The invention relates to an apparatus and method for isolating I and Q components of a received IF signal and for filtering the received signal. In particular, the invention relates to an apparatus and method for performing the combined functions of a digital down converter and a filter.
A Digital Down Converter (DDC) is a key component of digital receivers in many communication systems. The DDC is some kind of digital mixer and is used to recover the in-phase I and quadrature Q components of a received signal.
As shown in
Sk=Ak cos{2πfIFkTs+φk}
where k is the sample number (also known as the order), Ak is amplitude of the sampled IF signal at sample Sk, fIF is the intermediate frequency, Ts is the time between one sample and the next i.e. the sampling interval and φk is the phase of the sampled IF signal at sample k.
The I channel is formed by multiplying Sk by cos{2πfIFkTs} at block 101 and the Q channel is formed by multiplying Sk by −sin{2πfIFkTs} at block 103. The resulting I and Q components therefore have the form:
Ik=Ak cos{2πfIFkTs+φk}cos{2πfIFkTs}
and
Qk=−Ak cos{2πfIFkTs+φk}sin{2πfIFkTs}.
Usually, the DDC is followed by some sort of low pass filter. The low pass filter primarily reduces noise outside the required bandwidth but may also perform pulse shaping, to reduce the bandwidth of the signal without introducing intersymbol interference. The filter may be a simple low pass filter (LPF) or a raised cosine filter or a root raised cosine (RRC) filter or any other suitable type of filter. Because the I and Q components are already isolated by the DDC, two filters are required (one for the I channel and one for the Q channel) as shown in
In some cases, the DDC can be simplified. For example, if the sampling frequency fs of the DDC is four times the intermediate frequency fIF i.e. fs=4fIF, then we can simplify the sine and cosine functions. This is because, over one cycle, cos x takes the values 1, 0, −1 and 0 and −sin x takes the values 0, −1, 0 and 1. So, to produce the I channel, we multiply the incoming signal Sk by 1, 0, −1 and 0 at successive samples and, to produce the Q channel, we multiply the incoming signal Sk by 0, −1, 0 and 1 at successive samples. This produces I components of the forms Ik=Sk,0,−Sk,0 over a single cycle of the intermediate frequency signal and Q components of the forms Qk=0,−Sk,0,Sk over a single cycle of the intermediate frequency signal.
Some known arrangements make use of this DDC simplification to also simplify the subsequent filters. One such known arrangement is shown in
It is an object of the invention to provide a method and apparatus for performing the combined functions of a DDC and filter, which mitigate or substantially overcome the problems associated with known arrangements described above.
According to a first aspect of the invention, there is provided apparatus for isolating an in-phase component I and a quadrature component Q of a received intermediate frequency (IF) signal and for filtering the received signal, the apparatus comprising:
The apparatus acts as a combined digital down converter (DDC) and filter i.e. it performs two functions. Firstly, it isolates the I and Q components from the received IF signal. Secondly, it filters the received signal to reduce noise outside the required bandwidth. By using separate filter portions for the odd k and even k samples, so that the total number of taps in the two portions is n, the filter can provide the performance of a n-tapped filter while allowing each filter portion to have about half the complexity of a n-tapped filter.
Preferably, the DDC is arranged to input samples where k is even into the first filter portion and to input samples where k is odd into the second filter portion. Preferably, n is odd and
Thus, the first filter portion handles samples k=0, 2, 4, . . . , n−1 and the second filter portion handles samples k=1, 3, 5, . . . , n−2. An odd value of n is preferred because this means that the peak of the filter's frequency response is defined by a single value rather than two equal values. However, n could be even, in which case x may equal n/2 and y may equal n/2.
In one embodiment, the first and second filter portions are finite impulse response (FIR) filters. In that embodiment, in one case n=25. In that embodiment, pulse shaping is performed only in the transmitter which transmitted the signals and no pulse shaping is performed in the filter.
