SIGNAL PROCESSING

Abstract: A filter (26) is used to average power values of a signal that has been subjected to varying gain. The gain is factorised into two factors, the first being the highest power of 2 within the gain, and the second being a factor which can be multiplied with the first factor to reproduce the gain. The power values are divided by the second factor before reaching the filter (26) and by the first factor after passing through the filter (26). The signal values already in the filter (26) are bit-shifted when the exponent in the first changes so that the filter is not disrupted by changes in the gain.

Description: SIGNAL PROCESSING The invention relates to signal processing, and in particular to methods and apparatus for filtering digital signals that have been produced under variable gain conditions. It is known to amplify a received signal under variable gain conditions. For example, it is known to use automatic gain control in a receiver for the purpose of amplifying a received signal which is fading. A problem is that subsequent signal processing performed on the amplified signal is complicated by the fact that the gain used in the amplification is variable. It is an aim of the present invention to provide simpler signal processing of such signals. According to one aspect, the invention provides signal processing apparatus for operating on a signal that has been amplified under variable gain, the apparatus comprising a digital filter for filtering the signal values, and adapting means for adapting at least one signal value when the gain changes so as to ameliorate the biasing of the filtering operation by the change in gain. According to anotlier, and related, aspect, the invention also provides a method of processing a signal that has been amplified under variable gain, comprising digitally filtering the signal values and adapting at least one signal value when the gain changes so as to ameliorate the biasing of the filtering operation by the change in gain. Hence, the invention provides a more reliable way of applying a desired filtering characteristic to a digital signal. In a preferred embodiment, the signal values involved in the filtering process are adapted by bit-shifting. Advantageously, this is simpler than generating adapted signal values by arithmetic. In one embodiment, the variable gain is factored into two factors, the first factor being the highest exponent of two present in the gain and the second factor being such that when multiplied with the first factor, the result is the gain value. Then, the signal values can be divided by the second factor such that the only gain variations present in the signal values are variations in the exponent of the first factor, e.g. variations such as the first factor changing from 2 m to 2 m+1 . Advantageously, the filter need only then be able to cope with variations of this constrained kind. By then bit-shifting the signal values involved in the filtering process, it is possible to render the filtering operation insensitive to gain variations by a factor which is an exponent of two. Preferably, when bit-shifted, a signal value is displaced by a number of binary places corresponding to the numerical change in the exponent of the first factor. In a preferred embodiment, the filter averages the signal values passing through it. Preferably, this averaging process is a weighted averaging process. In a preferred embodiment, the filtered signal is a signal representing one of a noise signal and a wanted signal that is to be used for assessing the wanted signal relative to the noise signal (e.g. by calculating the ratio of the power in the wanted signal relative to the power in the noise signal). By way of example only, an embodiment of the invention will now be described with reference to the accompanying figures, in which: Figure 1 is a block diagram of a radio receiver; and Figure 2 is a block diagram of the amplitude adaptive filter of Figure 1. The radio receiver of Figure 1 comprises an antenna 11 for receiving transmitted wireless signals. Signals picked up by antenna 11 are supplied to radio frequency circuitry 12 which recovers a signal from amongst the signals received at antenna 11. The recovered signal is passed to A-D converter 13 for conversion to the digital domain. One of the roles of RF circuitry 12 is to amplify the recovered signal. This amplification is done using automatic gain control and accordingly the recovered signal sent to A-D converter 13 has been produced under a varying gain. The gain which the RF circuitry 12 can apply to the recovered signal can adopt one of a plurality of discrete values as directed by the automatic gain control process, which process operates in a known manner. Downstream of A-D converter 13, the digital version of the recovered signal is supplied to the main processing unit 18 of the radio receiver, where the information contained in the recovered signal is used in its intended way. The radio receiver is also arranged to calculate the signal to noise power ratio of the recovered signal. For this purpose, the digital version of the recovered signal output by A-D converter 13 is supplied to each of a signal power measuring section 14 and to a noise power measuring section 15. The signal power measurement section 14 calculates the power present in a wanted component of the digital version of the recovered signal. The noise power measurement section 15 calculates the power present in an unwanted component (i.e. a noise component) of the digital version of the recovered signal. The wanted component is isolated by a known correlation process and is subtracted from the recovered signal to produce the noise component. The signal indicative of the measured noise power (from measurement section 15) is supplied to amplitude adaptive filter 16 which averages the noise power signal. The filter 16 has a number of coefficients that serve as the weighting factors used in the averaging process. The amplitude adaptive filter 16 is supplied with the power gain values used in RF circuitry 12 to amplify the recovered signal, and the amplitude adaptive filter 16 uses these gain values (in a manner which will be described in more detail later) to ensure that the averaging process that it performs is not biased by any changes in the gain used to amplify the recovered signal. The output of filter 16 is a signal indicative of the average power