Signal processor for 1-bit signals, comprising a nth order Delta-Sigma modulator

Abstract: A signal processor for 1-bit signals comprises a fifth order Delta-Sigma Modulator (DSM) having an input 4 for receiving a 1-bit signal and an output 5 at which a processed 1-bit signal is produced by a quantizer Q. The quantizer Q receives a p-bit signal from a series of 5 signal integration stages. Each stage comprises a first 1-bit multiplier An coupled to the input 4, a second 1-bit multiplier Cn coupled to the output 5, an adder 6n which sums the outputs of the coefficient multipliers and an integrator 7n which integrates the output of the adder 6n. A final stage comprises a coefficient multiplier An+1 and an adder 6n+1. The adder 6n+1 sums the output of the coefficient multiplier An+1 and the output of the integrator of the preceding integration stage. The coefficients An and Cn are chosen to provide an overall attenuation of the input signal and of the quantization noise produced by the quantizer and also to provide a low pass filter which provides a compensating gain to the input signal. In this way the quantization noise outside the band of the input signal is reduced. Reduction of quantization noise enhances stability of the circuit and also allows several DSMs to be connected in series by preventing the build-up of excess noise which may compromise stability.

Description: The present invention relates to a 1-bit signal processor comprising an n th order Delta-Sigma Modulator having a filter section where n is at least three. Preferred embodiments of the invention relate to processing audio signals but the invention is not limited to audio signal processors. Background to the present invention will now be described by way of example with reference to Figures 1, 2 and 3 of the accompanying drawings of which Figure 1 is a block diagram of a known Delta-Sigma Modulator, Figure 2 is a block diagram of a previously proposed Delta-Sigma Modulator configured as an 3rd order (n=3) filter section and Figure 3 shows a noise shaping characteristic. It is known to convert an analogue signal to a digital form by sampling the analogue signal at at least the Nyquist rate and encoding the amplitudes of the samples by an m bit number. Thus if m = 8, the sample is said to be quantized to an accuracy of 8 bits. In general m can be any number of bits equal to or greater than 1. For the purpose of quantizing to only 1 bit, it is known to provide an analogue to digital converter (ADC) known either as a "Sigma-Delta ADC" or as a "Delta-Sigma ADC". Herein the term "Delta-Sigma" is used. Such an ADC is described in for example "A Simple Approach to Digital Signal Processing" by Craig Marven and Gillian Ewers ISBN 0-904.047-00-8 published 1993 by Texas Instruments. Referring to Figure 1 in an example of such an ADC, the difference 1 (Delta) between an analogue input signal and the integral 2 (Sigma) of the 1-bit output signal is fed to a 1-bit quantizer 3. The output signal comprises bits of logical value 0 and 1 but representing actual values of -1 and +1 respectively. The integrator 2 accumulates the 1-bit outputs so that value stored in it tends to follow the value of the analog signal. The quantizer 3 increases (+1) or reduces (-1) the accumulated value by 1-bit as each bit is produced. The ADC requires a very high sampling rate to allow the production of an output bit stream the accumulated value of which follows the analogue signal. The term "1-bit" signal as used in the following description and in the claims means a signal quantized to an accuracy of 1 digital bit such as is produced by a Delta-Sigma ADC. A high-order Delta-Sigma Modulator (DSM) is described in US Patent 5,181,032. The DSM has a single feedback loop including a linear network and quantiser. The quantiser comprises a plural-bit analog-to-digital converter, a digital compensator and a single-bit digital-to-analogue converter. The linear network comprises a cascade of integrators, a second-order resonator, a cascade of second order resonators or a cascade or second order resonators with an additional integrator. A DSM configured as n th order filter section for directly processing a 1-bit signal was proposed by N.M. Casey and James A.S. Angus in a paper presented at the 95th AES Convention 7-10 October 1993 New York, USA entitled "One Bit Digital Processing of Audio Signals" - Signal Processing: Audio Research Group, The Electronics Department, The University of York, Heslington, York YO1 5DD England. Figure 2 shows a 3rd order (n=3) version of such a DSM filter section. Referring to Figure 2, the DSM has an input 4 for a 1-bit audio signal and an output 5 at which