In an alternative embodiment, the filter may be arranged to perform pulse shaping of the received signal. In that embodiment, in a first arrangement, the first and second filter portions are finite impulse response (FIR) filters. In that case, the apparatus may further comprise apparatus for pulse shaping the received signal. The pulse shaping apparatus may comprise a raised cosine filter (in which case, the FIR filters perform low pass filtering, the raised cosine filter(s) perform pulse shaping, and no pulse shaping is performed in the transmission side). Alternatively, the pulse shaping apparatus may comprise a root raised cosine (RRC) filter (in which case, the FIR filters perform low pass filtering, the RRC filter(s) perform pulse shaping, and some pulse shaping is also performed in the transmission side).
In that embodiment, in a second arrangement, the first and second filter portions each comprise all or part of a raised cosine filter. In that case, the first and second filter portions may each comprise a root raised cosine (RRC) filter. In that embodiment, in one case, n=97. If the first and second filter portions each comprise a RRC filter, there will usually be at least one other RRC filter in the transmitter which transmitted the signals. The at least one RRC filter in the receiver together with the at least one RRC filter in the transmitter together provide raised cosine function pulse shaping of the signal.
The apparatus may further comprise a differential decoder for performing differential detection of I and Q over a given symbol span. The differential decoder may be arranged to perform differential detection of I and Q over a symbol span of one symbol. Of course, other symbol spans can also be envisaged.
The differential decoder may comprise a decision block for converting the differentially decoded I into an I output and for converting the differentially decoded Q into a Q output, the I output and the Q output each taking a value of either 0 or 1.
In one embodiment, the apparatus further comprises a converter for converting the received signal to a digital signal. In that embodiment, the converter may be an analogue to digital converter (ADC). The ADC may operate at a frequency which is four times the frequency of the received signal. Alternatively, in that embodiment, the converter may be a hard limiter.
In one arrangement, the apparatus further comprises a decimator for reducing sampling frequency of the received signal. This is particularly advantageous if the apparatus includes a hard limiter for digitizing the received signal. In that embodiment, the hard limiter may operate at a frequency which is sixteen times the frequency of the received signal and the decimator may reduce the frequency to four times the frequency of the received signal. A decimator may also be included if the apparatus uses an ADC for digitizing the received signal.
The decimator may be a cascaded integrator comb (CIC) filter. A CIC filter is advantageous for performing decimation since it does not include multipliers.
The received signal may be a differentially encoded phase shift keyed (DPSK) signal. In one example, the received signal is
modulated.
According to the invention, there is also provided a receiver for intermediate frequency signals, the receiver comprising apparatus according to the first aspect of the invention.
According to a second aspect of the invention, there is provided a method for isolating an in-phase component I and a quadrature component Q of a received intermediate frequency (IF) signal and for filtering the received signal, the method comprising the steps of:
Preferably, n is odd and
Thus, the first filter portion handles samples k=0, 2, 4, . . . , n−1 and the second filter portion handles samples k=1, 3, 5, . . . , n−2.
The first and second filter portions may be finite impulse response (FIR) filters.
The method may further comprise the step of pulse shaping the received signal. In that case, the first and second filter portions may each comprise a finite impulse response (FIR) filter; then, the pulse shaping is performed separately. Alternatively, the first and second filter portions may each comprise all or part of a raised cosine filter. The first and second filter portions may each comprise a root raised cosine (RRC) filter. In that case, there will usually be at least one other RRC filter in the transmitter which transmitted the signals. The at least one RRC filter in the receiver together with the at least one RRC filter in the transmitter together provide raised cosine function pulse shaping of the signal.
The method may further comprise the step of performing differential detection of I and Q over a given symbol span. The differential detection of I and Q may be performed over a symbol span of one symbol. The method may further comprise the steps of converting the differentially decoded I into an I output and converting the differentially decoded Q into a Q output, the I output and the Q output each taking a value of either 0 or 1.
