Abstract:
An improved echo cancellation system ( 400 ) includes a double talk detector ( 406 ) configured for detecting a double talk condition by monitoring voice energy in a first frequency band ( 503 ). An adaptive filter ( 420 ) is configured for producing an echo signal based on a set of coefficients, and holds the set of coefficients constant when the double talk detector ( 406 ) detects the double talk condition. A microphone system ( 402 ) inputs audible signals ( 404 ) in a second frequency band ( 520 ) that is wider and overlaps the first frequency band ( 503 ). The echo signal is used to cancel echo in the input signal. A loud speaker ( 401 ) is configured for playing voice data in a third frequency band ( 501 ) essentially equal to a difference of the first and second frequency bands ( 502  and  503 ). The first and third frequency bands ( 503  and  501 ) essentially makeup the second frequency band ( 502 ).

Description:
BACKGROUND  
         [0001]    1. Field of the Invention  
           [0002]    The disclosed embodiments relate to the field of echo cancellation system, and more particularly, to echo cancellation in a communication system.  
           [0003]    2. Background  
           [0004]    Echo cancellation is generally known. Acoustic echo cancellers are used to eliminate the effects of acoustic feedback from a loudspeaker to a microphone. Generally, a device may have both a microphone and a speaker. The location of the microphone may be close to the speaker such that the audio from the speaker may reach the microphone at sufficient level. When the feedback from the speaker to the microphone is not cancelled through an echo cancellation process, the far end user that produced the audio in the speaker may hear its own speech in a form of echo. The near end user may also speak into the microphone while the far end user speech is being played by the speaker. As a result, the far end user may hear its own echo and the near end user speech at the same time. In such a situation, the far end user may have difficulty hearing the near end user. The impact of echo is especially annoying during conversation. An echo canceller may be used by the near end user device to cancel the echo of the far end user before any audio generated by the near end user microphone is transmitted to the far end user. However, it is difficult to cancel the echo in the audio frequency band when the near end user speech is also in the audio frequency band.  
           [0005]    A block diagram of a traditional echo canceller  199  is illustrated in FIG. 1. The far-end speech f(t) from a speaker  121  may traverse through an unknown acoustic echo channel  120  to produce the echo e(t). The echo e(t) may be combined with the near-end speech n(t) to form the input (n(t)+e(t))  129  to the microphone  128 . The effect of acoustic echo depends mainly on the characteristics of acoustic echo channel  120 . Traditionally, an adaptive filter  124  is used to mimic the characteristics of the acoustic echo channel  120  as shown in FIG. 1. The adaptive filter  124  is usually in the form of a finite impulse response (FIR) digital filter. The number of taps of the adaptive filter  124  depends on the delay of echo that is attempted to be eliminated. The delay is proportional to the distance between the microphone and the speaker and processing of the input audio. For instance, in a device with microphone and speaker mounted in close proximity, the number of taps may be smaller than 256 taps if the echo cancellation is performed in 8 KHz pulse code modulation (PCM) domain.  
           [0006]    In a hands-free environment, an external speaker may be used. The external speaker and the microphone are usually set far apart, resulting in a longer delay of the echo. In such a case, the adaptive filter  124  may require  512  taps to keep track of the acoustic echo channel  120 . The adaptive filter  124  may be used to learn the acoustic echo channel  120  to produce an echo error signal e 1 (n). The error signal e 1 (n) is in general a delayed version of far end speech f(t). The input audio picked up by the microphone  128  is passed through an analog to digital converter (ADC)  127 . The ADC process may be performed with a limited bandwidth, for example 8 kHz. The digital input signal S(n) is produced. A summer  126  subtracts the echo error signal e 1 (n) from the input signal S(n) to produce the echo free input signal d(n). When the adaptive filter  124  operates to produce a matched acoustic echo channel, the estimated echo error signal e 1 (n) is equal to the real echo produced in the acoustic echo channel  120 , thus:  
           d(n)=s(n)−e 1 (n)=[n(n)+e(n)]−e 1 (n)=n(n)  
           [0007]    where n(n) and e(n) are discrete-time version of n(t) and e(t) respectively after 8 KHz ADC. A voice decoder  123  may produce the far end speech signal f(n) and passed on to an ADC  122  to produce the signal f(t). Moreover, the signal d(n) is also passed on to a voice encoder  125  for transmission to the far end user.  
