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

Discussion in 'Embedded' started by Guy Macon, May 9, 2006.

  1. Guy Macon

    Guy Macon Guest

    I am doing some preliminary "back of the envelope" design work on
    a high-end split phase / three phase 50 Hz. to 400 Hz. AC power
    supply & analyzer. Advice/comments/ridicule/brickbats welcome...

    Generating the 2 or 3 phases is straightforward, but monitoring
    the current and voltage is a bit more tricky -- especially when
    monitoring the far end of a transformer, line filter, or power
    factor compensator. I wish to determine the following values
    for a single-cycle non-repeating transient:

    Phase A/B/C to N (Neutral) voltage
    Phase A-B/B-C/C-A voltage
    Phase A/B/C/N current
    Phase A/B/C apparent power (VA)
    Phase A/B/C RMS power
    Phase A-B/B-C/C-A voltage phase shift
    Phase A-B/B-C/C-A current phase shift
    Phase A/B/C THD+N

    The first thing that comes to mind is to use some high-end 24-bit,
    192 kHz audio ADCs, possibly interleaving if the sample rate needs
    to be higher. I am leaning toward a sample rate that is divisible
    by two and three (for split-phase 180 degree and three phase 120
    degree measurements) instead of the traditional power of two sample
    rate.

    Let's say that I wish to measure phase with 0.1 degree precision
    at 400 Hz. That's a range of 3600 (0.0 degrees to 359.9 degrees).
    So I need at least 3600 x 400 = 1.44 MSPS, right?

    If I need that many samples per second using 192 kHz audio ADCs,
    I would have to interleave 8 of them per measurement channel,
    times 3 voltage measuring points and 4 current measuring points,
    or 56 of them total (and that's assuming that I can make the
    interleaving work right). Suddenly the low-cost audio parts
    aren't looking so inviting...

    Looking at faster parts, a quick Google search comes up with parts
    such as the Analog Devices AD7621 (16-Bit, 3 MSPS) and AD7641
    (18-Bit, 2 MSPS) and the Linear Technology LTC2208 (16-bit, 130 MSPS).
    Hmmmm... could a fast enough ADC allow me to measure multiple channels
    with a single ADC? Or do I really need simultaneous sampling?

    Or could it be that I am on the wrong track and there is a better
    way than the brute-force lots-of-samples, lots-of-resolution lots-
    of-processing-power solution that is the first thing to comes to mind?
     
    Guy Macon, May 9, 2006
    #1
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  2. 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
     
    Paul Keinanen, May 9, 2006
    #2
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  3. 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
     
    Jim Granville, May 9, 2006
    #3
  4. Guy Macon

    MK Guest

    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
    www.mkesc.co.uk
     
    MK, May 9, 2006
    #4
  5. Guy Macon

    Guy Macon Guest

    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:
    http://www.pearsonelectronics.com/current-monitor-products/standard-current-monitor.htm
     
    Guy Macon, May 10, 2006
    #5
  6. Guy Macon

    Guy Macon Guest

    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.
     
    Guy Macon, May 10, 2006
    #6
  7. Guy Macon

    Guy Macon Guest

    <Big happy smile> 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.

    http://www.calinst.com/ixseries.html



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

    http://www.pacificpower.com/products/amx-series.aspx



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

    http://www.voltech.com/products/pwr_anl/index.htm
    http://www.voltech.com/products/pwr_anl/pm3000/index.htm
    http://www.voltech.com/products/pwr_anl/PM6000/pm6frame.htm



    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!
     
    Guy Macon, May 10, 2006
    #7
  8. 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
     
    Jim Granville, May 10, 2006
    #8
  9. 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
     
    Paul Keinanen, May 10, 2006
    #9
  10. Guy Macon

    Guy Macon Guest

    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, May 12, 2006
    #10
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