DIGITAL LOW PASS FILTER

Abstract: A digital low pass filter of the FIR type stores the input signal into register means with a plurality of taps. These taps are connected to address a look up table and the filtered output signal is derived from the output of the look up table. The filter may e.g. be used in systems where a one-bit direct stream digital (DSD) signal has to be requantized.

Description: Digital low pass filter The present invention relates to a digital low pass filter comprising register means with a plurality of taps to which register means a digital input signal is applied and from whose taps a series to parallel converted signal is obtained and means for deriving, from the series to parallel converted signal, a low pass filtered output signal. Such kind of filter, wherein the output signal of each tap is multiplied with a coefficient in a coefficient- multiplier and wherein the output signals of all coefficient-multipliers are summed to constitute the low pass filter output signal, is well known in the art e.g. from the book "Digitale Signaalbewerking" of A.W.M. van den Enden and N.A.M. Nerhoekx, and belongs to the category of filters usually referred to as a finite impulse response filters. For low pass filtering digital signals, often infinite impulse response (IIR) filters are used. However, in some cases finite impulse response (FIR) filters are preferred, because the group delay of FIR filters, especially symmetrical FIR filters, is more constant and because the word length of the signals in a FIR filter is usually smaller than in IIR filters. However, the above described FIR filter, despite its advantages, is not very suitable to filter digital signals with high sampling rates, because the multiplication operations are relatively time consuming. Substantially increasing the processing speed of such filter would require higher power consumption and larger chip area, which is often not acceptable. Therefore, it is the primary object of the invention to provide a digital low pass filter of the above described kind, which is faster in operation and which is therefore able to handle digital signals with a relatively high sampling rate and the digital low pass filter of the present invention is therefore characterized by a look up table, which is addressed by the series to parallel converted signal and by means to derive the low pass filter output signal from the so addressed look up table. The basic idea of this invention is that the combination of bits provided by the taps of the register means and belonging to a plurality of samples of the input signal, forms a digital word. This digital word can than advantageously be used to address a look up table. The low pass filter of the present invention is especially suitable for those input signals having a small word length, because usually the smaller the word length the higher the sampling rate is. More particularly, the invention may be used advantageously in those applications where single-bit bitstream signals have to be filtered where the digital input signal has a word-length of 1. An important application for the low pass filter of the present invention is in Super Audio Compact Disc (SACD®) systems, where single-bit signals have often to be requantized e.g. for the purpose of mixing two signals, or for compressing/decompressing such signals or for error concealment in such signals. In those applications the single bit signals have to be low pass filtered prior to the requantization, which can e.g. be performed in a sigma-delta-modulator. Examples of those applications are described in applicants prior filed European patent applications (ID 602583 and ID 602604). It will be apparent that the number of taps of the register means determines the word length of the signal with which the look up table is addressed and consequently the number of the different values which can be obtained from the look up table. For instance, when the register means has 16 taps, the look up table is addressed by 16-bit words and can then provide 2 16 = 65536 different values of output signal. The higher this value is, the higher the resolution of the filtered output signal can be but each more tap would double the magnitude of the look up table. However, it may be considered as a disadvantage that a large memory has to be used when a sufficiently high resolution has to be achieved. This disadvantage may be substantially minimized when, according to a further aspect of the invention, the digital low pass filter of the invention is further characterized in that the register means are divided into a plurality of registers, that each of the registers is arranged to address one of a plurality of look up tables and that the results of the addressing of the look up tables are added to derive the low pass filtered output signal. Therefore, in stead