           [0008]    A gradient descent or least mean square (LMS) error algorithm may be used to adapt the filter tap coefficients from time to time. When only the far-end speaker is speaking, the adaptive filter attempts to gradually learn the acoustic echo channel  120 . The speed of the learning process depends upon the convergence speed of algorithm. However, when the near-end speaker is also speaking, the adaptive filter  124  holds its tap coefficients constant since the adaptive filter  124  needs to determine the coefficients based on the far-end speech. When the adaptive filter  124  adjusts its tap coefficients while the near end speech n(n) exists, the adaptation is based on the near end speech, which is the form of: d(n)=n(n)+[e(n)−e 1 (n)], instead of the signal in the form of: d(n)=e(n)−e 1 (n). Therefore, if the filter adaptation is allowed while the near-end speech exists, the filter coefficients may diverge from an ideal acoustic echo channel estimate. One of the most difficult problems in echo cancellation is for the adaptive filter  124  to determine when it should adapt and when it should stop adapting. It is difficult to discern whether the input audio is a loud echo or a soft near-end speech. Normally, when the speaker is playing the far end user speech and the near end user is speaking, a double talk condition exists. A condition of double talk detection is achieved by comparing the echo return loss enhancement (ERLE) to a preset threshold. ERLE is defined as the ratio of echo signal energy (σ e   2 ) and the error signal energy (σ d   2 ) in dB:  
       ERLE   =       10          log        (       σ   e   2       σ   d   2       )                               
 
           [0009]    The echo and error signal energy is estimated based on short-term frame energy of e(n) and d(n). ERLE reflects the degree for which the acoustic echo is canceled and how effective the echo canceller cancels the echo. When double talk occurs, ERLE becomes small in dB since the near end speech exists in d(n). A traditional double talk detector may be based on the value of ERLE. If ERLE is below a preset threshold, a double talk condition is declared.  
           [0010]    However, the double talk detection based on ERLE has many drawbacks. For instance, ERLE can change dramatically when the acoustic echo channel changes from time to time. For example, the microphone may be moving, thus creating a dynamic acoustic echo channel. In a dynamic acoustic echo channel, the adaptive filter 124  may keep adapting based on ERLE values that are not reliable enough to determine a double talk condition. Moreover, adapting based on ERLE provides a slow convergence speed of the adaptive filter  124  in a noisy environment.  
           [0011]    Furthermore, the near end user device may utilize a voice recognition VR system. The audio response of the near end user needs to be recognized in the VR system for an effective VR operation. When the audio response is mixed with echo of the far end user, the audio response of the near end user may not be recognized very easy. Moreover, the audio response of the near end user may be in response to a voice prompt. The voice prompt may be generated by the VR system, voice mail (answering machine) system or other well-known human machine interaction processes. The voice prompt, played by the speaker  121 , also may be echoed in the voice data generated by the microphone  128 . The VR system may get confused by detecting the echo in signal d(n) as the voice response of the near end user. The VR system needs to operate on the voice response from the near end user.  
           [0012]    Therefore, there is a need for an improved echo cancellation system.  
         SUMMARY  
         [0013]    Generally stated, a method and an accompanying apparatus provides for an improved echo cancellation system. The system includes a double talk detector configured for detecting a double talk condition by monitoring voice energy in a first frequency band. An adaptive filter is configured for producing an echo signal based on a set of coefficients. The adaptive filter holds the set of coefficients constant when the double talk detector detects the double talk condition. A microphone system is configured for inputting audible signals in a second frequency band. The second frequency band is wider and overlaps the first frequency band and the echo signal is used to cancel echo in the input signal. A loud speaker is configured for playing voice data in a third frequency band essentially equal to a difference of the first and second frequency bands. The first and third frequency bands essentially makeup the second frequency band. A control signal controls the adaptive filter, to hold the set of coefficients constant, based on whether the double talk detector detects the double talk condition. 
       
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
       [0014]    The features, objects, and advantages of the disclosed embodiments will become more apparent from the detailed description set forth below when taken in conjunction with the drawings in which like reference characters identify correspondingly throughout and wherein:  
         [0015]    [0015]FIG. 1 illustrates a block diagram of an echo canceller system;  
         [0016]    [0016]FIG. 2 illustrate partitioning of a voice recognition functionality between two partitioned sections such as a front-end section and a back-end section;  
         [0017]    [0017]FIG. 3 depicts a block diagram of a communication system incorporating various aspects of the disclosed embodiments.  
