ADC/DAC for 50-400Hz 3-phase?

How are you actually going to measure the phase shift between phases or between voltage and current ?

Trying to detect the time difference between the zero crossings would work in an ideal situation with perfectly clean sinusoids, however, in practice, the mains voltage is badly polluted by all kinds of noises. With even order harmonic distortion, the half waves are different, so it is difficult to even establish the zero-line.

Trying to detect the top of the waveform would even be worse.

So in practice, multiple mains cycles would be required to determine the phase difference to such accuracy. As long as the sampling frequency is not an _exact_ multiple of the mains frequency, a repetitive waveform will be accurately reconstructed anyway.

Calculating the complex FFT for a sufficiently long sample should give the phase and magnitude for the fundamental and for each harmonics in the waveform.

Thus, I do not think that you would need such huge sampling frequency for the phase measurement, but of course for transient recording the high sampling rate is desirable.

There are a few problems with typical audio ADCs. The DC and low frequency characteristics seem to be poor, sometimes artificially limited at 3 Hz to avoid drift problems. This may be an issue if you want to measure any DC component on the mains (which will increase transformer noise and ultimately saturate the cores even with a few volts of imbalance).

Even if some ADCs boost 192 kHz and 24 bits, the SNR quoted is at best around 120 dB and this is usually specified for the 20 kHz audio bandwidth only. The SNR for the full 90+ kHz bandwidth possible with

192 kHz sample rate can be much worse, corresponding to 18-20 bit ideal ADCs.

How did you intend the current measurement ? Have you looked at the current transformer phase and frequency response or are you planning to use shunting resistors in each phase and using separate floating power supplies for each ADC at mains phase potential and bring down the digital sample values in an optical fiber (and apparently feed the ADC with a common clock source using an other optical fiber) ?

Paul

Why do you need 0.1' / 24 bit ? Most standards specs I've seen quote harmonics only to a certain number, and if you are doing an analyser do you need to be 70x that ?

Many Audio DACS have poor impulse response - a 20KHz response is gives out to the 50th harmonic of 400Hz.

-jg

Hello Guy,

You are indeed on the wrong track,

To measure everything about a 400 Hz signal you need to sample at 800Hz and a bit - in real life you need more samples than this to deal with non ideal anti-alias filters etc. (and don't forget that uncertainty in the sample time (jitter) will affect accuracy just as much as amplitude errors).

Most mains analysers allow you to measure up to 20th Harmonic - 1kHz for

50Hz mains, 8kHz for your 400Hz top limit. A 40kHz sample rate will do fine. I wouldn't use audio parts for this - they are a pain to multiplex. There are single chip parts with 4 and 8 way multiplexors that should work quite well. You get on much better measuring the phase if you use all the samples rather than just the few near the zero crossing - Google for Goertzel - there is a lot of 'noise' but you should find something. You don't need simultaneous sampling - just to know when your samples were sampled - you can calculate out any errors due to the exact sampling time.

Michael Kellett

What I usually do is to calculate a Fourier transform on the signal, discarded all harmonics leaving only the fundamental, calculate a reverse Fourier transform to reconstruct the signal, then measure the zero crossing.

Alas, my customers really do want to measure transient waveforms. For example, when load X was suddenly connected to power, how many cycles did it take to recover?

Indeed it does, and one of the requirements of an AC power source is to shut down if there is more than a small amount of DC at the output. I am likely to design in a seperate circuit for that.

Audio bandwidth is fine; my signal is in the 50 to 400 Hz. range.

24 bits is massive overkill given the noise and distortion typical of real-world powerlines. I don't think I will end up using the cheap audio ADCs, but it won't be because of SNR or LF issues.

Look here:

24 bits is a huge overkill, it's just what the 192KHz low-cost audio ADCs put out. Measuring to a fraction of a degree is real; the users of these things tend to drive large inductive loads and to add capacitor banks to correct the power factor, and they do that by measuring the phase between voltage and current.

Now *that's* the kind of advice that I was hoping for! Far better to find out now when I am doing preliminary design work...

A California Instrument 9003iX does waveform analysis and waveform synthesis to the 51st harmonic and has a 16 Hz to 500 Hz Hz full power bandwidth.

A Pacific Power Source 320AMX does waveform analysis and waveform synthesis to the 51st harmonic and has a 20-5000 Hz full power bandwidth.

A Voltech PM300 measures up to 250kHz. A PM3000A goes to 1 MHz, and a PM6000 goes to 10MHz.

It seems that somebody out there seems to think that you have to go past a few KHz.

I was under the impression that the Goertzel Algorithm was a way to detect frequecies with less computation than a DFT or FFT. I usually do an FFT, throw out all the harmonics, reconsruct a fundamental-only version of the signal, and them meaure the zero crossing point.

Hmmm. That makes sense, but I have never done it. If it turns out that I am wrong about bandwidth and can multiplex, I will revisit this.

I doubt that I will be multiplexing. Too much bandwidth loss. If I end up using audio parts I may end up interleaving them!

Could you not derive phase for power factor terms very accurately, by using a whole-cycles samples. You do not need _each_ sample to be 0,1', all you need is the phase of the 'best fit sine'.

Quite a low number of samples will give you that : If you know the exact times of the samples. ~0,5us time delta is ~1 part in

5000, and 12 bit ADC is ~1 part in 4096 ad that's better than 0.1'

That leaves you with transient capture, which sounds partly real, and partly 'bragging rights' stuff.

For the numbers game, look at the TMS320F28x, they have 12bit ADCs with 12.5 Msps, and enough crunching you could do fancy compression on the transient info : Store the delta from a ideal sine - so you don't waste storage on clean waveforms....

Or ADIs newest ADuC7128 - that has 1 MSps ADCs, 12 bit, DACs and a DDS core as well.

-jg

It depends what you want to measure, if you are interested in measuring lightning current waveforms, then at least 1 MHz sample rate should be used, since the current slew rate is several kA/us and the peak is reached in the order of 10-30 us.

For transients in typical man made electric systems, the transient current is limited by the circuit inductances (including wire inductances) and hence the current slew rate. Voltage transients are limited by the stray capacitances. Thus you would have to analyse what kind of voltage and current transients are expected in your environment, before deciding the sample rate.

Paul

I have been doing some more "back of the envelope" design work on a split phase / three phase 50 Hz to 400 Hz AC power source. The comments some here have posted about ACD/DAC bandwidth have been very helpful and are much appreciated.

I obtained a couple of different kinds of commercially available AC Power sources and spent a couple of days running tests on them in order to see what I am competing against. I found that the real-world bandwidth limit isn't fixed at a particular roll-off but is instead slew-rate limited, and thus the small signal bandwidth -- and the number of useful harmonics that are used in the signal synthesis function -- is considerably better than the bandwidth at full voltage swing.

As a thought experiment, imagine a 400 Hz AC power source that has just enough output-DAC bandwidth to generate a 400 Hz sine wave, bumping right up against the Nyquist limit. This wouldn't be enough in the real world, where one very common set of waveforms that the users desire are sine waves with hard clipping at 1%, 5%, 10% THD etc. Then again, some of them want square waves or triangle waves... This requires more DAC bandwidth, but the question is, how much? I am inclined to design in enough DAC bandwidth so that the slew-rate-limiting of the output stage dominates from 10% to 100% amplitude, and to give the waveform- capture ADC a bandwidth of maybe twice that. Comments?

As always, many of these decisions will change as the design progresses; I am just setting a starting point for the preliminary design work.

--
Guy Macon