Patent Publication Number: US-2007096964-A1

Title: System and method for efficient upsampled signal processing

Description:
BACKGROUND  
      1. Technical Field  
      This invention relates generally to a system and method for efficient upsampling and filtering, and more specifically to a system and method for upsampling and filtering without the use of a low-pass filter.  
      2. Background Art  
      Digital audio systems rely upon analog to digital converters for their operation. Analog to digital converters allow a conventional analog waveform, like acoustic energy for example, to be sampled and quantized into a data stream of binary numbers. The conversion from an analog waveform to digital number sequence allows computers and microprocessors to accurately and reliably store, recall, transform and reproduce audio sound.  
      During this sampling process, the analog wave is effectively “chopped up” into discrete digital values. The sampling process, by its very nature, introduces quantization noise into the analog to digital system, as the formerly smooth analog wave essentially becomes a piece-wise linear approximation of that wave. In addition to the quantization noise, a phenomenon known as “aliasing” causes multiples of the analog wave to appear as duplicate images above the frequency of interest. These duplicates will compromise the quality of signal when the discrete digital values are converted back into an analog wave.  
      For example, an audio signal having frequency components ranging from 0 to 22 kHz must be sampled, in accordance with Nyquist&#39;s Theorem, at a frequency of at least 44 kHz. In addition to quantization noise, duplicate images will appear at multiples of the original analog frequencies. A first duplicate appears from 22 kHz to 44 kHz, a second duplicate appears at 44 kHz to 66 kHz, and so forth.  
      To eliminate quantization and oversampling noise, prior art systems employed very high-order, complex low-pass filters to eliminate frequency components above the desired frequency. Using the example in the preceding paragraph, a designer may employ a fourth or fifth order Chebychev filter to try and eliminate all frequency components above 22 kHz. The problem with these filters is not only are they expensive, but they often became unstable and introduce other forms of noise into the system.  
      To rectify these problems, designers have begun using a process known as oversampling. In oversampling, rather than sampling the analog signal at twice the maximum frequency of interest, the system may sample the analog signal at four, eight or even sixteen times the maximum frequency of interest. Continuing with the example from above, the 0-22 kHz signal may be sampled at 88 kHz, 176 kHz or 352 kHz. This oversampling causes the duplicate images to be pushed to higher and higher frequencies. For example, at 4×oversampling, the first image appears at 66-88 kHz, rather than 22-44 kHz. Additionally, as the oversampling rate increases, the quantization noise becomes spread across a larger spectrum, thereby increasing the signal to noise ratio. As a result, simpler, lower-order low-pass filters may be used to eliminate the duplicates.  
      Designers also realized that a similar process, upsampling, could be carried out in the digital domain to increase the overall quality of the signal. The process of upsampling is similar to oversampling, in that a digital data stream of one frequency is increased to a higher frequency. In upsampling, the process is carried out by inserting additional samples between the initial samples. These additional samples may be either zeroes or interpolated values between adjoining data points.  
      As with oversampling, in conventional upsampling systems an upsampled data stream is passed through a low-pass filter to remove any duplicate images introduced by the upsampling process. The filtered, upsampled data stream may then be passed on to other signal processing modules, circuits and components.  
      The problem with these prior art upsampling systems is that the low-pass digital filter requires both time and processing power when in operation. This time and processing power, especially in the realm of battery-powered, mobile electronics, can come at the expense of other functions. Additionally, the processing power required to implement these filters is “multiply and accumulate” processing power. In the world of microprocessors, performance is sometimes determined by how many multiply and accumulate (MAC) operations the processor can perform in one second. As there is a finite amount of MAC operations that a device may perform, for every one that must be dedicated to filtering, one less is available for other functions.  
      There is thus a need for an improved upsampling system with increased efficiency. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
      The accompanying figures, where like reference numerals refer to identical or functionally similar elements throughout the separate views and which together with the detailed description below are incorporated in and form part of the specification, serve to further illustrate various embodiments and to explain various principles and advantages all in accordance with the present invention.  
       FIG. 1  illustrates a prior art upsampling system employing a low-pass filter.  