of a single look up table of e.g. 65536 locations, which is addressed by 16 register taps, the filter may comprise e.g. two look up tables of 256 locations each, which are each addressed by 8 taps of the shift register means. Then, at the cost of one more addition operation, the total numbers of locations is reduced to 512. Because the two look up tables are addressed independently from each other, the low pass filter may still provide up to 65536 different results. A further increase of the ability to handle input signals with high sampling rates may be obtained when the digital low pass filter of the invention is further characterized in that one of the plurality of registers is loaded by the digital input signal and the other registers are parallel loaded from said one register at a rate which is equal to the sampling rate of the digital input signal divided by the number of samples stored in each of said plurality of registers. In this way a kind of down-sampling is obtained for the addressing- and addition-operations. When the output signal of the filter has to be available at the original sampling rate of the input signal, an up-sampling at the output has to take place with the result that the up-sampled signal is kept constant during a plurality of samples and then jumps to the new value. This operation resembles a sample and hold operation which results in an output signal having a staircase character. If such a staircase output signal is undesired, because it contributes to the noise level of the signal, an improvement may be achieved by interpolation. To this end the digital low pass filter of the invention may still further be characterized by means to recover from the addressing of the plurality of look up tables two successive values of the low pass filtered output signal and means to derive from said successive values of the low pass filtered output signal intermediate values at a rate which is equal to the sampling rate of the digital input signal. The number of samples to be stored in each of the plurality of registers and consequently the word length of the addresses of the look up tables may be any suitable number, however preferably said number is equal to eight, because this choice allows the use of standard, fast and easily available byte-oriented circuitry. Another choice is the word length of the digital signals stored in the look up table and delivered to the output of the digital filter. The larger the output word length is, the more accurate the value of the output signal can be and the less noise is introduced by the filter. On the other hand the word length should be adapted to the arrangement which receives the filtered signal. E.g. when the filter is followed by a sigma-delta-requantizer, the output word length should be equal to the word length to which the filters in the requantizer are designed. Therefore, the invention also relates to a system for processing single bit digital signals comprising a requantizer which is preceded by a digital low pass filter according to the present invention. The invention will be further explained with reference to the attached figures. Herein shows: Figure 1 a prior art digital low pass filter and Figure 2, 3, 4, and 5 different embodiments of a digital low pass filter in accordance with the invention in a system for requantizing single bit signals.. The prior art low pass filter of figure 1 comprises a shift register R to which a digital input signal is applied through an input terminal I. The shift register R consists of N-l delay elements D 1 ....DN -1 , each of which delays the input signal by one period T s of the sample rate. The output of the shift register R consists of N taps PI....PN, so that the signal at each tap is delayed by one period T s with respect to the signal at the previous tap. Each of the tap-signals is applied to a coefficient multiplier CJ...CN and the output signals of these multipliers are added in an adder A to constitute the filtered output signal at an output terminal O. In practice this low pass filter may e.g. be used to filter audio signals having a sample rate of 44.1 kHz and a word length of 14 bits. The multipliers C^-CN and the adder A increase the word length so that the output signal at the output terminal O may have a word length of e.g. 20 bits. The coefficients of the multipliers CI...CN determine the characteristics of the filter, such as cut off frequency, ripple and group delay. Preferably the filter is symmetrical (Ci = C-N, C 2 = C N-I, etc.) because in that case the group delay of the filter is constant. The filter of figure 1 is not suitable for filtering digital input signals having a substantially higher sampling rate, such as the DSD (Direct Stream Digital) format which is used for the Super Audio CD (SACD®) format. This is a one bit signal format with a sampling rate