         [0018]    [0018]FIG. 4 illustrates partitioning of a voice recognition system in accordance with a co-located voice recognition system and a distributed voice recognition system;  
         [0019]    [0019]FIG. 5 illustrates various blocks of an echo cancellation system in accordance with various aspects of the invention; and  
         [0020]    [0020]FIG. 6 illustrates various frequency bands used for sampling the input voice data and the limited band of the speaker frequency response in accordance with various aspects of the invention.  
     
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT  
       [0021]    Generally stated, a novel and improved method and apparatus provide for an improved echo cancellation system. The echo cancellation system may limit the output frequency band of the speaker of the device. The output of the microphone of the device is expanded to include a frequency band larger than the limited frequency band of the speaker. An enhanced double talk detector detects the signal energy in a frequency band that is equal to the difference of the limited frequency band of the speaker and the expanded frequency band of the microphone. The adaptation of an adaptive filter in the echo cancellation system is controlled in response to the double talk detection. The parameters of the adaptive filter are held at a set of steady values when a double talk condition is present. The near end user of the device may be in communication with a far end user.  
         [0022]    The device used by the near end user may utilize a voice recognition (VR) technology that is generally known and has been used in many different devices. A VR system may operate in an interactive environment. In such a system, a near end user may respond with an audio response, such as voice audio, to an audio prompt, such as a voice prompt, from a device. The voice generated by the speaker of the device, therefore, may be a voice prompt in an interactive voice recognition system utilized by the device. The improved echo cancellation system removes the echo generated by the voice prompt when the near end user is providing a voice response while the voice prompt is being played by the speaker of the device. The device may be a remote device such as a cellular phone or any other similarly operated device. Therefore, the exemplary embodiments described herein are set forth in the context of a digital communication system. While use within this context is advantageous, different embodiments of the invention may be incorporated in different environments or configurations. In general, various systems described herein may be formed using software-controlled processors, integrated circuits, or discrete logic. The data, instructions, commands, information, signals, symbols, and chips that may be referenced throughout are advantageously represented by voltages, currents, electromagnetic waves, magnetic fields or particles, optical fields or particles, or a combination thereof. In addition, the blocks shown in each block diagram may represent hardware or method steps.  
         [0023]    Referring to FIG. 2, generally, the functionality of VR may be performed by two partitioned sections such as a front-end section  101  and a back-end section  102 . An input  103  at front-end section  101  receives voice data. The microphone  128  may originally generate the voice data. The microphone through its associated hardware and software converts the audible voice input information into voice data. The input voice data to the back end section  102  may be the voice data d(n) after the echo cancellation. Front-end section  101  examines the short-term spectral properties of the input voice data, and extracts certain front-end voice features, or front-end features, that are possibly recognizable by back-end section  102 .  
         [0024]    Back-end section  102  receives the extracted front-end features at an input  105 , a set of grammar definitions at an input  104  and acoustic models at an input  106 . Grammar input  104  provides information about a set of words and phrases in a format that may be used by back-end section  102  to create a set of hypotheses about recognition of one or more words. Acoustic models at input  106  provide information about certain acoustic models of the person speaking into the microphone. A training process normally creates the acoustic models. The user may have to speak several words or phrases for creating his or her acoustic models.  
         [0025]    Generally, back-end section  102  compares the extracted front-end features with the information received at grammar input  104  to create a list of words with an associated probability. The associated probability indicates the probability that the input voice data contains a specific word. A controller (not shown), after receiving one or more hypotheses of words, selects one of the words, most likely the word with the highest associated probability, as the word contained in the input voice data. The system of back end  102  may reside in a microprocessor. The recognized word is processed as an input to the device to perform or respond in a manner consistent with the recognized word.  
         [0026]    In the interactive VR environment, a near end user may provide a voice response to a voice prompt from the device. The voice prompt may or may not be generated by the VR system. The voice prompt from the device may last for a duration. While the voice prompt is playing, the user may provide the voice response. The microphone  128  picks up both the voice prompt and the voice response. As a result, the input voice data  103  is a combination of the voice prompt and the user voice response. As a result, the input voice data  103  may include a more complex set of voice features than the user voice input alone. When the user voice features are mixed with other voice features, the task of extracting the user voice features is more difficult. Therefore, it is desirable to have an improved echo cancellation system in an interactive VR system.  