       FIG. 2  illustrates an efficient upsampling system and method in accordance with the invention.  
       FIG. 3  illustrates a method of efficiently upsampling and filtering in accordance with the invention.  
       FIG. 4  illustrates an electronic device having an upsampling and filtering system in accordance with the invention. 
    
    
      Skilled artisans will appreciate that elements in the figures are illustrated for simplicity and clarity and have not necessarily been drawn to scale. For example, the dimensions of some of the elements in the figures may be exaggerated relative to other elements to help to improve understanding of embodiments of the present invention.  
     DETAILED DESCRIPTION OF THE INVENTION  
      Before describing in detail embodiments that are in accordance with the present invention, it should be observed that the embodiments reside primarily in combinations of method steps and apparatus components related to efficient upsampling, which is useful in front end encoders. Accordingly, the apparatus components and method steps have been represented where appropriate by conventional symbols in the drawings, showing only those specific details that are pertinent to understanding the embodiments of the present invention so as not to obscure the disclosure with details that will be readily apparent to those of ordinary skill in the art having the benefit of the description herein.  
      It will be appreciated that embodiments of the invention described herein may be comprised of one or more conventional processors and unique stored program instructions that control the one or more processors to implement, in conjunction with certain non-processor circuits, some, most, or all of the functions of efficient upsampling and filtering described herein. The non-processor circuits may include, but are not limited to, a radio receiver, a radio transmitter, signal drivers, clock circuits, power source circuits, and user input devices. Further, it is expected that one of ordinary skill, notwithstanding possibly significant effort and many design choices motivated by, for example, available time, current technology, and economic considerations, when guided by the concepts and principles disclosed herein will be readily capable of generating such software instructions and programs and ICs with minimal experimentation.  
      A preferred embodiment of the invention is now described in detail. Referring to the drawings, like numbers indicate like parts throughout the views. As used in the description herein and throughout the claims, the following terms take the meanings explicitly associated herein, unless the context clearly dictates otherwise: the meaning of “a,” “an,” and “the” includes plural reference, the meaning of “in” includes “in” and “on.” Further, relational terms such as first and second, top and bottom, and the like may be used solely to distinguish one entity or action from another entity or action without necessarily requiring or implying any actual such relationship or order between such entities or actions.  
      The present invention offers an efficient method and system for upsampling and filtering signals. Specifically, the method and system of this invention allow a signal to be upsampled and processed without a low-pass filter where a predetermined characteristic of the signal is known. One application well suited for the system and method of this invention is an encoder module in a portable electronic device like a radiotelephone, music player or video player.  
      In one embodiment of the invention a controller works in conjunction with an upsampler. The upsampler inserts data between existing samples of an incoming data stream. For instance, if the incoming sample stream is a sampled audio signal, the upsampler may upsample by inserting one or more zeroes between each sample of the sampled audio signal. In so doing, the upsampler increases the sample frequency of the overall data stream.  
      In many applications, predetermined characteristics are known about the incoming signals. For instance, in a telephone-based application, not all audible frequencies are transmitted from one telephone to the next. In conventional applications like a public switched telephone network for example, only frequencies ranging from 300 Hz to 3 kHz are transmitted. The controller of this invention takes advantage of this predetermined characteristic and adjusts certain post processing operations accordingly to eliminate the need of any low-pass filtering after the upsampling stage. The net result is increase overall efficiency, as both less memory and less multiply and accumulate commands are required in signal processing. This increased efficiency leads to longer battery life and more processing power that may be dedicated to other circuits and modules.  
      Turning briefly to  FIG. 1 , illustrated therein is a conventional upsampling system  100  that may be employed with an encoder. The system  100  includes an upsampling module  101  and an encoder  104 . The upsampling module  101  includes an upsampler  102  and a low-pass filter  103 . As noted above, when an incoming signal  109  having a sampled frequency of Fs is upsampled, a number of samples are introduced between each sample in the incoming signal  109 . The result is an upsampled signal  110  having an upsampled frequency Fs′. By way of example, if the sample frequency is 4 kHz, and two zeroes are inserted between each sample, the upsampled frequency would be 12 kHz.  