a processed a 1-bit signal is produced. The bits of the 1-bit signal are clocked through the DSM by known clocking arrangements which are not shown. The output 1-bit signal is produced by a 1-bit quantizer Q which is for example a comparator having a threshold level of zero. The DSM has three stages each comprising a first 1-bit multiplier a 1 , a 2 , a 3 connected to the input 4, a second 1-bit multiplier c 1 , c 2 , c 3 connected to the output 5, an adder 6 1 , 6 2 , 6 3 and an integrator 7 1 , 7 2 , 7 3 . The 1-bit multipliers multiply the received 1-bit signal by p bit coefficients A 1 , A 2 , A 3 , C 1 C 2 , C 3 producing p bit products which are added by the adders 6 1 , 6 2 , 6 3 and the sums passed to the integrators 7. In the intermediate stages the adders 6 2 , 6 3 also sum the output of the integrator of the preceding stage. A final stage comprises another 1-bit multiplier A 4 connected to the input which multiplies the input signal by a p bit coefficient A 4 and an adder 6 4 which adds the product to the output of the integrator 7 3 of the preceding stage. The sum is passed to the quantizer 2. Within the DSM, two's complement arithmetic maybe used to represent the positive and negative p bit numbers. The input to the quantizer Q may be positive, quantized at the output as +1 (logical 1) or negative quantized at the output as -1 (logical 0). As observed by Casey and Angus "a one bit processor .. will produce a one bit output that contains an audio signal that is obscured by noise to an unacceptable level and it is imperative the quantization noise is suitably shaped". The noise which obscures the audio signal is the quantization noise produced by the quantizer Q. The quantizer Q may be modelled as an adder which has a first input receiving an audio signal and a second input receiving a random bit stream (the quantization noise) substantially uncorrelated with the audio signal. Modelled on that basis, the audio signal received at the input 4 is fed forward by multipliers a 1 , a 2 , a 3 , a 4 to the output 5 and fed back by multipliers c 1 , c 2 , c 3 from the output 5. Thus coefficients A1 to A4 in the feed forward path define zeros of the Z-transform transfer function of the audio signal and coefficients C1-C3 in the feed back path define poles of the transfer function of the audio signal. The noise signal, however is fed-back from the quantizer by the multipliers C 1 -C 3 so that coefficients C1-C3 define poles of the transfer function of the noise signal. The transfer function of the noise signal is not the same as that of the input signal. The coefficients A1 to A4 and C1 to C3 are chosen to provide circuit stability amongst other desired properties. The coefficients C1-C3 are chosen to provide noise shaping so as to minimise quantization noise in the audio band, as shown for example in Figure 3 by the full line 31. The coefficients A1-A4 and C1-C3 are also chosen for a desired audio signal processing characteristic. The coefficients A1-A4 and C1-C3 may be chosen by: a) finding the Z-transform H(z) of the desired filter characteristic - e.g noise shaping function: and b) transforming H(z) to coefficients. This may be done by the methods described in the papers "Theory and Practical Implementation of a Fifth Order Sigma-Delta A/D Converter, Journal of Audio Engineering Society, Volume 39, no. 7/8, 1991 July/August by R. W Adams et al." and in the paper by Angus and Casey mentioned hereinbefore and using the knowledge of those skilled in the art. One way of calculating the coefficients is outlined in the accompanying Annex A. It is proposed herein that a signal processor may comprise a plurality of DSMs coupled in series or cascaded, to process 1-bit signals. Such a proposal is not known from the papers mentioned above. It is desirable to provide signal filtering between DSMs and also to maintain a 1-bit signal stream between DSMs. Such filtering may be needed for example to prevent undesired build-up of quantisation noise in successive DSMs. However, suitable digital filters would at least sum bits of the stream resulting in multi-bit numbers and/or multiply the bit stream by one or more p bit coefficients. All the coefficient multipliers of a DSM downstream of such a filter would then receive a p bit signal and would need to be p bit multipliers which are uneconomic. According to one aspect of the present invention, there is provided a signal processor for 1-bit signals, representing information, comprising a plurality of nth order Delta-Sigma Modulators (DSMS) arranged in series, where n is at least 3, each DSM having an input for receiving a 1-bit input signal, a quantizer for requantising a