Method of filtering high resolution digital signals and corresponding architecture of digital filter

Abstract: A method of filtering digital signals having a high dynamic range comprises the steps of: splitting the sampled input signal (S) into at least two portions; addressing each of said portions (S1,S2,...,Sm) to a respective program filter (2), and performing each filtering operation in parallel and independently; and reconstituting an output signal by summing together the digital outputs from each filter (2).

Description: This invention relates to a method of filtering digital signals having a high dynamic range, that is, signals encoded with a large number of bits. The invention also concerns a digital filter architecture obtained according to said method. The field of application of the invention is particularly, though not exclusively, related to digital filters of the non-recursive linear phase type, also known as FIR (Finite Impulse Response) filters, and the description hereinbelow will make reference to that field of application merely for convenience of illustration. Background Art As is well known, digital filters are devices operative to convert an input sampled signal into another sampled, output signal having predetermined frequency response characteristics. A sampled signal is understood to mean a digital coding comprising a predetermined number N of bits which determines the accuracy or resolution of the filter. Digital filters are used primarily in digital oscilloscopes, spectrum analyzers, and audio and video signal processors. Furthermore, the use of such filters is an ever expanding one, on account of the many advantages they afford over corresponding analog filters. For the same function, in fact, digital filters allow to achieve very narrow transmission bands and are also stabler, both over time and with respect to variations in power supply and operating temperature. In current practice, digital filters are implemented in the form of integrated circuit using digital multipliers and adders. Digital multipliers may be arranged to include a non-volatile memory structure of the so-called "look-up table" type, wherein the multiplication results of the input sample signals by the filter transfer function coefficients are stored. Such a structure is described, for example, in the article "30-MSamples/s Programmable Filter Processor", IEEE Journal of Solid-State Circuits, Vol.25, No. 6, December 1990, and in Italian Patent Application No.22954-A/88 by the same Applicant. While being in many ways advantageous, this prior approach still has the drawback discussed hereinbelow. If "n" is the number of bits used to sample the input signal, "N" the overall number of the filter coefficients, and "p" the number of bits required for storing the multiplication result of the samples by the coefficients, then the memory size is found to be 2nNp. It is clear from the above that to increase, even by a single bit, the sampled description of the input signal in order to increase the filter resolution, means to double the size of the memory. Let's consider, as as an example, a hypothetical transition from an 8-bit sampling, typical in current practice, to a n=12 bits sampling -- such as would be highly desirable in order to improve the filter performance with audio and video signals. An appropriate hypothetical memory structure for the purpose would have to be sixteen times as large as that required by the first-mentioned 8-bit coding. Such a memory would, therefore, take up an inordinate amount of space on an integrated circuit. In addition, it would have insufficient data accessing speed because of access time being heavily dependent on both the increased coding complexity and the memory size, specifically the number of bits per row. The prior art has failed to bring forth any useful solution to obviate the need for so great a memory expansion when improved filter accuracy or resolution is sought. A known solution is disclosed in the volume: "Radio and Electronic Engineer", vol. 46, No. 8/9, August 76, London, GB, pages 393-400; M.A. Bin Nun et al.: "A Modular Approach to the Hardware Implementation of Digital Filters". This document discloses a method of filtering digital signals which comprises the steps of: splitting the input sampled signal into at least two portions; addressing each of said signal portions to a respective program filter and performing each filtering operation independently and in parallel; reconstructing an output signal by summing together the digital outputs from each filter. The performances of this solution are limited by the fact that the overall filter section depends primarily on the structure of the basic module and the manner in which the computing algorithm is processed. The technical problem underlying this invention is to provide a filtering method and a digital filter for use therewith, having respectively such functional and structural features as to enable high-resolution processing of digital sampled signals encoded with a large number of bits, thereby to overcome the limitations of state-of-art approaches. Summary of the Invention The idea on which this invention stands is that of splitting the coding of the sampled signal into portions of at least eight bits each, and filtering each portion independently of the other by means of respective digital filters, to then reconstitute the output sampled signal. On the basis of this idea, the technical problem is solved by a filtering method as indicated above and defined in the characterizing portion of Claim 1. The technical problem is also solved by a digital filter architecture as defined in Claim 5. The features and advantages of the inventive method will become more clearly apparent from the following detailed description of an exemplary, non-limitative embodiment thereof, given with reference to a digital filter architecture illustrated by the accompanying drawings. Brief Description of the Drawings Figure 1 shows diagramatically a digital filter architecture embodying this invention. Figure 2 is a diagram of a second embodiment of the filter shown in Figure 1. Detailed Description With reference to the drawing figures, generally and diagramatically shown at 1 is a digital filter architecture which embodies this invention and is effective to filter digital signals having high dynamic ranges. Basically, the architecture 1 can operate