      As also noted above, the upsampling causes duplicate images to appear at frequencies of above the frequency of interest. To eliminate these aliased images, the low-pass filter  103  is used to cut out duplicate images above the frequency of interest. Continuing the example from above, assuming a 4 kHz incoming signal and an upsampling by 3, the low-pass filter  109  would remove all duplicate images above 2 kHz. (After upsampling, the upsampled sample frequency is 12 kHz. Per Nyquist&#39;s Theorem, 6 kHz is the maximum available frequency. Dividing 6 kHz by M, where M=3, yields 2 kHz.)  
      In more general terms, for an incoming signal  109  with a sample frequency Fs, the frequency of interest, staying in accordance with Nyquist&#39;s Theorem, could be no more than 0 to Fs/2. Normalized, this may be represented as ranging from 0 to π. The period of the incoming signal may be expressed as 2π. When the upsampler  102  upsamples by a factor of M, the frequency of interest gets “compressed” by a factor of M. It is thus periodic with a period of 2π/M. The low-pass filter  103 , to effectively eliminate the images, must remove all frequencies above π/M.  
      When used in an encoding application, the upsampled, low-pass filtered signal  111  may then be fed into a bank of band-pass filters  105 - 108  for processing, thereby yielding a series of upsampled, low-pass filtered, band-pass filtered signals  112 - 115 . Systems like that shown in  FIG. 1  are often employed in front-end modules of consumer electronics.  
      Turning now to  FIG. 2 , illustrated therein is a system for efficient upsampling  200  in accordance with the invention. The system  200 , which may be used for processing a digitized signal and is suitable for encoder applications, includes a controller  216 , and upsampler  202  and a plurality of filters  205 - 208  coupled to the upsampler  202 . Each filter  205 - 208  has a corresponding output  212 - 215 . The modules of the system  200  may be employed or effected in the front end module of a signal encoder. Alternatively, they may be disposed in a stand-alone module, or integrated into one component, like a microprocessor, application specific integrated circuit (ASIC) or digital signal processor.  
      The upsampler  202  is capable of upsampling an incoming signal  209  by a factor of M. In one embodiment, the upsampling factor M is an integer. As the incoming signal  209  is a digitized signal including a series of samples, the upsampler  202  upsamples by inserting at least one zero between each of the series of samples, thereby converting the digitized signal sampling frequency to an “upsampled” digitized signal sampling frequency. It will be clear to those of ordinary skill in the art having the benefit of this disclosure that values other than zeroes may be inserted between each of the series of samples. For example, the average values of the adjoining samples may be inserted rather than samples with a zero value.  
      For signals characterized by a predetermined characteristic, like a maximum frequency of interest for example, the controller  216  eliminates the need for a low-pass filter by adjusting at least one filter output, e.g. output  215  of filter  208  based upon the predetermined characteristic. As such, no low-pass filter is coupled between the plurality of filters  205 - 208  and the upsampler  202 .  
      For instance, where the predetermined characteristic is a maximum frequency of interest, the controller  216  may adjust one of the filters, e.g. filter  208 , by either deactivation, i.e. not running the filter, or by setting it&#39;s output  215  to zero. Note that in one embodiment, each of the plurality of filters  205 - 208  is a digital filter that may be implemented mathematically within a processing component. As such, the “deactivating” may be deliberately not running that filter within the processing component, or not applying the filter to the upsampled signal. Where output values of that filter are required by post processing circuits, the controller my supply a series of zeroes. It will be clear to those of ordinary skill in the art that the deactivating or setting to zero may be done in a variety of ways, but the net result is the same: the output, if and when required, is zero.  
      Taking the telephone example from above, where the predetermined characteristic of the incoming signal  209  is a maximum frequency of 3 kHz, the controller assumes that an upsampled, low-pass filtered signal would cause zeroes to be forthcoming from some of the plurality of filters  205 - 208 , as frequencies above the maximum frequency of interest, scaled by the upsampling factor, would have been eliminated by the low-pass filter in a conventional system. The invention takes advantage of this knowledge and eliminates the need for any memory allocation or multiply and accumulate commands that would be required by a low-pass filter. The invention does so with the controller  216 , which can set at least one filter, e.g. filter  208 , to zero by not running the filter or forcing its output  215  to be zero.  