p-bit signal where p is greater than one, to 1-bit form, the requantized signal being the output signal of the DSM, a plurality of signal combiners including a first combiner for forming an integral of an additive combination of the product of the input signal and a coefficient and of the product of the output signal and a coefficient, at least two intermediate combiners each for forming an integral of an additive combination of the product of the input signal and a coefficient and of the product of the output signal and a coefficient, and of the integral of the preceding stage, and a final combiner for forming an additive combination of the product of the input signal and a coefficient and of the integral of the preceding stage to form the said p bit signal which is requantized by the quantizer, to form the 1-bit output signal, in which said coefficients are chosen to provide noise shaping with a predetermined gain in the frequency band of the information and a gain less than the predetermined gain outside the frequency band of the information. According to an embodiment of the invention the said coefficients are chosen to provide an overall attenuation of the input signal and the quantization noise and to also provide a low pass filter which provides a compensating gain to the input signal. By providing the overall attenuation and the compensating gain to the input signal, at least some noise outside the frequency band of the input signal (out of band noise) is reduced. It is believed that build up of noise whether within the band of the input signal (in-band noise) or out of band noise can at least reduce the stability of a DSM. Some noise which exists in-band and adjacent the in-band frequency zone will be subject to the compensating gain because the low pass filter characteristic will have a "roll-off". However, overall noise is reduced. By providing the overall attenuation and compensating gain in a DSM, 1-bit signals may be received, processed and fed to a subsequent DSM in 1-bit form avoiding the need for p bit coefficient multipliers in the signal combiners of the subsequent DSM. For a better understanding of the present invention, reference will now be made by way of example to Figures 4 to 7 of the accompanying drawings of which: Figure 4 is a schematic block diagram of a Delta-Sigma Modulator according to the present invention; Figure 5 comprises frequency amplitude diagrams a, b and c illustrating the effect of the Delta-Sigma Modulator of Figure 4 on an input signal; Figure 6 is a block diagram of an integrator of the DSM of Figure 4; Figure 7 is a schematic block diagram of modification of the DSM of Figure 4; Figure 8 is a schematic block diagram of a plurality of the DSMs of Figures 4, 5 and 6 in series; and Figure 9 is a frequency response diagram. The Delta-Sigma Modulator (DSM) of Figure 4 is a fifth order DSM having five integrator sections and a final section. The DSM has an input 4 for receiving a 1-bit signal and an output 5 at which a processed 1-bit signal is produced. The bits are clocked through the DSM by know clocking arrangements (not shown). The output signal is produced by a quantizer Q in the final stage. Quantizer Q receives a p bit signal which may be, for example, in twos complement form to represent positive and negative numbers. The quantizer may be a comparator having a threshold of zero. The quantizer quantizes positive signals as + I (logical 1) and quantizes negative signals as -1 (logic 0). The first integrator section comprises a first 1-bit multiplier a1 connected to the input 4, a second 1-bit coefficient multiplier c1 connected to the output 5, an adder 61 which sums the outputs of the 1-bit multipliers a1 and c1 and an integrator 71 which integrates the output of the adder 61. The 1-bit coefficient multipliers multiply the 1-bit signals by p-bit coefficients A1 and C1. Each of the four intermediate integrator sections likewise comprises a first 1-bit coefficient multiplier a2, a3, a4, a5, connected to the input 4, a second 1-bit coefficient multiplier c2, c3, c4, c5 connected to the output 5, an adder 62, 63, 64, 65 and an integrator 72, 73, 74, 75. The adders 62, 63, 64, 65 receive in addition to the outputs of the coefficients multipliers the output of the integrator of the preceding stage. An example of an integrator 71, 72, 73, 74, 75 is shown in Figure 6. The integrator comprises an adder 610 in series with a delay element 611. The output of the delay element is fed back to the adder to accumulate the integral of the output of the adder which sums the outputs of the coefficient multipliers. The adder 610 of Figure 6 may be implemented by the adder 61-65 which