hello,erevyone.Recently I have been designing a blind channel equalizer based on the Constant Modulus Alogrithm.But I have a question that I can't think over.The question is as follows: The alogrithm I use is below: y(k)=3D =A1=C6f(k)x(N-k); % f(k+1)=3Df(k)-mu*e(k)*x(k); % e(k)=3D4*conj(y(k))*(abs(y(k))^2-R2); the transmitted signal is modulated by the QPSK.The samples out of AD must be real,I guess. If I was right ,then how to deal with the conjucate in the DSP program.If I am wrong,please tell me why. Many thanks.
cma equalizer
Started by ●April 18, 2006
Reply by ●April 18, 20062006-04-18
How come you have real samples if the constellation is QPSK. This modulation format implies complex samples, or a real I and a real Q pair per sample. Phil
Reply by ●April 18, 20062006-04-18
tirst of all,thank you for you reply.Now I know,when equalizing the qpsk signal, I have to sample the I and Q channel respctivly,where the I and Q stand for the real part and imaginant part.However I met another question,that is the simulating property is not stable.Sometimes,it's very good,then sometimes it becomes very bad.The property relates to the channel .But it's the blind equalizer.How can I guarntee the implemention on dsp is good? Adjusting the step size mu ,or increase the SNR,and the length of the equalizer? I find when the step size becomes smaller,the result does not become better. Any suggestion appreciate.
Reply by ●April 19, 20062006-04-19
If I understand correctly, you are observing the equaliser lock and then loose lock on your signal. I assume that you are basing this on observation of e(k). Actually, to avoid inacurate assumptions and wasted time, could you better explain the channel dynamics and the algorithms/circuits that precede and follow the equaliser? Also clarify what you are observing when you state that it is not stable. It is easier to make a recommendation if the mechanism(s) causing the problem can be isolated.
Reply by ●April 20, 20062006-04-20
thanks,what i was doing is the blind channel equalizer implemented in the dsp.The alogrithm i used is the Constant Modulus Alogrithm.the signal input is the BPSK . My tutor said to me that when checking the results,what i only to is generate a BPSK signal with the signal generator,and let that pass a channel,then the channel output is sent to the DSP to equalize.But the channel property i don't know.when simulating in matlab,i used a Ralay channel and the QPSK signal.the chanel model likes this:h=randn(1,5)+j*randn(1,5). When the signal is bpsk,i remove the imagination ,so the chanel becomes:h=randn(1,5).the next step is :x=fliter(h,1,s),and x=x+randn(1,T),I think this is the channel output,however my tutor said to me he don't know the channel.I puzzled.how can i choose the channel model? I don't know. any help or suggestion appreciated.
Reply by ●April 21, 20062006-04-21
Have you run the BPSK scenario in MATLAB as well? If not, how do you know what to expect? This should be the first thing you do so that you have a basis for comparison. Further, make sure that the data and channel are exactly the same through both the MATLAB model and the DSP. What information are you monitoring from the DSP? Apart from the data samples, are you observing the e(k) and f(k)? If this comes down to a DSP debugging problem, I am of no use. I have no experience in coding up DSPs. All my algorithms have been for others to implement in FPGA and ASIC. Sorry for not being more helpful.
Reply by ●April 21, 20062006-04-21
yes, i have to change my qpsk scenario to bpsks,because there are not enough AD channels for the I and Q channel.so i change this to bpsk scheme.i think you are quite right that make sure that the data and channel are exactly the same through both the matlab model and the DSP.now i have some knowledge about my bpsk scenario,and i have realized the data flow in matlab.what i observed is the (1/N)*sum(e(k)^2),where the N stands for the number of realization.and also with the constellation diagrams before and after equalization. but i have met new troubles.when i check the equipment ,i find that the bpsk is 250 K symbol/s,while my AD conversion is 22K,maybe in the end i have to do some bpsk modulation and demodulation in hardware or what i don't know, In the end ,thanks sincerely for you helps,Phil.
Reply by ●April 24, 20062006-04-24
Sorry that I couldn't be more help. It sounds like your sampling rates are definitely a problem. I wasn't sure if you were actually converting your signal to analog or not. If you can get around your rate problems, perhaps you can use the same input to provide your QPSK I and Q samples in serial. So long as you sort them properly in the DSP, you can then test your algorithm that way. Just an idea.
Reply by ●April 24, 20062006-04-24
Sampling the I and Q channel serialy,that sounds like OK,but it needs strict sampling time control.As to the specification, i don't know. Any suggestion? thanks
Reply by ●April 24, 20062006-04-24
for the time control, you can precede your data with a known data pattern (preamble) that you lock onto. For example, have the preamble look like "1 -1 1 -1 1 -1 1 1 -1 -1". When the data goes from nothing to this pattern, you can make your algorithm begin the I and Q pairing. Your tutor may be able to help you with this.