufl is shown as element  1001  in  FIG. 10 . Trpt is the length of decimator transients and these are represented by elements  1002  and  1004  in  FIG. 10 . Rt_Buf[BufL+TrPt] is a memory allocation for real-time data. Real-time data will be stored in a real-time buffer in the host. This could be 256 ms+8 ms if 8 ms is the stitch point. pWR and pRD are write and read pointers and these are initialized to zero. Extl is the length of the extension buffer in the microphone and is represented by element  1003 . IS is the interpolated sample. 
     At step  908 , a check is made by the host to see if line  130  of  FIG. 1  is active. If it is not, a return to step  902 . At step  910 , if it is active, then input the 2 channels (stereo) of data is made. The data is decimated from PDM format to PCM format. 
     At step  914 , the approach is dealing with transient period lengths TrPt  1002  and  1004  lengths, which are assumed to be equal. A check is made to see if pWR is in that area of data. 
     If the pWR pointer is not in the transient area, at step  916  buffered PCM data is output from the host and at step  918  pWR (which is the pointer used in the buffer to store real-time data in the host) is incremented. 
     If the approach has reached the transient portion, pWR is somewhere in the middle of zone  1006 . At step  920 , a check is made to see if pWR is out of that zone  1006 . If the answer is negative, then at step  922  interpolate the output data based on weighting. PCM data that is interpolated is output from the host at step  924 . pWR and pRD are incremented at step  926 . 
     If the determination made is that the pointers are out of the  1006  zone, then control continues with step  928  where a determination is made as to whether pRD is out of zone  1004 . If not out of zone  1004 , at step  930  output real-time buffer data RT_BUF[pRD+TrPt]. At step  932 , the pointer pRD is incremented. 
     If the process moves out of zone  1004  (by the determination at step  928 ), real-time (unbuffered) data is output from the host at step  934 . 
     It can be seen that an interpolated region in the output steam avoids the transients. The output is a buffered PCM data region  1030 ; interpolated region  1032  (that avoids the transients of regions  1002  and  1004 ); and real-time buffer region  1034  (from the real-time buffer); and region  1036 , which is real-time data that is unbuffered. 
     It will be understood that different interpolation approaches may be used. If infinite input response (IIR) filters are used in decimation, then the transient persists in perpetuity though with decreasing energy to meet design goals. In some situations, the stitch point still shows some broadband noise at the stitch point when basic stitching is used. In interpolated stitching, an allowance is made for the most significant energy of the transients to die down. Then, the intermediate time interval is used to linearly interpolate between the buffered and real-time data. The interpolation may be performed in one example as follows. 
     Let the time interval be given by discrete variable n. The start of the buffered audio may be considered n=0. An assumption may be made that the time for the most significant energy of the transients to die down is TrPt. The output for each section is given by the following equations respectively. 
     For the first segment  1030 :
 
 op ( n )=BUFPCM( n ) for 0&lt; n ≦(Buf L+TrPt )
 
     This equation describes that the output of the host is determined solely based on buffered data. 
     For the intermediate segment  1032 :
 
 op ( n )=α( n )×Ext L ( n )+[1−α( n )]×RTBUF( n )
 
for (Buf L+TrPt )&lt; n  
 
≦(Buf L +Ext L  
 
− TrPt )
 
where α( n )= n /(Ext L− 2× TrPt )
 
     This equation describes that data in the intermediate segment is linearly interpolated in both data streams. 
     For the segment  1034 :
 
 op ( n )=RTBUF( n ) for (Buf L +Ext L−TrPt )&lt; n  
 
     This equation describes that the output of the host is determined solely based on real-time buffered data. The above approach results in significantly lower transient broadband energy in the segment where the output is in transition from the buffered data stream to the real-time data stream. 
     In the equation above, op(n) is output at processing cycle n, n is counter of processing cycles, BUFPCM(n) is buffer PCM sample of processing cycle n, RTBUF(n) is real-time PCM sample of processing cycle n, ExtL(n) is extension buffer PCM sample of processing cycle n, and α(n) is time varying weight factor of processing cycle n. In one aspect, α(n) is defined to increase linearly from 0 to 1 with increasing n. 
     The first and last equations determine when the output is determined solely by the Buffered data and the Real-Time data and the intermediate equation determine how the data in the intermediate segment is linearly interpolated from both data streams. 
     This results in significantly lower transient broadband energy in the segment where the output transitions from the buffered data stream to the real-time data stream. In other words, buffered data is used more at the beginning of the interpolation, while real-time data is used less. Real-time data is used less at the beginning and more at the end. The degree of use for each may be described as a linear function. 
     Preferred embodiments 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.