of 64*44.1 kHz = 2.8224 Mhz. With present technics the multiplication/addition operation of the filter of figure 1 cannot economically be done fast enough. In the digital low pass filter of figure 2 the multipliers CI...O N and the adder A of figure 1 are replaced by a look up table L which is addressed by the N-bit digital word from the taps Ti-.TN of the register R. The output of the look up table L constitutes the output of the filter. Often, DSD format signals have to be requantized and to this end the filter output signal may be applied to a sigma-delta modulator SD. Because the addressing of the look up table in the arrangement of figure 2 requires less time than the multiplication and addition operation in the arrangement of figure 1, the filter of figure 2 can handle substantially higher sampling rates. The values, stored in the look up table, can be chosen such that the filter characteristics are identical to those of figure 1. With other words, the design of the filter of figure 2 can be done by firstly designing the coefficients ^...C N of the filter of figure 1 in the usual way, and then calculating therefrom the contents to be stored in the look up table. As is already observed in the preamble, the arrangement of figure 2 requires a relatively large look up table. With a practical value of N = 16 the look up table contains 2 16 = 65536 locations. A substantial reduction of the size of the look up table is obtained with the arrangement of figure 3 wherein, as in figures 4 and 5, by way of preferred example, the total number of taps N of the register means is chosen equal to 16. In the arrangement of figure 3 the look up table L of figure 2 is split up into two smaller look up tables Li and L 2 . The look up table L\ is addressed by the first half Ti ^ .Ts of the register taps and the look up table L 2 is addressed by the second half T 9 ...T 16 of the register taps. The output signals of the two look up tables are added in an adder Ai. Because each of the look up tables Lj and L 2 is addressed by only 8 taps, these look up tables each have only 2 8 = 256 locations, so that a substantial reduction of a factor 128 in locations is obtained. Nevertheless, because the look up tables are addressed independently from each other, the output signal delivered by the adder A \ may still have 65536 different values. A low pass filter with a further improved handling of high speed sampling rates is represented in figure 4. In this figure corresponding elements with those of figure 3 have been given corresponding references. The register D ϊ ...Di 5 of the arrangement of figure 3 has been split up into two registers Rι and R 2 in which register Ri comprises the delay elements D 1 ...D 7 and register R 2 the delay elements D 9 ...D 15 . Delay element D 8 is left out and instead a parallel 8-bit wide bus B connects register R 2 to register Ri. In operation, suppose that register Rt contains a byte bi. Before a next byte b 2 of 8 bits of the input signal is shifted into register R ls the byte bj is parallel loaded through the bus B into register R 2 . When the shift of the byte b 2 into register Ri is finished, the two look up tables are addressed; look up table L 2 is addressed by byte bi, look up table \ is addressed by byte b 2 and the results of the two address operations are added in adder Ai. Eight sample periods later, the register R 2 contains byte b 2 and register Rt contains byte b 3 and a new addressing of the two look up tables and a new addition by adder Al takes place. Consequently, when Ei(b 2 ) denotes the output of look up table L 1? when addressed by byte b 2 , and E 2 (bj) denotes the output of look up table L 2 , when addressed by byte bi, then the adder Ai delivers the signal E (b ! ) + Ei(b 2 ). Eight sample periods later the adder Ai delivers the signal E 2 (b 2 )+E 1 (b 3 ) and so on. When the sampling rate of the input signal is denoted by f s , the rate at which the bytes are transferred, the look up tables are addressed and the adder Ai operates is fs/8, so that a down-sampling with a factor 8 has taken place. Because the requantizer SD needs an input signal at the original sampling rate f s , up-sampling has to take place at the output of the adder Ai, which means that eight times the same signal level is outputted to the requantizer SD before the signal is refreshed, thereby giving the signal a staircase character. Therefore, the arrangement of figure 4 performs a down-sampled low pass filtering and up-sampling and hold at the output. If the sample and hold action at the output of the lowpass filter of figure 4 gives a too noisy signal, an improvement may be achieved by an interpolation operation at the output of the filter. An example hereof has been depicted in figure 5 in which corresponding elements with those of figure 4 have been given corresponding