      Where the plurality of filters  205 - 208  has N filters, the number of filters that may be deactivated or set to zero depends upon the predetermined characteristic. In one embodiment, suitable for use in an encoder, each of the filters  205 - 208  is a band-pass filter having a pass bandwidth associated therewith. In such a scenario, the pass bandwidth associated with each filter, for the filters to evenly cover the spectrum, is the upsampled digitized sampling frequency divided by 2*N. The controller  216  is then be able to adjust the outputs of the filters having pass band frequencies greater than the upsampled digitized signal sampling frequency divided by M by deactivating those filters or setting their outputs to zero.  
      The operation of this system  200 , set forth in general above, is perhaps more easily understood by way of an example. The following example is illustrative only and is presented to better explain the operation of the components in the system  200 . Presume for the purposes of this example that the plurality of filters  205 - 208  is a bank of eight filters, each of which support a sampling rate of at least 44.1 kHz. Also presume that the input signal  209  has a sample frequency, Fs, of 11.025 kHz, as may be the case in an audio application. The upsampler  202  upsamples by inserting three zeroes between every sample of the incoming signal  209 , thereby giving the upsampled signal  210  an upsampled digitized signal sampling frequency, Fs′, of 44.1 kHz. (M equals 4.) Presume that the input signal  209  has frequency components in the entire bandwidth, represented normally as extending from −π to π.  
      The upsampler  202  begins the process by receiving the incoming signal  209  and inserting zeroes between each of the samples. Where the incoming signal  209  has samples represented x[k], the upsampler output signal  210  may be represented:  
         Z   ⁡     [   k   ]       =     x   ⁡     [   k   ]           
     where     
       k   =     4   *   n         
     and     
         n   ⁢           ⁢   is   ⁢           ⁢   an   ⁢           ⁢   interger     ,     
     ⁢     and   ⁢           =     0   ⁢           ⁢     otherwise   .             
 
      The upsampled signal  210  has a bandwidth of interest that has now been compressed by M, which in this case is a factor of 4. As such, the bandwidth is now −π/4 to π/4. The upsampled signal  210  also includes aliased duplicate images that fall within the −π to −π/4 and π/4 to π bandwidths. With prior art systems, a “brick wall” low-pass filter, with a pass band of π/4 would be needed to remove the duplicates. However, as the predetermined characteristic is known, specifically a maximum frequency of interest at π/4, the controller  216  is able to deactivate or otherwise set to zero certain band-pass filters, thereby eliminating the duplicates.  
      As there are eight band-pass filters (represented in  FIG. 2  by filters  205 - 208 ) being employed, they have the following pass band frequencies: −π/8 to π/8; π/8 to π/4; π/4 to 3π/8; 3π/8 to π/2; π/2 to 5π/8, 5π/8 to 3π/4 to 7π8; and 7π/8 to π. Since the predetermined characteristic, or maximum frequency of interest is π/4, all filters having pass bands over π/4 will be deactivate or set to zero by the controller  216  in accordance with the invention. In other words, the filters having the following pass bands will be set to zero or deactivated: π/4 to 3π/8; 3π/8 to π/2; π/2 to 5π/8, 5π/8 to 3π/4 to 7π/8; and 7π/8 to π. As noted above, the outputs of these filters may be set to zero, or in the alternative, as they may be mathematical filters applied by a processor, they need not even be run or applied to the upsampled signal  210 .  
      The lack of a low-pass filter increases efficiency by not only reducing the number of multiply and accumulate procedures, but by also reducing the number of operations accessing memory. For the preceding example, if a conventional low-pass filter would have had 40 taps, it would have required 40 multiply and accumulate commands in addition to 80 memory access operations to retrieve the samples and filter tap coefficients. Neglecting memory access operations, the invention saves at least 40*44100 multiply and accumulate procedures, which results in a savings of over a million instructions per second.  
