Hi
I have written Matlab script for BPSK modulation. The way my code works is as follow:
series of 0 and 1----> NRZ data (continuous rectangular pulses with amplitude -1 and 1)--->multiply by carrier----> add noise--> multiply by carrier---> integration--->detection---> Bit Error collection?
I would like to know can anyone tell me where I should apply DAC and ADC ?
I am wondering should it be after NRZ data or before that for DAC ?
:!:
Hi
I have written Matlab script for BPSK modulation. The way my code works is as follow:
series of 0 and 1----> NRZ data (continuous rectangular pulses with amplitude -1 and 1)--->multiply by carrier----> add noise--> multiply by carrier---> integration--->detection---> Bit Error collection?
I would like to know can anyone tell me where I should apply DAC and ADC ?
I am wondering should it be after NRZ data or before that for DAC ?
:!:
You can work on the equivalent discrete channel, too. Anyway, if you insist on continuous time, I think, and I am not sure, you need in this case to define a time vector like t=0:TsK-1)*Ts, where Ts is the symbol time. Then you need to over-sample you data and create the corresponding time vector for the oversampled data. After receiving the signal at the other end, you get rid of the carrier frequency, and down-sample the remaining part. I think this is how it looks like. Remember, in MATLAB, there is nothing like truly continuous signal. You always have to deal with discrete samples in a way.
Thanks I know that I already implemented the code. I am wondering if I want to use data acquisition toolbox in Matlab to do the Analog to digital converter or vice versa where should I apply them. For example lets say I multiply my rectangular pulse by carrier should I apply DAC now ?
Thanks I know that I already implemented the code. I am wondering if I want to use data acquisition toolbox in Matlab to do the Analog to digital converter or vice versa where should I apply them. For example lets say I multiply my rectangular pulse by carrier should I apply DAC now ?
Again, I am not sure, but I think oversampling is the DAC, because each bit is repeated many times within the symbol time interval, which gives the rectangular pulse shape of the symbol. On the other hand undersampling at the receiver is the DAC. Again, double check this, and I hope someone else could comment to affirm or negate what I am saying. I have never worked using continuous-time, but I have this vague idea.