references. The low pass filter of figure 5 comprises an additional register R 3 coupled to register Rt and an additional register R 4 coupled to register R 2 . Furthermore two additional look up tables Lj a and L 2a are provided which have the same content as the look up tables Li and L 2 respectively. While the outputs of the look up tables Li and L 2 are added in an adder A l3 the outputs of the look up tables L la and L 2a are added in an adder A la . In operation, when a byte bi has been shifted into register Ri this byte is subsequently transferred to register R and to register R 3 . Eight sample periods later, when a byte b 2 has been loaded into register R ls the byte bi in R 2 is transferred to i and the byte b 2 in Ri is transferred to R 2 and R 3 . Again eight sample periods later, when a byte b 3 has been loaded into Ri - and before the bytes are transferred to other registers - the respective look up tables are addressed: Li is addressed by byte b 3 from Ri, L 2 is addressed by byte b 2 from R 2 , L la is addressed by byte b 2 from R 3 and L 2a is addressed by byte bi from Rj. Subsequently byte b 2 is transferred from R 2 to i, byte b 3 is transferred from Ri to R 2 and R 3 , byte b 4 is shifted into Ri and so on. The result of the addressing of the look up tables is that look up table Li delivers the signal Eι(b 3 ) and look up table L 2 delivers the signal E (b ). These two signals are added in adder Ai to obtain the signal E 2 (b 2 ) + Ei(b 3 ), exactly as was the case in the arrangement of figure 4. Additionally look up table L la delivers the signal Ei(b 2 ) and the look up table L a the signal E 2 (bi) so that the adder A la delivers the signal E 2 (bi) + Ei(b 2 ), which is the same as the signal obtained in the arrangement of figure 4 eight sampling periods (one sub-sampling period) earlier. The two signals are combined in an interpolator IP, where a linear interpolation between the two signals is performed at the original sampling rate f s . This [(8-i){E2(bi) + Ei(b 2 )}+i{E2(b 2 ) + Ei(b 3 )}] o interpolation may be expressed by the following formula: wherein i is an integer increasing from 0 to 8. Alternative interpolation schemes may be used. E.g. the output of the adder Ai in figure 4 may be stored in a register, so that this output is available during the next sub- sampling period. Then this output of the register and the refreshed output of the adder Ai may be combined in a linear or nonlinear interpolator.

Claims: CLAIMS: 1. A digital low pass filter comprising register means (R) with a plurality of taps (T) to which register means a digital input signal is applied and from whose taps a series to parallel converted signal is obtained and means for deriving, from the series to parallel converted signal, a low pass filtered output signal, characterized by a look up table (L), which is addressed by the series to parallel converted signal and by means to derive the low pass filter output signal from the so addressed look up table. 2. * A digital low pass filter as claimed in claim 1 characterized in that the digital input signal has a word-length of 1. 3. A digital low pass filter as claimed in claim 1 , characterized in that the register means are divided into a plurality of registers (Ri, R 2 ), that each of the registers is arranged to address one of a plurality of look up tables (Li, L 2 ) and that the results of the addressing of the look up tables are added (Ai) to derive the low pass filtered output signal. 4. A digital low pass filter as claimed in claim 3, characterized in that one of the plurality of registers (Ri) is loaded by the digital input signal and the other registers (R 2 ) are parallel loaded from said one register (Ri) at a rate which is equal to the sampling rate of the digital input signal divided by the number of samples stored in each of said plurality of registers (Ri, R 2 ). 5. A digital low pass filter as claimed in claim 4 characterized by means to recover from the addressing of the plurality of look up tables (L ls L 2 , L la , L 2a ) two successive values of the low pass filtered output signal and means (IP) to derive from said successive values of the low pass filtered output signal intermediate values at a rate which is equal to the sampling rate of the digital input signal. 6. A digital low pass filter as claimed in claim 3, characterized in that said number of samples stored in each of said plurality of registers (R ls R 2 ) is eight. 7. A system for processing single bit digital signals comprising a requantizer which is preceded by a digital low pass filter as claimed in any of the preceding claims.

Inventors: DERK REEFMAN, NUIJTEN PETRUS A C M, DERK, REEFMAN, NUIJTEN, PETRUS, A., C., M.
Applicants: KONINKL PHILIPS ELECTRONICS NV, KONINKLIJKE PHILIPS ELECTRONICS N.V.
Application year: 2001
Main classification: H03H17
IPC codes: H03H17