The method may further comprise the step of converting the received signal to a digital signal. In one embodiment, the step of converting the received signal to a digital signal is performed in an analogue to digital converter (ADC). In that embodiment, the ADC may operate at a frequency which is four times the frequency of the received signal. In an alternative embodiment, the step of converting the received signal to a digital signal is performed in a hard limiter. In that embodiment, the hard limiter may operate at a frequency which is sixteen times the frequency of the received signal.
The method may further comprise the step of reducing sampling frequency of the received signal. The step of reducing the sampling frequency may be performed in a cascaded integrator comb (CIC) filter. A CIC filter is advantageous for performing decimation since it does not include multipliers.
The received signal may be a differentially encoded phase shift keyed (DPSK) signal. The received signal may be
modulated.
According to the invention, there is also provided apparatus for carrying out a method according to the second aspect of the invention. According to the invention, there is also provided a receiver for intermediate frequency signals, for carrying out a method according to the second aspect of the invention.
According to a third aspect of the invention, there is provided a method for isolating an in-phase component I and a quadrature component Q of a received intermediate frequency (IF) signal and for filtering the received signal, the method comprising the steps of:
Features described in relation to one aspect of the invention may also be applicable to another aspect of the invention.
A known arrangement has already been described with reference to FIGS. 1 to 3 of the accompanying drawings, of which
By way of example, preferred embodiments of the invention will now be described with reference to FIGS. 4 to 8 of the accompanying drawings, of which:
a, are frequency plots showing how the invention may be used with 8b and 8c bandpass sampling.
Two embodiments will be described and both show receivers for
but the invention is not limited to
and could apply to any IF signal, whether or not differentially encoded phase shift keyed.
intermediate frequency (IF) receiver 401 according to a first embodiment of the invention. The receiver includes an IF Hard Limiter 403, a Cascaded Integrator Comb (CIC) filter 405, a combined DDC and filter 407 comprising a DDC 409, and Root Raised Cosine (RRC) filters 411a and 411b and a differential decoder and decision block 413. The input to the receiver 401 is an analogue IF signal and the outputs of the receiver 401 are I and Q signals.
The IF Hard Limiter 403 receives the analogue IF signal and performs digitization. The output of the IF Hard Limiter 403 is a 2-level i.e. 1-bit IF signal. The Hard Limiter in this embodiment operates at a sampling rate fs which is sixteen times the frequency of the IF signal i.e. fs=16fIF.
The CIC filter 405 performs decimation of the signal from the IF Hard Limiter 403. A CIC filter is very efficient for performing decimation (or interpolation) since it does not contain multipliers. The CIC filter 405 may have its usual structure i.e. an integrator portion comprising N integrator stages clocked at rate fs, a downsampler for reducing the sampling rate to fs/R and a comb portion comprising N comb stages clocked at rate fs/R. In this embodiment, R=4 so that the sampling rate at the output of the CIC filter 405 and the input to the DDC 409 is
As already discussed, the purpose of a DDC is to isolate the I and Q components of the digitized IF signal. The purpose of an RRC filter is to perform pulse shaping and to reduce noise outside the required bandwidth. A standard RRC comprises a number n of taps in succession. Any number n can be chosen for the RRC filter (as long as the filter's frequency response meets the system requirements) but we usually prefer an odd number of taps so that the centre of the filter's impulse response will be a peak (rather than two equal values). Also, the greater the number of taps, the more attenuation there will be outside the desired bandwidth, but the greater the filter complexity and delay. In the combined DDC and filter 407, the two functions (of a DDC and of RRC filter(s)) are not strictly separated and the combined DDC and filter 407 overall performs the two functions together.