      Note that if it is not possible to cause the outputs  212 - 215  to be forced to zero, the controller  216  may be coupled to a decoder or synthesis filter bank. The appropriate filters may be zeroed out at that point, thereby accomplishing the same effect.  
      Any audio component that utilizes an analysis filter bank, like a plurality of band-pass filters, to separate a signal into sub-bands, may only support a limited number of sampling frequencies. As such, the band-pass bandwidths may not correlate exactly to the maximum frequency of interest as in the preceding example. In such cases, most audio applications will accommodate a zeroing of the band-pass filter with a lower cut-off frequency just below the maximum frequency of interest. For example, if the maximum frequency of interest was 4 kHz, and the band-pass filters had band-pass frequencies of 0-3 kHz, 3-6 kHz, 6-9 kHz, the filters including and beyond the 3-6 kHz filter could be zeroed without significantly affecting audio performance.  
      Turning now to  FIG. 3 , illustrated therein is a method for processing a sampled signal in accordance with the invention. At step  301 , a sampled signal having a specific characteristic is received. At step  302 , the sampled signal is upsampled. As the sampled signal comprises a plurality of samples, the step of upsampling comprises inserting at least one additional sample between each of the plurality of samples. For example the step of upsampling may include upsampling by a factor of M, where M−1 additional samples are inserted between each of the plurality of samples. As noted above, in one embodiment, the at least one additional sample is a zero.  
      At step  303 , shown in dashed line because it is a step of omission, no low pass filter is applied to the upsampled signal. At step  304 , the upsampled, sampled signal is then delivered to a plurality of band-pass filters, as would be used to segregate the signal in an audio encoder application.  
      At step  305 , an output of at least one band-pass filter is adjusted based upon a specific characteristic of the upsampled, sampled signal. In one embodiment, the specific characteristic may be a frequency range of less than a predetermined sampling frequency. The step of adjusting an output may include either causing the output of at least one band-pass filter to be zero or deactivating at least one of the band-pass filters.  
      Turning now to  FIG. 4 , illustrated therein is one application for an efficient upsampling system and method in accordance with the invention. The illustrative embodiment of  FIG. 4  is that of an electronic device  400 . The electronic device  400  includes a microprocessor  401  that serves as the central brain of the device  400 . The electronic device  400  may be a radiotelephone, audio player, video player or other device.  
      The device  400  also includes an upsampling module  402  capable of converting a discrete data stream having a first sampling frequency associated therewith, perhaps received and sampled from radio frequency circuitry coupled to the microprocessor  401  for receiving and transmitting electromagnetic signals, into a discrete data stream having a second sampling frequency associated therewith. The upsampling module  402  converts the data stream from the first sampled frequency to the second sampled frequency, in one embodiment, by inserting zeroes between values in the discrete data stream. As noted above, the upsampling module  402  may be integrated into the microprocessor  401 . Alternatively, the upsampling module  402  may be effected in a separate component like a digital signal processor or the front end of an audio encoder.  
      A plurality of band-pass filters  403  are coupled to the upsampling module  402 . Each of the plurality of band-pass filters  403  has a unique pass band frequency associated therewith. A control module  404  is coupled to the plurality of band-pass filters. The control module  404  is capable of forcing the outputs of any of the band-pass filters  403  to zero, either by deactivating them or by causing them to yield a zero output.  
      The control module  404  executes an operation that either adjusts at least one band-pass filter output to zero based upon a predetermined criterion or deactivates at least one band-pass filter based upon a predetermined criterion. The predetermined criterion, for example, may be a maximum frequency of interest. In accordance with the invention, this action of the control module  404  allows the upsampling module to deliver its output to the plurality of band-pass filters  403  without the application of a low-pass filter, thereby saving memory and processing power.  
      In the foregoing specification, specific embodiments of the present invention have been described. However, one of ordinary skill in the art appreciates that various modifications and changes can be made without departing from the scope of the present invention as set forth in the claims below. Thus, while preferred embodiments of the invention have been illustrated and described, it is clear that the invention is not so limited. Numerous modifications, changes, variations, substitutions, and equivalents will occur to those skilled in the art without departing from the spirit and scope of the present invention as defined by the following claims.