Abstract:
A system ( 100 ) and method ( 300 ) are disclosed for filtering signals. A system that incorporates teachings of the present disclosure may include, for example, a speech processor ( 102 ) having an audio system ( 212 ) for audibly transmitting a rendition of a message, and for removing a portion of the rendered message embedded in a received signal as a result of at least one among electrical and electrical-magnetic interference between the rendered message and the received signal, thereby generating a filtered received signal. The audio system can capture the received signal while audibly transmitting the rendered message. Additional embodiments are disclosed.

Description:
FIELD OF THE DISCLOSURE 
       [0001]    The present disclosure relates generally to signal processing techniques, and more specifically to a method and apparatus for filtering signals. 
       BACKGROUND 
       [0002]    Audio circuits often suffer from a problem where the output signal is fed back into an input channel due to poor isolation. This feedback can be caused by any number of sources such as for example a leakage or crosstalk path in the audio circuit, audio loop back, an echo, and so on. 
         [0003]    A need therefore arises for a method and apparatus for filtering signals. 
     
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         [0004]      FIG. 1  depicts an exemplary embodiment of a communication system; 
           [0005]      FIG. 2  depicts an exemplary embodiment of a processor operating in the communication system; 
           [0006]      FIG. 3  depicts an exemplary method operating in the processor; and 
           [0007]      FIGS. 4-8  depict exemplary embodiments of the method operating in the processor. 
       
    
    
     DETAILED DESCRIPTION 
       [0008]      FIG. 1  depicts an exemplary embodiment of a communication system  100 . The communication system  100  can comprise a number of processors  102  wirelessly coupled to a network  101  for communicating with a server  104 . The speech processors  102  can utilize common wireless access technologies such as Bluetooth™, Wireless Fidelity (WiFi), Worldwide Interoperability for Microwave Access (WiMAX), Ultra Wide Band (UWB), software defined radio (SDR), Zigbee, or cellular for accessing the network  101 . The network  101  can comprise a number of dispersed wireless access points that supply the speech processors  102  wireless communication services in an expansive geographic area according to any of the aforementioned wireless protocols. The server  104  can comprise a scalable computing device for performing the operations depicted in the present disclosure. The communication system  100  can have many applications including among others a means for task processing in a medical services environment, or managing logistics of a commercial enterprise such as inventory management, shipping, distribution, and so on. 
         [0009]      FIG. 2  depicts an exemplary embodiment of the speech processor  102 . The speech processor  102  can comprise a wireless transceiver  202 , a user interface (UI)  204 , a headset  205 , a power supply  214 , and a controller  206  for managing operations of the foregoing components. The wireless transceiver  202  can utilize common communication technologies to support singly or in combination any number of wireless access technologies of the network  101  including without limitation Bluetooth™, WiFi, WiMax, Zigbee, UWB, SDR, and cellular access technologies such as CDMA-1X, W-CDMA/HSDPA, GSM/GPRS, TDMA/EDGE, and EVDO. SDR can be utilized for accessing public and private communication spectrum with any number of communication protocols that can be dynamically downloaded over-the-air to the speech processor  102 . Next generation wireless access technologies can also be applied to the present disclosure. 
         [0010]    The UI  204  can include a keypad  208  with depressible or touch sensitive keys, a touch sensitive screen, and/or a navigation disk for manipulating operations of the speech processor  102 . The UI  204  can further include a display  210  such as monochrome or color LCD (Liquid Crystal Display) for conveying images to the end user of the speech processor  102 , and an audio system  212  for conveying audible signals to the end user and for intercepting audible signals from the end user by way of a tethered or wireless headset  205 . 
         [0011]    The power supply  214  can utilize common power management technologies such as rechargeable and/or replaceable batteries, supply regulation technologies, and charging system technologies for supplying energy to the components of the speech processor  102  and to facilitate portable applications. The controller  206  can utilize computing technologies such as a microprocessor and/or digital signal processor (DSP) with associated storage memory such a Flash, ROM, RAM, SRAM, DRAM or other like technologies for controlling operations of the speech processor  102 . 
         [0012]      FIG. 3  depicts an exemplary method  300  operating in the speech processor  102 . Method  300  can operate in a portion of the speech processor  102  as software, hardware, or combinations thereof.  FIGS. 4-8  depict exemplary embodiments of portions of method  300 . 