with digital signals S encoded with at least k bits, wherein k may be any number between eight and sixteen. This architecture 1 has a predetermined number of programmable digital filters 2 denoted by the acronym PFP. Such filters 2 have a structure known per se and described, for example, in Italian Patent Application 22954-A/88 by this Applicant. Each filter 2 receives a given eight-bit sampled signal x(n), and outputs another sampled signal y(n) having a predetermined frequency response characteristic. The filter 2 is a so-called linear phase FIR (Finite Impulse Response) type, which means that, at each discrete time, the input signal x(n) and the output signal y(n) will be tied by the following relation: from which it may be deduced that the output sample y(n) is only dependent on the present and past input samples. The coefficients a(i) are a finite number N and are obtainable from the response to the filter pulse. Furthermore, the filter 2 may be identified by a so-called transfer function H(z), expressed versus a complex variable z and defined as the ratio H(z) of the transform Z of a sequence of output sampled signals y(nT) to the transform Z of a sequence of input sampled signals x(nT). The circuit-wise synthesis of that transfer function practically enables implementation of the filter 2 as a single PFP integrated circuit, whereby for each digital input x(n), an output y(n) given by equation (1) would be produced. The signal inputs to the filter 2 are denoted by A0,...,A7, whilst the digital outputs are denoted by O0,...,O15. Advantageously, the architecture 1 includes a number m=2n of filters 2, where n=(k-8), in parallel with one another. Each filter 2 has the same transfer function. The architecture of this invention exploits the ability of each filter 2 to handle a digital signal quantized to eight bits. To this purpose, the input signal S is split into m portions, each represented by an eight-bit coding. As shown in Figure 1, the signal S is applied to a decoder D which selects the eight most significant bits and outputs a signal S1. This signal S1 is a first component of the original signal S, and is applied directly to the inputs A0,...,A7 of a first filter 2. The signal S1 is also applied to a adder block 4 which receives on another input the bit having the greatest weight of the least significant portion of the signal S resulting from the selection performed in block D. As the embodiment shown in Figure 1 represents an instance where the signal S has a nine-bit coding, a single bit a0, the very least significant bit in the S coding, will be transferred from the decoder D to the adder 4. Thus, at the output of the adder 4, a second, eight-bit signal S2 is produced which will be addressed to the input side of a corresponding second PFP filter. The outputs from the first and second filters are summed together in a block 5 to reconstitute the output signal correctly. To the purpose of showing a more general aspect of the inventive concept, it is observed here that when a higher dynamic range is provided for the input signal S, i.e. the signal is encoded with a larger number of bits, it will be sufficient to merely increase the numbers of the filters 2 and the adders 4 arranged in parallel. The filtering method of this invention provides for the ouput sample signal to be found by the following computation: where, N is the filter size, k is the dynamic range of the input signal, and m is equal to 2(k-8). To further elucidate how the inventive method can be carried out, the instance will now be discussed of an input signal S being encoded with ten bits. As a consequence, S is a digital signal having a value of from 0 and 1020 and which should be splitted, in accordance with the inventive method, into m=2(10-8)=4 signal components, each with an eigh-bit coding. If S is expressed as a sequence of bits aj, each with an indication of its respective weight, it is found that: S = a9 a8 a7 a6 a5 a4 a3 a2 a1 a0 Hence, the four components into which that signal can be splitted are: S1 = INT [S/4] S2 = INT [S/4] + a2 S3 = INT [S/4] + a1 S4 = INT [S/4] + a0 where, INT is a symbol representing the integer value of the ratio in brackets. Each component is encoded with eight bits, and is addressed to the input side of a corresponding filter 2 to be filtered in accordance with the specifications of a predetermined transfer function. The outputs of the various parallel-connected filters 2 are summed together to reconstitute the proper coding sequence for the output signal. In an alternative embodiment shown in Figure 2, the possibility of performing the final sum through the same integrated circuits which make up the filters 2, is exploited. The filters 2 have, in fact, a second set of inputs, denoted by B0,...,B15, which usually receive a carry digital signal in those applications where the filters 2 are connected in series to one another. Now, the outputs of the second filter, that is the one operative to filter the least significant component of the signal S, are connected directly to the carry inputs B0,...,B15 of the preceding filter. Likewise, the outputs of a filter occupying the m-th position may be connected to such second inputs of the filter in the m-1 position. In this way, the final summing operation of the codings being output from the various filters can be performed internally of each circuit 2. This alternative embodiment of the invention is made possible by the provision of a delay block T between the output of the decoder D and the first eight inputs of the first filter. The delay introduced by the block T is equivalent to the latency required by the filter 2 to process the signal. The architecture and method described in the foregoing are most advantageous where the input signal S is encoded with nine bits. In this case, in fact, no more than two PFP filters in parallel are needed to achieve highly accurate filtering within a very short time. Obviously, any increase in the dynamic range of the input signal enhances the circuit complexity; on the other hand, it should not be overlooked that the integrated circuits making up the filters 2 are available at a fairly low price. Accordingly, the method of this invention has a major advantage in that it enables plural PFP filters, of well-proven reliability, to be used in lieu of alternative approaches which are much more costly either at design or at production level.