         [0027]    The remote device in the communication system may decide and control the portions of the VR processing that may take place at the remote device and the portions that may take place at a base station. The base station may be in wireless communication with the remote device. The portion of the VR processing taking place at the base station may be routed to a VR server connected to the base station. The remote device may be a cellular phone, a personal digital assistant (PDA) device, or any other device capable of having a wireless communication with a base station. The remote device may establish a wireless connection for communication of data between the remote device and the base station. The base station may be connected to a network. The remote device may have incorporated a commonly known micro-browser for browsing the Internet to receive or transmit data. The wireless connection may be used to receive front end configuration data. The front end configuration data corresponds to the type and design of the back end portion. The front end configuration data is used to configure the front portion to operate correspondingly with the back end portion. The remote device may request for the configuration data, and receive the configuration data in response. The configuration data indicates mainly filtering, audio processing, etc, required to be performed by the front end processing.  
         [0028]    The remote device may perform a VR front-end processing on the received voice data to produce extracted voice features of the received voice data in accordance with a programmed configuration corresponding to the design of the back end portion. The remote device through its microphone receives the user voice data. The microphone coupled to the remote device takes the user input voice, and converts the input into voice data. After receiving the voice data, and after configuring the front end portion, certain voice features in accordance with the configuration are extracted. The extracted features are passed on to the back end portion for VR processing.  
         [0029]    For example, the user voice data may include a command to find the weather condition in a known city, such as Boston. The display on the remote device through its micro-browser may show “Stock Quotes|Weather|Restaurants|Digit Dialing|Nametag Dialing|Edit Phonebook” as the available choices. The user interface logic in accordance with the content of the web browser allows the user to speak the key word “Weather”, or the user can highlight the choice “Weather” on the display by pressing a key. The remote device may be monitoring for user voice data and the keypad input data for commands to determine that the user has chosen “weather.” Once the device determines that the weather has been selected, it then prompts the user on the screen by showing “Which city?” or speaks “Which city?”. The user then responds by speaking or using keypad entry. The user may begin to speak the response while the prompt is being played. In such a situation, the input voice data, in addition to the user input voice data, includes voice data generated by the voice prompt, in a form of feed back to the microphone from the speaker of the device. If the user speaks “Boston, Mass.”, the remote device passes the user voice data to the VR processing section to interpret the input correctly as a name of a city. In return, the remote device connects the micro-browser to a weather server on the Internet. The remote device downloads the weather information onto the device, and displays the information on a screen of the device or returns the information via audible tones through the speaker of the remote device. To speak the weather condition, the remote device may use text-to-speech generation processing. The back end processings of the VR system may take place at the device or at VR server connected to the network.  
         [0030]    In one or more instances, the remote device may have the capacity to perform a portion of the back-end processing. The back end processing may also reside entirely on the remote device. Various aspects of the disclosed embodiments may be more apparent by referring to FIG. 3. FIG. 3 depicts a block diagram of a communication system  200 . Communication system  200  may include many different remote devices, even though one remote device  201  is shown. Remote device  201  may be a cellular phone, a laptop computer, a PDA, etc. The communication system  200  may also have many base stations connected in a configuration to provide communication services to a large number of remote devices over a wide geographical area. At least one of the base stations, shown as base station  202 , is adapted for wireless communication with the remote devices including remote device  201 . A wireless communication link  204  is provided for communicating with the remote device  201 . A wireless access protocol gateway  205  is in communication with base station  202  for directly receiving and transmitting content data to base station  202 . The gateway  205  may, in the alternative, use other protocols that accomplish the same or similar functions. A file or a set of files may specify the visual display, speaker audio output, allowed keypad entries and allowed spoken commands (as a grammar). Based on the keypad entries and spoken commands, the remote device displays appropriate output and generates appropriate audio output. The content may be written in markup language commonly known as XML HTML or other variants. The content may drive an application on the remote device. In wireless web services, the content may be up-loaded or down-loaded onto the device, when the user accesses a web site with the appropriate Internet address. A network commonly known as Internet  206  provides a land-based link to a number of different servers  207 A-C for communicating the content data. The wireless communication link  204  is used to communicate the data to the remote device  201 .  