         [0013]    With this in mind, method  300  begins with step  302  in which a first audio signal is transmitted to an end user of the speech processor  102 . The audio signal can be, for example, a “low battery” chirp or a voice message (such as a logistics command, medical directive, or status) transmitted by way of a speaker or audio transducer circuit of the audio system  212 . In applications where the speech processor  102  is configured for full duplex communications, a second audio signal can be received in step  304  by the audio system  212  while the first audio signal is transmitted. The second audio signal can include voice signals of the end user such as a command, or speech responsive to the first audio signal, as well as other ambient sounds. 
         [0014]    Because both input and output channels are concurrently active in the audio system  212 , leakages, crosstalk, reflections, audio loopback, echoes or any number of other distortions from the first audio signal can be inadvertently injected electrically or electro-magnetically into the second audio signal by, for example, a tethered headset  205  that couples to the audio system  212  with a common ground shared between the speaker and microphone elements of the headset  205 . Steps  306 - 308  can be applied to the speech processor  102  for removing this distortion. In step  306 , the audio system  212  can be designed or programmed to generate delayed samples of the first audio signal according to a delay estimated between the first and second audio signals. In step  308 , the audio system  212  can be designed to remove a portion of the first audio signal from the second audio signal by using the delayed samples of the first audio signal, the second audio signal, and a filtered received signal generated thereby. 
         [0015]      FIG. 4  depicts an exemplary embodiment of steps  306 - 308 . In this embodiment, the controller  206  is coupled to the audio system  212  by way of a digital interface. The audio system  212  comprises a codec  402 , a delay estimation module  404  and a filtration module  406 . The codec  402  includes a common digital to analog converter (DAC) for transforming digital samples of a first audio signal generated by the controller  206  into a first analog signal. The first analog signal is coupled to a common speaker circuit (not shown) of the audio system  212  for conveying audible signals to the end user. 
         [0016]    The codec  402  further includes a common analog to digital converter (ADC) for transforming a second analog signal intercepted by a common microphone (not shown) of the audio system  212  into digital samples representing a second audio signal. The first audio signal can be supplied to the delay estimation module  404  from a feedback path located prior to the codec  402 , or from a digital feedback path (FB) within the codec  402 . 
         [0017]      FIG. 5  depicts an exemplary embodiment of the delay estimation module  404 . The delay estimation module  404  can comprise a delay estimator  502  and associated delay element  504  for generating as discussed in step  306  delayed samples of the first audio signal according to an estimated delay between the first and second audio signals. The delay estimator  502  can utilize a common correlator for estimating the delay between the first and second audio signals. The delay element  504  utilizes common technology for delaying digital samples of the first audio signal according to the delay estimated by the delay estimator  502 . The delay estimator  404  time-aligns the signals that are received by the filtration module  406  with each other. It estimates and accounts for the difference in time between the first audio signal and the portion of the first audio signal received in the second audio signal. This difference can be due, for example, to asynchronous buffering (depicted by the letter “B” in  FIGS. 4 and 7 ) at the interfaces of the codec  402 . In an alternative embodiment, the first audio signal can be constructed by the controller  206  with a marker signal which the delay estimation module  404  can utilize for assessing delay. 
         [0018]    The filtration module  406  can comprise an adaptive filter such as, for example, a recursive least squares filter.  FIG. 6  depicts an exemplary embodiment of the adaptive filter which comprises a filter estimator  602  and corresponding filter  604  coupled to a difference element  606 . The filter  604  can be instantiated as a finite impulse response (FIR) filter (herein referred to as FIR filter  604 ). The filter estimator  602  can comprise a recursive least squares estimator for adjusting the filter coefficients of the FIR filter  604 . The FIR filter  604  generates according to the delayed samples of the first audio signal and the coefficients determined by the filter estimator  602  a signal that approximates the portion of the first audio signal embedded in the second audio signal. Accordingly, the difference element  606  removes in whole or in part the portion of the first audio signal embedded in the second audio signal thereby generating the filtered signal which is in large part free of the distortions introduced by the first audio signal. 