Claims: A method of filtering digital signals having a high dynamic range, including the steps of: splitting the input sampled signal (S) into at least m separate signal portions (S1, S2,...), m being an integer equal or larger than two; addressing each of said signal portions (S1,S2,...,Sm) respectively to one of m programmable filters (2) and performing each filtering operation in parallel and independently; and, reconstituting an output signal by having the digital outputs from each filter (2) summed together, characterized in that said splitting is effected by: selecting a first signal porttion consisting of the eight most significant bits of the input signal, storing this first portion and filtering this first portion by a first programmable filter (2), obtaining the other signal portion or portions from a parallel sum of said first signal portion and each remaining bit of the input signal, and filtering this other signal portion or these other signal portions by the remaining m-1 programmable filters (2) respectively, whereby a sequence of the m-1 remaining programmable filters is determined by the sequence of the remaining bits; and in that the reconstitution of the output signal is effected by connecting the outputs of the first and the remaining programmable filters (2) to a final summing block (5) or by connecting the outputs of the or each remaining programmable filter to the carry inputs (B0,...,B15) of the preceding programmable filter respectively. A method according to Claim 1, characterized in that the step of splitting the input sampled signal (S) includes the step of splitting the input sampled signal (S) into a first (S1) and a second signal portions (S2), the first signal portion (S1) being filtered by the first programmable filter (2), and the second signal portion (S2) of eight bits being directly applied to corresponding inputs of a second programmable filter. A method according to Claim 1, characterized in that the integer m is equal to 2 n , where n is the difference between the number of bits of the input signal and the number of input bits of the programmable filter or filters. A method according to Claim 1, characterized in that the step of splitting the input sampled signal (S) includes the step of splitting the input sampled signal (S) into signal portions (S1, S2, ...) of eight bits each. A high-resolution, digital filter architecture comprising: an input terminal receiving as input a sampled digital signal; means (D) coupled to the input terminal for splitting the input digital signal into m separate signal portions, m being an integer equal or larger than two; a first programmable filter (2), receiving a first signal portion, consisting of the eight most significant bits of the sampled digital signal, through a storage structure, and m-1 remaining programmable filters (2) coupled to said means for splitting for receiving the remaining signal portion or portions of the sampled digital signal; one of m-1 summing blocks (4) being arranged ahead of each remaining filter (2) respectively and receiving as a first input said first signal portion (S1) and as a second input a single bit of the remaining signal portion or portions respectively, whereby a sequence of the m-1 remaining programmable filters is determined by the sequence of the remaining bits; the digital outputs (O0,...,015) of the first programmable filter and of the remaining filter or filters (2) being connected to a final summing block (5) or the digital output of the or each remaining programmable filter being connected to corresponding carry inputs (B0,...,B15) of the preceding programmable filter respectively. An architecture according to Claim 5, characterized in that the remaining filter or filters (2) and corresponding summing blocks (4) are provided in a number m equal to 2 n , where n is the number of bits in excess of an eight-bit coding for the input sampled digital signal. An architecture according to Claim 5, characterized in that said means for splitting comprises a decoder (D) for the input sampled digital signal adapted to separate the first signal portion (S1) consisting of the first eight most significant bits from the remaining signal portion or portions. An architecture according to Claim 5, characterized in that the means for splitting (D) splits the input sampled signal (S) into a first (S1) and a second signal portions (S2), the first signal portion (S1) being filtered by the first programmable filter (2), and the second signal portion (S2) of eight bits being directly applied to corresponding inputs of a second programmable filter, which is in turn directly connected to said means for splitting (D). An architecture according to Claim 5, characterized in that the means for splitting (D) splits the input sampled signal (S) into signal portions (S2) of eight bits each.

Inventors: CREMONESI ALESSANDRO, GOLLA CARLA, CREMONESI, ALESSANDRO, GOLLA, CARLA
Applicants: ST MICROELECTRONICS SRL, STMICROELECTRONICS S.R.L.
Application year: 1993
Main classification: H03H17
IPC codes: G06F17, H03H17, H03H21