In this case, the sampling rate at the input to the combined DDC and filter 407 is 4fIF so we can make use of the simplification described above with reference to
Thus, in the DDC 409 we take samples (of order k) four times in each cycle of the received signal. The even ordered samples are input into the I channel RRC filter 411a and the odd-ordered samples are input into the Q channel RRC filter 411b. By separating odd and even ordered samples, the effect is to isolate the I and Q components. RRC filter 411a is used for the I channel so is only interested in even-ordered samples i.e. where k=0, 2, 4, 6 . . . . Thus, if we want the overall effect of an n-tapped RRC filter where n is odd, in RRC filter 411a, we actually only need to implement
taps. The
taps will be for the even-ordered samples (k=0, 2, 4, . . . , n−1), because the odd-ordered samples will always equal zero. Similarly, RRC filter 411b is used for the Q channel so is only interested in odd-ordered samples i.e. where k=1, 3, 5, 7 . . . . Thus, if we want the overall effect of an n-tapped RRC filter where n is odd, in RRC filter 411b, we actually only need to implement
taps. The
taps will be for the odd-ordered samples (k=1, 3, 5, . . . , n−2), because the even-ordered samples will always equal zero.
Thus, filter 411a has
taps and filter 411b has
taps, making a total of n taps between the two filters. In this embodiment, n=97 so that filter 411a has 49 taps and filter 411a has 48 taps. Thus, the filters between them have exactly the same performance as a 97-tap filter but the complexity of each filter is approximately halved.
In the combined DDC and filter 407, we are able to perform filtering as well as I and Q isolation. By combining the DDC and filter functions together, we are able to approximately halve the complexity of each filter without introducing the degradation associated with prior art arrangements like that shown in
Referring once again to
Iout(k)=Iin(k)*Iin(k−1)+Qin(k)*Qin(k−1)
Qout(k)=Qin(k)*Iin(k−1)−Iin(k)*Qin(k−1)
After differential decoding, the I and Q signals are input into the decision portion of block 413. The decision rule might be something like
If Iout>0, I=0 else I=1
If Q3>0, Q=0 else Q=1
or any other suitable decision algorithm.
intermediate frequency (IF) receiver 601 according to a second embodiment of the invention. The receiver includes an Analogue-to-Digital Converter (ADC) 603, a combined DDC and filter 605 comprising a DDC 607 and Finite Impulse Response (FIR) filters 609a and 609b, a differential decoder 611 and a decision block 613. As with the first embodiment, the input to the receiver 601 is an analogue IF signal and the outputs of the receiver 601 are I and Q signals.
In this embodiment, the ADC 603 converts the received analogue signal to a digital signal. The operating sampling frequency fs of the ADC is four times the frequency of the IF signal i.e. fs=4fIF. Note that, in the first embodiment, signal digitization was performed by the IF hard limiter 403 whereas, in this second embodiment, signal digitization is performed by the ADC 603. Both arrangements have advantages and disadvantages: an ADC will usually be able to operate at a lower sampling rate than a hard limiter which will result in a lower power consumption, but the complexity and circuit size of an ADC based system could be higher than that of a hard limiter based system.
In this embodiment, because of the lower sampling rate of the ADC, no decimation of the signal is required. So, no CIC filter (or other downsampler) is included in the receiver 601. In other embodiments, however, a decimator might be required with the ADC.
As usual, the purpose of a DDC is to isolate the I and Q components of the digitized IF signal. As before, since the sampling rate of the DDC is 4fIF, we can simplify the cosine function (for the I channel) by multiply the incoming signal by 1, 0, −1 and 0 at successive samples and we can simplify the sine function (for the Q channel) by multiplying the incoming signal by 0, −1, 0 and 1 at successive samples. So, once again, we are only interested in the even orders (k=0, 2, 4, . . . ) for the I channel and in the odd orders (k=1, 3, 5, . . . ) for the Q channel.
In this embodiment, we are using simple FIR filters 609a and 609b rather than RRC filters because we do not require any pulse shaping in the receiver 601. This may be the case if all the pulse shaping has already been performed in the transmitter or if we are performing some or all of the pulse shaping elsewhere in the receiver side. Thus, the purpose of the FIR filters is simply to select the required bandwidth and cut out noise outside that bandwidth. Thus, the FIR filters act as simple low pass filters. A standard FIR filter comprises a number n of taps in succession, the number of taps being chosen in accordance with the desired frequency response.