         [0019]      FIG.7  provides an alternative embodiment to the embodiment of  FIG. 4 . In this embodiment, the first audio signal is fed back in analog form through the codec or by way of an external input channel thereby incurring the same or similar delay as the portion of the first audio signal that exists in the second audio signal. With a predictable delay applied to the first audio signal by way of the loopback internal or external to the codec  402 , the delay estimator can be removed and the filtration module  406  can operate as described earlier. This approach can be utilized when the two audio input channels (i.e., the second audio signal and the looped back first audio signal ) are synchronized. The second audio signal and the looped back first audio signal can be synchronized much like left and right stereo input channel signals are commonly synchronized in time. 
         [0020]      FIG. 8  provides yet another alternative embodiment for steps  306 - 308  in which a common gain element  802  included in the codec  402  feeds back an adjusted first audio signal into a difference element  804  which removes in whole or in part a portion of the first audio signal embedded in the second signal thereby generating the filtered signal. This difference operation can be performed on either analog or digital signals. In this embodiment, the controller  206  can be programmed to perform signal processing on the filtered signal similar in operation to the filter estimator  602  and thereby adjust the gain element  802  to remove the embedded first audio signal in the incoming second audio signal. 
         [0021]    Once the second audio signal has been filtered as described by the foregoing embodiments of  FIGS. 4-8 , voice signals of the end user can be processed by the controller  206  in step  310  of  FIG. 3  according to common voice processing techniques (e.g., speech recognition, speaker identification, speaker verification, and so on). According to the voice signal supplied by the end user, the controller  206  can be programmed in step  312  to transmit the processed voice signal to the server  104  of  FIG. 1  (as text or unadulterated speech), or it can respond to said voice signals with a third audio signal. In a logistics or medical services application, for example, the end user&#39;s voice signals can represent commands or responses to commands emanating from the server  104 , or locally within the speech processor  102 . 
         [0022]    It would be evident to an artisan with ordinary skill in the art that the aforementioned embodiments of method  300  for removing distortion associated with the first audio signal embedded in the second audio signal can be modified, reduced, or enhanced without departing from the scope and spirit of the claims described below. For example, all or a portion of the delay estimation module  404  and filtration module  406  can be embedded in the codec  402  or the controller  206 . Additionally, a portion of the controller  206  can be embedded in the codec  402  also. System  400  can be utilized as a single chip solution embodied in a computing device or audio headset. Similarly, all or a portion of the delay estimation module  404  and filtration module  406  can be implemented in software, hardware or firmware. These are but a few examples of modifications that can be applied to the present disclosure. Accordingly, the reader is directed to the claims below for a fuller understanding of the breadth and scope of the present disclosure. 
         [0023]    The illustrations of embodiments described herein are intended to provide a general understanding of the structure of various embodiments, and they are not intended to serve as a complete description of all the elements and features of apparatus and systems that might make use of the structures described herein. Many other embodiments will be apparent to those of skill in the art upon reviewing the above description. Other embodiments may be utilized and derived therefrom, such that structural and logical substitutions and changes may be made without departing from the scope of this disclosure. Figures are also merely representational and may not be drawn to scale. Certain proportions thereof may be exaggerated, while others may be minimized. Accordingly, the specification and drawings are to be regarded in an illustrative rather than a restrictive sense. 
         [0024]    Such embodiments of the inventive subject matter may be referred to herein, individually and/or collectively, by the term “invention” merely for convenience and without intending to voluntarily limit the scope of this application to any single invention or inventive concept if more than one is in fact disclosed. Thus, although specific embodiments have been illustrated and described herein, it should be appreciated that any arrangement calculated to achieve the same purpose may be substituted for the specific embodiments shown. This disclosure is intended to cover any and all adaptations or variations of various embodiments. Combinations of the above embodiments, and other embodiments not specifically described herein, will be apparent to those of skill in the art upon reviewing the above description. 
         [0025]    The Abstract of the Disclosure is provided to comply with 37 C.F.R. §1.72(b), requiring an abstract that will allow the reader to quickly ascertain the nature of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or meaning of the claims. In addition, in the foregoing Detailed Description, it can be seen that various features are grouped together in a single embodiment for the purpose of streamlining the disclosure. This method of disclosure is not to be interpreted as reflecting an intention that the claimed embodiments require more features than are expressly recited in each claim. Rather, as the following claims reflect, inventive subject matter lies in less than all features of a single disclosed embodiment. Thus the following claims are hereby incorporated into the Detailed Description, with each claim standing on its own as a separately claimed subject matter.