of the noise component. The output of measuring section 14 is a signal indicative of the power in the wanted component of the recovered signal. The signals from section 15 and filter 16 are supplied to subtractor 17 which subtracts the noise power signal from the wanted power signal. This subtraction produces a signal to noise power ratio, since at this stage the signal power signal and the noise power signal each represent the logarithm of measured power so that the action of the subtractor is equivalent to division. The signals indicative of wanted power, noise power and the signal to noise power ratio are supplied to other components of the radio receiver and used in the control of the radio receiver in a known manner. The construction of the amplitude adaptive filter 16 is illustrated in Figure 2. The noise power signal, in its raw form, is supplied as a stream of values Pk (where k=l, 2, 3 ) from measuring section 15 to an input of multiplier 23. The power gain applied to the recovered signal is supplied in logarithmic format from RF circuitry 12 to formatting unit 21. The logarithmic signal supplied to formatting unit 21 represents the receiver gain in centibels, cB (lOcB = ldB). The formatting unit 21 converts the centibel gain G, into a new representation which is the sum of two factors, M and 3 ON. Accordingly, G = M+30N. To convert the centibel gain into the new representation, the centibel gain G is divided by 30, and the integer part of the result is the value of N. Once N has been determined, M is determined from the equation M=G-30N. As time progresses, the values, P k , of the noise power signal pass through the amplitude adaptive filter 16, and Pt-ι is the preceding (in time) value of the noise power signal and P k+ ι i the succeeding (in time) value of the noise power signal. Accordingly, it is appropriate to apply the same nomenclature to the gain and refer to the logarithmic gain as G . However, it will be appreciated that G will only vary relative to a neighbouring gain value (say Gk + i) if the gain is actually changed by the automatic gain control process in the RF circuitry 12. Similarly, the additive components of the gain representation produced by formatting unit 21 are more properly referred to as M and Nk. Clearly, M k and Nk will only change if G /c changes. Nk will only change when there is a change in the integer part of G÷30. It will be apparent to the reader that each time N increases by one, the linear gain (i.e. 10 G *, and not the logarithmic gain Gt) changes by a factor of approximately 2, that is, JQ(30÷IOO) pjj 2 Conversely, decrementing N by 1 leads to a halving of the linear gain. The value M from formatting unit 21 is supplied to converter 22 which transforms this component of the gain to a linear value of 10 ^ 100 Λ Converter 22 then calculates the inverse X X) X^X) of this linear value, i.e. 10 ^ 100 λ The value 10 ^ J is supplied to the other. input of multiplier 23. Accordingly, multiplier 23 divides Pk by the linear value corresponding to Mk to produce a scaled power value P k . The variation of the noise power values with changes in the linear gain has only been partially removed in that the linear gain remaining in the power values output by multiplier 23 may still vary by a factor which is an exponent of 2. For example, consider that there is a gain change such that G k is not equal to G k+\ ■ If N k = N k+ i, then the gain residually present in the output of multiplier 23 does not change. If, on the other hand, Nk + i≠Nk, then the residual gain present in ^. +1 is a factor of 2 b (b is a positive or negative integer) different from the residual gain present in P k . In summary, it will be seen that variations in the gain have not been eliminated in the output of multiplier 23, but rather the gain in the P k values has been restricted to take only values with ratios of Returning now to formatting unit 21, the value Nk is supplied to a delay unit 24, a comparison unit 25 and to a correction unit 28. The delay unit 24 delays the value of Nk for one sample period of the digital signals (i.e. until N k+ s issued by formatting unit 21) and then presents Nk to an input of the comparison unit 25. Thus, it will be seen that comparison unit 25 always compares consecutive values of N, e.g. N k and N k -i • If the comparison unit detects a change in consecutive values of N, then it instructs a scaling of the signal values of P that are already in digital filter 26. The digital filter 26 performs an averaging operation on the values of P received from multiplier 23. Digital filter 26 produces an output P k in response to an input P k . P k is produced by performing a weighted average on a group of consecutive values of P . The filter coefficients provide the weighting factors in the weighted averaging process performed by filter 26. Consider now the case where the gain G changes so as to affect N such that Nk + ι — Nk = b , where b can take positive or negative integer values. Upon the filter calculating already within the filter 26 will initially need to be multiplied by a factor of 2 b (which is diminishing if b is negative) to ensure that the filtering process is not biased by the difference in the residual gain between P r> A ' +1 and the P values already in the filter. To achieve this multiplication, each of the values of P already within the filter are bit-shifted by a number of bits |b| in the direction of increasing bit-significance if b is positive or, if b is negative, in the direction of decreasing bit significance. Hence, the modification of the P values is achieved without recourse to time consuming arithmetic. The filtered power values P r> k " output by filter 26 are converted from linear values to logarithmic values on the centibel scale by scaling unit 27. The centibel P k values are then supplied to corrector unit 28 which subtracts Nk from P k , which is equivalent to dividing the linear value corresponding to P k by the linear gain factor corresponding to N k . The output of corrector unit 28 is a filtered noise power in centibels that is insensitive to changes in the power gain used to amplify the recovered signal in the RF circuitry 12. Whilst the invention has been described in the context of a signal which is a train of power values, it will be appreciated by the skilled person that the signal values could, equally, represent (for example) the amplitude or level of a signal.