In the DDC 607, we take samples (of order k) four times in each cycle of the received signal. The even ordered samples are input into the I channel FIR filter 609a and the odd-ordered samples are input into the Q channel FIR filter 609b. By separating odd and even ordered samples, we are effectively able to isolate the I and Q channels. FIR filter 609a is used for the I channel so is only interested in even-ordered samples i.e. where k=0, 2, 4, 6 . . . . Thus, if we want the effect of an n-tapped FIR filter where n is odd, in FIR filter 609a, we actually only need to implement
taps. The
taps will be for the even-ordered samples (k=0, 2, 4, . . . , n−1), because the odd-ordered samples will always equal zero. Similarly, FIR filter 609b is used for the Q channel so is only interested in odd-ordered samples i.e. where k=1, 3, 5, 7 . . . . Thus, if we want the effect of an n-tapped FIR filter where n is odd, in FIR filter 609b, we actually only need to implement
taps. The
taps will be for the odd-ordered samples (k=1, 3, 5, . . . , n−2), because the even-ordered samples will always equal zero.
Thus, filter 609a has
taps and filter 609b has
taps, making a total of n taps between the two filters. In this embodiment, n=25 so that filter 609a has 13 taps and filter 609b has 12 taps. Thus, the filters have exactly the same performance as a 25-tap filter but the complexity of each filter is approximately halved.
Just like in the combined DDC and filter 407 shown in
Referring once again to
Iout(k)=Iin(k)*Iin(k−1)+Qin(k)*Qin(k−1)
Qout(k)=Qin(k)*Iin(k−1)−Iin(k)*Qin(k−1)
In
After differential decoding, the I and Q signals are input into the decision block 613. The decision rule might be something like
If Iout>0, I=0 else I=1
If Qout>0, Q=0 else Q=1
or any other suitable decision algorithm.
It should be noted that, in both first and second embodiments described above, the total number of taps in the filter (n) is an odd number. This is usually preferred because it results in the frequency peak of the filter's frequency response being defined by a single value rather than two equal values. However, other embodiments could be envisaged where n is an even number. In that case, the I channel filter (whether an RRC, an FIR, an LPF or another type of filter) would most likely implement n/2 taps and the Q channel filter (whether an RRC, an FIR, an LPF or another type of filter) would most likely implement n/2 taps.
An application of the invention will now be described with reference to
signals.
In this application, bandpass sampling is used to perform signal digitization. Bandpass sampling is described in Gary J. Saulnier, etc “A VLSI Demodulator for Digital RF Network Applications: Theory and Results”, IEEE Journal on Selected Areas in Communications, Vol. 8. No. 8 pp. 1500-1511, October 1990 and also in Rodney G Vaughan, etc “The Theory of Bandpass Sampling”, IEEE Transactions on Signal Processing, Vol. 39, No. 9 pp. 1973-1984, September 1991. The idea of bandpass sampling is that the sampling rate can be lowered from twice the maximum frequency of the signal 2fMAX to 2(fMAX−fMIN) where the signal being sampled is bandpass i.e. has non-zero content only in the band between fMIN and fMAX. That is, in this invention, the sampling rate can be lower than the frequency of the IF signal.
a shows the signal spectrum in the analogue domain i.e. before analogue to digital conversion. The signal is centered on the IF frequency.
b shows the signal spectrum in the digital domain i.e. after analogue to digital conversion using bandpass sampling. If the intermediate frequency is fIF and the sampling frequency is fs, in order to be able to make the required DDC simplification of the invention, we require
and, as shown in
Note that the sampling rate should be selected carefully to make sure no aliased signal appears at half the sampling rate.
In one example, shown in
So, in this example, one of the aliased signals appears at 8 MHz, which is one quarter of the sampling rate. Thus, we can make use of the DDC simplification as described above.
Number | Date | Country | Kind |
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200505405-1 | Aug 2005 | SG | national |