         [0031]    In addition, in accordance with an embodiment, a network VR server  206  in communication with base station  202  directly may receive and transmit data exclusively related to VR processing. Server  206  may perform the back-end VR processing as requested by remote station  201 . Server  206  may be a dedicated server to perform back-end VR processing. An application program user interface (API) provides an easy mechanism to enable applications for VR running on the remote device. Allowing back-end processing at the sever  206  as controlled by remote device  201  extends the capabilities of the VR API for being accurate, and performing complex grammars, larger vocabularies, and wide dialog functions. This may be accomplished by utilizing the technology and resources on the network as described in various embodiments.  
         [0032]    A correction to a result of back end VR processing performed at VR server  206  may be performed by the remote device, and communicated quickly to advance the application of the content data. If the network, in the case of the cited example, returns “Bombay” as the selected city, the user may make correction by repeating the word “Boston.” The word “Bombay” may be in an audio response by the device. The user may speak the word “Boston” before the audio response by the device is completed. The input voice data in such a situation includes the names of two cities, which may be very confusing for the back end processing. However, the back end processing in this correction response may take place on the remote device without the help of the network. In alternative, the back end processing may be performed entirely on the remote device without the network involvement. For example, some commands (such as spoken command “STOP” or keypad entry “END”) may have their back end processing performed on the remote device. In this case, there is no need to use the network for the back end VR processing, therefore, the remote device performs the front end and back end VR processings. As a result, the front end and back end VR processings at various times during a session may be performed at a common location or distributed.  
         [0033]    Referring to FIG. 4, a general flow of information between various functional blocks of a VR system  300  is shown. A distributed flow  301  may be used for the VR processing when the back end processing and front end processings are distributed. A co-located flow  302  may be used when the back end and front end processings are co-located. In the distributed flow  301 , the front end may obtain a configuration file from the network. The content of the configuration file allows the front end to configure various internal functioning blocks to perform the front end feature extraction in accordance with the design of the back end processing. The co-located flow  302  may be used for obtaining the configuration file directly from the back end processing block. The communication link  310  may be used for passing the voice data information and associated responses. The co-located flow  302  and distributed flow  301  may be used by the same device at different times during a VR processing session.  
         [0034]    Referring to FIG. 5, various blocks of an enhanced echo cancellation system  400  is shown in accordance with various embodiments of the invention. A speaker  401  outputs the audio response of an audio signal  411 . The bandwidth of the audio signal  411  is limited in accordance with various aspects of the invention. For example, the bandwidth may be limited to zero to 4 kHz. Such a bandwidth is sufficient for producing a quality audio response from the speaker  401  for human ears. The audio signal  411  may be generated from different sources. For example, the audio signal  411  may be originated from a far end user in communication with a near end user of the device or a voice prompt in an interactive VR system utilized by the device. The far end audio signal f(n)  495  in digital domain may be processed in an ADC  499  with a limited bandwidth in accordance with various aspects of the invention. The far end signal  411  with a limited bandwidth is produced. For example, if the sampling frequency of the ADC  499  is set to 8 kHz, the audio signal  411  may have a bandwidth of approximately 4 kHz. The signal f(n)  495  may have been received from a voice decoder  498 . A unit  410  may produce the input to voice decoder  498  in a form of encoded and modulated signal. The unit  410  may include a controller, a processor, a transmitter and a receiver. The signal decoded by voice decoder  498  may be in a form of audio PCM samples. Normally, the PCM samples data rate is 8K samples per second in traditional digital communication systems. The audio PCM samples are converted to analog audio signal  411  via 8 KHz ADC  499  and the played by speaker  401 . The produced audio, therefore, is band limited in accordance with various aspects of the invention.  
         [0035]    The audio signal  411  also may have been produce by a voice prompt in a VR system. The unit  410  may also provide the data voice packets to the voice decoder  498 . The audio signal  411  is then produced which carries the voice prompt. The data voice packets may be encoded off-line from band limited speech and thus can be used to reproduce the band limited voice prompts. The voice prompt may be generated by the network or by the device. The voice prompt may also be generated in relation to operation of an interactive VR system. The VR system in part or in whole may reside in the device or the network. Under any condition, the produced audio signal  411  is band limited in accordance with various aspects of the invention.  