sums the outputs of the coefficient multipliers of the stage. Thus it is not essential to have separate adders for the coefficient multipliers and for the integrator. For the situation shown in Figure 4 where the coefficients a1 to a5 and c1 to c5 are fixed and a separate adder is provided in the integrator, the coefficient multipliers a and c and the adder 6 which sums the outputs of the coefficient multipliers maybe replaced by a look-up table. For a 1-bit signal multiplied by a coefficient a1 and by a coefficient c1 the outputs are +a1, -a1, +c1, -c1. A look-up table can conveniently store all the possible combinations of +a1 and -a1 with + c1 and - c1; the store would be addressed by the 1-bit signals. As discussed above the coefficients a1 to a6 and c1 to c5 may be chosen by the methods described in the above-mentioned papers. The coefficients are, of course, chosen to provide circuit stability. The coefficients chosen to provide noise shaping as schematically indicated by the filter characteristic in Figure 5a whereby noise in the region of the input signal 52 (in-band noise) is reduced. Figure 5a illustrates the principle of noise-shaping as background to the present invention. In accordance with an embodiment of the present invention as shown in Figures 4 and 5b the coefficients chosen to provide noise shaping and also provide an overall attenuation by e.g. 6dB of the input signal and the noise as shown in Figure 5b and to provide in addition a low pass filter characteristic which as indicated in Figure 5c, compensates by e.g. 6dB the attenuation of the input signal (and also any in-band noise) whilst retaining the attenuation of the out of band noise. As is evident from Figure 5c the low pass filter characteristic has a roll-off whereby the compensating gain is applied to at least some of the noise. Figures 5A to 5C are based on a design principle in which a 2nd order low pass filter is combined with a 3rd order DSM. In the resulting 5th order DSM, the resulting coefficients are distributed throughout the DSM. Stability of a DSM is compromised by excessive noise whether in-band or out of band. The reduction of noise allows a plurality of DSMs to be cascaded in series as shown in Figure 8 for example where 3 DSMs 90, 91, 92 are provided in series. Referring to Figure 7 and as more fully described in UK patent EP 0 845 868 a DSM may be constructed as a signal mixer by providing 2 inputs 4a and 4b the input 4a being connected to the adder 61 to 66 by the coefficient multipliers a1 to a6 and the input 4b being connected to the same adders by further coefficient multipliers b. Figure 7 shows the arrangement for just one integrator stage. Referring to Figure 8 and as further explained in the UK patent EP 0 845 868 the DSMs 90, 91 and 92 may be signal mixers and adders. Figure 9 shows an alternative filter characteristic in which a predetermined gain e.g. unity gain is applied to the audio band and lesser gain e.g. less than unity gain is applied outside the audio band. The coefficients of the 5th order DSM of Figure 4 are chosen to implement the characteristic of Figure 9. CALCULATING NOISE SHAPING COEFFICIENTS This annex outlines a procedure for analysing a fifth order DSM and for calculating coefficients of a desired filter characteristic. A fifth order DSM is shown in Figure A having coefficients a to f and A to E, adders 6 and integrators 7. Integrators 7 each provide a unit delay. The outputs of the integrators are denoted from left to right s to w. The input to the DSM is a signal x[n] where [n] denotes a sample in a clocked sequence of samples. The input to the quantizer Q is denoted y[n] which is also the output signal of the DSM. The analysis is based on a model of operation which assumes quantizer Q is simply an adder which adds random noise to the processed signal. The quantizer is therefore ignored in this analysis. The signal y[n] = fx[n] + w[n] i.e. output signal y[n] at sample [n] is the input signal x[n] multiplied by coefficient f plus the output w[n] of the preceding integrator 7. Applying the same principles to each output signal of the integrators 7 results in Equations set 1. y [ n ] = fx [ n ]+ w [ n ] w [ n ] = w [ n -1]+ ex [ n -1]+ Ey [ n -1]+ v [ n -1] v [ n ] = v [ n -1]+ dx [ n -1]+ Dy [ n -1]+ u [ n -1] u [ n ] = u [ n -1]+ cx [ n -1]+ Cy [ n -1]+ t [ n -1] t [ n ] = t [ n -1]+ bx [ n -1]+ By [ n -1]+ s [ n -1] s [ n ] = s [ n -1]+ ax [ n -1]+ Ay [ n -1] These equations are transformed into z-transform equations as well known in the art resulting in equations set 2. Y ( z ) = fX ( z )+ W ( z ) W ( z )(1- z -1 ) = z -1 ( eX ( z )+ EY ( z )+ V ( z )) V ( z )(1- z -1 ) = z -1 ( dX ( z )+ DY ( z )+ U ( z )) U ( z )(1- z -1 ) = z -1 ( cX ( z )+ CY ( z )+ T ( z )) T ( z )(1- z -1 ) = z -1 ( bX ( z )+ BY ( z )+ S ( z )) S ( z )(1- z -1 ) = z -1 ( aX ( z )+ AY ( z )) The z transform equations can be solved to derive Y(z) as a single function of X(z) (Equation 3) Y ( z ) = fX ( z ) + z -1 (1- z 1 ) ( eX ( z ) + EY ( z ) + z -1 1- z -1 ( dX ( z ) + DY (z) + z -1 1- z -1 ( cX ( z ) + CY ( z ) + z -1 1- z -1 ( bX ( z ) + BY ( z ) + z -1 1- z -1 ( aX ( z ) + AY ( z )))))) This may be reexpressed as shown in the right hand side of the following equation, Equation 4. A desired transfer function of the DSM can be expressed in series form Y ( z ) X ( z ) given in left hand side of the following equation and equated with the right hand side in Equation 4. Y ( z ) X ( z ) = α 0 + α 1 z -1 + α 2 z -2 + α 3 z -3 + α 4 z -4 + α 5 z -5 β 0 + β 1 z -1 + β 2 z -2 + β 3 z -3 + β 4 z -4 + β 5 z -5 = f (1- z -1 ) 5 + z -1 e (1- z -1 ) 4 + z -2 d (1- z -1 ) 3 + z -3 c (1- z -1 ) 2 + z -4 b (1- z -1 ) + z -5 a (1- z -1 ) 5 - z -1 E (1- z -1 ) 4 - z -2 D (1- x -1 ) 3 - z -3 C (1- z -1 ) 2 - z -4 B (1- z -1 )- Z -5 A Equation 4 can be solved to derive the coefficients f to a from the coefficients α 0 to α 5 and coefficients E to A from the coefficients β 0 to β 5 as follows noting that the coefficients α n and β n are chosen in known manner to provide a desired transfer function. f is the only z 0 term in the numerator. Therefore f = α 0 . The term α 0 (1-z -1 ) 5 is then subtracted from the left hand numerator resulting in α 0 + α 1 z -1 ...+...α 5 z -5 - α 0 (1-z -1 ) 5 which is recalculated. Similarly f(1-z -1 ) 5 is subtracted from the right hand numerator. Then e is the only z -1 term and can be equated with the corresponding α 1 in the recalculated left hand numerator. The process is repeated for all the terms in the numerator. The process is repeated for all the terms in the denominator.

Claims: A signal processor for 1-bit signals representing information, comprising a plurality of nth order Delta-Sigma Modulators (DSMs) (90, 91, 92) arranged in series, where n is at least 3, each DSM having an input (4) for receiving a 1-bit input signal, a quantizer (Q) for requantising a p-bit signal, where p is greater than one, to 1-bit form, the requantized signal being the output signal of the DSM, a plurality of signal combiners including a first combiner (a1, c1, 61, 71) for forming an integral of an additive combination of the product of the input signal and a coefficient (A 1) and of the product of the output signal and a coefficient (C1), at least two intermediate combiners (a2-a5, c2-c5, 62-65, 72-75) each for forming an integral of an additive combination of the product of the input signal and a coefficient (A2-A5) and of the product of the output signal and a coefficient (C2-C5), and of the integral of the preceding stage, and a final combiner (a6, 66) for forming an additive combination of the product of the input signal and a coefficient (A6) and of the integral of the preceding stage to form the said p bit signal which is requantized by the quantizer (Q), to form the 1-bit output signal, in which said coefficients (A1-A6, C1-C5) are chosen to provide noise shaping with a predetermined gain in the frequency band of the information and a gain less than the predetermined gain outside the frequency band of the information. A processor according to claim 1, wherein the predetermined gain is unity. A processor according to claim 1, wherein the said coefficients (A1-A6, C1-C5) are chosen to provide an overall attenuation of the input signal and the quantization noise within and outside the frequency band of the information and to also provide a low pass filter which provides a compensating gain within the frequency band of the information. A processor according to claim 3, wherein the said overall attenuation is 6dB and the compensating gain is 6dB. A processor according to claim 1, 2, 3 or 4, wherein n=5, there being four intermediate combiners. A processor according to claim 1, 2, 3, 4 or 5 wherein the DSM has a further input for (4B) receiving a further 1-bit signal and each combiner additionally combines the product of the further 1-bit signal with a further coefficient (BN) with the aforesaid additive combination. An audio signal processor comprising a signal processor according to any preceding claim.

Inventors: ANGUS JAMES ANDREW SCOTT, EASTTY PETER CHARLES, SLEIGHT CHRISTOPHER, THORPE PETER DAMIEN, ANGUS, JAMES ANDREW SCOTT, EASTTY, PETER CHARLES, SLEIGHT, CHRISTOPHER, THORPE, PETER DAMIEN
Applicants: SONY UK LTD, SONY UNITED KINGDOM LIMITED
Application year: 1997
Main classification: H03M7
IPC codes: H03M7