Claims: Claims 1. Signal processing apparatus for operating on a signal that has been amplified under variable gain, the apparatus comprising a digital filter for filtering the signal values, and adapting means for adapting at least one signal value when the gain changes so as to ameliorate the biasing of the filtering operation by the change in gain. 2. Apparatus according to claim 1, wherein the adapting means is arranged to bit-shift said at least one signal value to adapt said at least one signal value. 3. Apparatus according to claim 1 or 2, further comprising means for factorising the gain into two factors, the first factor being the highest power of 2 present in the gain and the second factor being such that when multiplied with the first factor the result is the gain value. 4. Apparatus according to claim 3, wherein the adapting means is arranged to change said at least one signal value upon detecting a change in the first factor. 5. Apparatus according to claim 4, wherein the adapting means is arranged to bit-shift said at least one signal value by a number of bits corresponding to the numerical change in the exponent of the first factor. 6. Apparatus according to any one of claims 3 to 5, further comprising first scaling means for scaling the signal values by the second factor before they reach the filter. 7. Apparatus according to any one of claims 3 to 6, further comprising second scaling means for scaling the signal values by the first factor after they have passed through the filter. 8. Apparatus according to any preceding claim wherein the gain values are logarithmic. 9. Apparatus according to any preceding claim wherein the filter is arranged to average signal values passing through it. 10. Apparatus according to any preceding claim wherein the signal is a noise signal or noise component. 11. Apparatus according to any preceding claim, further comprising means for assessing a wanted signal relative to a noise signal, wherein the filtered signal values provide one of the wanted signal and the noise signal. 12. A method of processing a signal that has been amplified under variable gain, comprising digitally filtering the signal values and adapting at least one signal value when the gain changes so as to ameliorate the biasing of the filtering operation by the change in gain. 13. A method according to claim 12, wherein adapting said at least one signal value comprises bit-shifting said at least one signal value. 14. A method according to claim 12 or 13, further comprising factorising the gain into two factors, the first factor being the highest power of 2 present in the gain and the second being such that when multiplied with the first factor the result is the gain. 15. A method according to claim 14, wherein said at least one signal value is changed upon detecting a change in the first factor. 16. A method according to claim 15, wherein changes to the said at least one signal value are implemented by bit-shifting said at least one signal value by a number of bits corresponding to the numerical change in the exponent of first factor. 17. A method according to any one of claims 14 to 16, further comprising scaling the signal values by the second factor before they are filtered. 18. A method according to any one of claims 14 to 17, further comprising scaling the signal values by the first factor after they have been filtered. 19. A method according to any one of claims 12 to 18, wherein the gain is logarithmic. 20. A method according to any one of claims 12 to 19, wherein the filtering operation averages the power values. 21. A method according to any one of claims 12 to 20, wherein the signal is a noise signal or noise component. 22. A method according to any one of claims 12 to 21, comprising assessing a wanted signal relative to a noise signal, wherein the filtered signal values provide one of the wanted signal and the noise signal. 23. A program which when executed on data processing apparatus performs the method of any one of claims 12 to 22. 24. Signal processing apparatus, substantially as hereinbefore described with reference to Figures 1 and 2. 25. A method of processing a signal, substantially as hereinbefore described with reference to Figures 1 and 2.

Inventors: THURSTON ANDREW, THURSTON, ANDREW
Applicants: THURSTON ANDREW, UBINETICS LTD, THURSTON, ANDREW, UBINETICS LIMITED
Application year: 2001
Main classification: H03G
IPC codes: H03H17, H04B1, H04B17