A
Anonymous
Guest
I have decided to begin a digital PI project. There is a simple block diagram below that shows what I have in mind. I will sample the received signal with an analog to digital converter, then integrate and filter the samples to produce an output signal that can be modulated for a speaker.
I have completed phase 0 of the project. That is the part that does not have much to do with PI. I looked at some issues about sampling and digital filtering before deciding if I even wanted to try doing this. My conclusion so far is that if the amplifier bandwidth is about 10 to 20 kHz then I can directly sample the output of the amplifier, but I will be at a disadvantage compared to conventional PI's that have a sample window of 10 usec or longer. That is, I expect to have a worse signal to noise ratio on the raw samples than an analog PI would have. On the other hand I think I can do a better job of integrating the samples than do the few PI integrators I have seen. Maybe I can make up for the worse SNR with better filtering.
Phase 1 of the project will be to piggyback a digital processor on a normal PI board. I will pick off the coil drive signal to find the turn off time, then sample the received signal at certain intervals after turn off. The digital samples will be integrated and filtered and output for comparison with the output of the PI integrator. If I cannot get results at this point that are approximately equal to the output from the PI integrator then the project will be over. If I can get comparable results then I will go on to the next phase.
The 10 bit A-D in the processor I have been using in my VLF project is a little slow for what I want to do now. It is rated at 65 usec per conversion. I would like to be able to take samples closer together than that, and I don't want to load the board up with sample and holds. So I will switch to another board that has a faster 5 usec 12 bit A-D. Unfortunately, this board uses a different instruction set, so I will have to rewrite all my filter routines. Also I don't have a PI board to piggyback on yet. So I will have to see about getting one by the time I get all the software written.
Phase 2 (if I get that far) will be the diagram below, a single board in which the processor does all the timing, sampling, filtering, and as much of the audio as possible.
Robert
I have completed phase 0 of the project. That is the part that does not have much to do with PI. I looked at some issues about sampling and digital filtering before deciding if I even wanted to try doing this. My conclusion so far is that if the amplifier bandwidth is about 10 to 20 kHz then I can directly sample the output of the amplifier, but I will be at a disadvantage compared to conventional PI's that have a sample window of 10 usec or longer. That is, I expect to have a worse signal to noise ratio on the raw samples than an analog PI would have. On the other hand I think I can do a better job of integrating the samples than do the few PI integrators I have seen. Maybe I can make up for the worse SNR with better filtering.
Phase 1 of the project will be to piggyback a digital processor on a normal PI board. I will pick off the coil drive signal to find the turn off time, then sample the received signal at certain intervals after turn off. The digital samples will be integrated and filtered and output for comparison with the output of the PI integrator. If I cannot get results at this point that are approximately equal to the output from the PI integrator then the project will be over. If I can get comparable results then I will go on to the next phase.
The 10 bit A-D in the processor I have been using in my VLF project is a little slow for what I want to do now. It is rated at 65 usec per conversion. I would like to be able to take samples closer together than that, and I don't want to load the board up with sample and holds. So I will switch to another board that has a faster 5 usec 12 bit A-D. Unfortunately, this board uses a different instruction set, so I will have to rewrite all my filter routines. Also I don't have a PI board to piggyback on yet. So I will have to see about getting one by the time I get all the software written.
Phase 2 (if I get that far) will be the diagram below, a single board in which the processor does all the timing, sampling, filtering, and as much of the audio as possible.
Robert