         [0036]    A microphone  402  picks up a combination of the audio response of the near end user and the audio from the speaker  401 . The audio from the speaker  401  may pass through the acoustic echo channel  497 . An ADC  403  converts the received audio to voice data  404 . The voice response of the near end user includes various frequency components, for example from zero to 8 kHz. The sampling frequency of ADC  403  is selected such that the voice data  404  includes frequency components in a frequency band greater than the limited frequency band of audio signal  411  played by speaker  401 , in accordance with various aspects of the invention. For example, a sampling frequency of 16 kHz may be selected. Therefore, the frequency band of the voice data  404  is approximately 8 kHz, which is greater than the 4 kHz frequency band of audio signal  411  in accordance with the example.  
         [0037]    A double talk detector  406  receives the voice data  404 . The voice data  404  includes the audio produced by speaker  401  and the near end user speaking into the microphone  402 . Since the bandwidth of the audio signal  411  is limited and less than the bandwidth of the voice data  404 , double talk detector  406  may determine if any frequency components in a frequency band equal to the difference between the limited frequency band of audio signal  411  and the entire frequency band of voice data  404  is present. If a frequency component is present, its presence is contributed to the near end person speaking into the microphone. Therefore, in accordance with various aspects of the invention, a double-talk condition of the near end user and audio from the speaker is detected.  
         [0038]    Referring to FIG. 6, a graphical representation of the different frequency bands is shown. The frequency response of the audio signal  411  may be shown by graph  501 . The frequency response of the voice data  404  is shown by the graph  502 . The frequency band  503  is the difference between the frequency response  501  and  502 . If any signal energy is present in the frequency band  503 , the signal energy is contributed to the near end user speaking into the microphone  402 . The signal energy may be contributed to any frequency component in the frequency band  503  or to any combination of frequency components or all the components. Double talk detector  406  may use a band pass filter to isolate the frequency components in the frequency band  503 . A comaparator may be used to compare an accumulated energy to a threshold for detecting whether any noticeable energy is present in the frequency band  503 . If the detected energy is above the threshold, a double talk condition may be present. The threshold may be adjusted for different conditions. For example, the conditions in a crowded noisy place, an empty room and in a car may be different. Therefore, the threshold may also be different for different conditions. A configuration file loaded in the system may change the threshold from time to time.  
         [0039]    An anti aliasing filter  405 , usually a low pass filter, followed by a sub-sampling block may be used to down-sample the voice data  404  by a factor of 2. The down sampling may be performed at different factors also. For example, if a down sampling factor of 3 is used with an ADC  403  sampling of 24 KHz, the frequency band  503  may be from 4 KHz to 12 KHz accordingly. An adaptive filter  420  produces a replica of acoustic echo signal e 1 (n) to be subtracted from the filtered voice data in an adder  407 . The coefficients of the adaptive filter  420  may be changing based on the signal  409 , representing d(n). The operation of adaptive filter  420  may be controlled by a control signal  421  in accordance with various embodiments of the invention. Control signal  421  is produced based on whether double talk detector  406  detects a double talk condition. If a double talk condition is detected, control signal  421  holds the coefficients of the adaptive filter  420  to reproduce the echo signal e 1 (n) in accordance with various aspects of the invention. The coefficients of the adaptive filter  420  may be held constant at the time of the double talk detection. In a double talk condition, adaptive filter  420  may not change the coefficients. When the double talk condition is not present, control signal  421  allows the adaptive filter  420  to operate. The coefficients of the adaptive filter  420  may be adjusted. The echo signal e 1 (n), whether the coefficients are being held constant or changing, is used in the summer  407 . The summer  407  produces the processed voice data  409 . The processed voice data is used by unit  410  as a response to a far end user or a voice response to a voice prompt. In case of using a VR system, the unit  410  may use the processed voice data  409  for VR operation. Therefore, an improved echo cancellation system is provided in accordance with various aspects of the invention. The improved system allows the far end user to hear the near end user more clearly without presence of an echo and diminishing the effect of the voice prompt in the received voice data. The received voice data may be processed by the VR system more effectively when the undesired components have been removed.  
         [0040]    The previous description of the preferred embodiments is provided to enable any person skilled in the art to make or use the present invention. The various modifications to these embodiments will be readily apparent to those skilled in the art, and the generic principles defined herein may be applied to other embodiments without the use of the inventive faculty.