Very high performance low frequency AWG? - Page 1
Forum section: electronics / testgear

author: maxwell3e10
author_rank: Frequent Contributor
timestamp: December 27, 2020, 11:14:37 pm
content: I am looking for state-of-the-art arbitrary waveform generator at very low frequencies, 1-100 Hz. It needs to have an external clock input and continuous data streaming (or large memory depth). An obvious choice would be a fancy audio DAC, but I am not sure how good is their performance near DC. Perhaps an audio analyzer would work if it can be configured in that way. Does anyone have any experience in this area?

author: nctnico
author_rank: Super Contributor
timestamp: December 28, 2020, 12:32:23 am
content: The Analog Discovery 2 comes to mind. Not that it is a high end piece of gear but it does have dual 14bit DACs so perhaps the API allows continous streaming to it. IMHO worth checking out.

author: DaJMasta
author_rank: Super Contributor
timestamp: December 28, 2020, 02:26:09 am
content: What makes your requirements state of the art? If your maximum frequency is 100Hz, your data rate is so minutely small that almost anything should be able to stream it if it supports data streaming, so the high performance requirements are what? High resolution converter? High stability oscillator? Extremely fine frequency gradations?If you have a stable sample clock generator, maybe you just need a 24 bit DAC evaluation board.

author: maxwell3e10
author_rank: Frequent Contributor
timestamp: December 28, 2020, 03:16:07 am
content: The requirements are for very low noise and THD, so it needs to be at least 24 or 32 bit. I was looking at some high-end audio DACs. The output stage and power supply is also tricky, so I would prefer a complete instrument.

author: _Wim_
author_rank: Super Contributor
timestamp: December 28, 2020, 08:34:02 am
content: Quote from: maxwell3e10 on December 28, 2020, 03:16:07 am24bit is -144.5 dB, so actual 24 bit performance will not be possible. If you look at the specs of the APx555 (https://www.ap.com/analyzers-accessories/apx555/), I think this will be close to the best currently achievable (the APx555 does not go down to 1Hz and does not have an AWG function I think, but it should give you an idea about the performance of a low frequency DAC)At 1Hz flicker noise will probably become the dominating factor, so if 1Hz performance is of high priority, probably some compromises will have to be made at higher frequencies...

author: _Wim_
author_rank: Super Contributor
timestamp: December 28, 2020, 08:38:41 am
content: I wonder if you could not use an SMU for this purpose?Edit: probably not. THD Specs are not given for the SMU's I checked, but looking at for example the specs of an HP 3245a, THD is really bad...

author: justanothername
author_rank: Regular Contributor
timestamp: December 28, 2020, 09:32:00 am
content: What about those DC-coupled audio interfaces like those from MOTU?

author: maxwell3e10
author_rank: Frequent Contributor
timestamp: December 28, 2020, 02:48:30 pm
content: Yes, I was looking at APx555 or some high end audio DACs. Some of them are DC coupled. I think APx555 can go to DC, but I haven't been able to find a loop-back noise spectrum at very low frequency. If someone has access to it, would be interesting to see how much 1/f noise it has.As for SMUs, they typically don't have clock signals accessible.

author: Sighound36
author_rank: Frequent Contributor
timestamp: December 28, 2020, 02:58:49 pm
content: Not sure if any of the high end approach DC maybe 6-8Hz?Maybe this would get you there?https://www.rohde-schwarz.com/us/product/upvb1-productstartpage_63493-8398.htmlWhy not build one as it very specific request?

author: maxwell3e10
author_rank: Frequent Contributor
timestamp: December 29, 2020, 03:12:55 am
content: The low distortion generator in R&S UPV as well as in APx555 is an analog generator for making pure sine waves. It wouldn't work for arbitrary waveforms.For building such AWG it seems the best option is still one of the audio chips, they can get about 21-22 bits dynamic range. Another option would be 20 bit R-2R converters such as DAC11001A or AD5791. Such DAC could be better at very low frequency, but generally is worse, for example THD is -105 dB for DAC11001A. For audio DACs there are dozens of companies trying to make low-noise systems with no costs spared. So I think even by accident one of them would have very good performance at low frequency that would be hard to replicate starting from scratch. Its just a matter of finding one. These fancy DACs cost about $1k, which is OK but one wouldn't want to test a lot of them.

author: bdunham7
author_rank: Super Contributor
timestamp: December 29, 2020, 05:01:51 am
content: If you are looking for and AWG plus amplifier with SINAD levels down in the Audio Precision domain, that is certainly state-of-the-art. And while I agree that an audio product is most likely to be your easiest answer, there are a lot of really, really bad (from a technical perspective) products out there at very high prices. And to get one that accepts your arbitrary external clock reduces the field even further. Many DAC products do have external clock, but I'm not sure that they can all accept inputs that are not a standard audio format clock (44.1kHz, 192kHz, etc).You might direct an inquiry to Amir Mejidimehr at Audio Science Review. He likes to obsess about the technical aspects of audio equipment, especially DACs and the like. He may know of a ready-made commercial product that can do what you want directly from a PC, or one that can be easily modified to do so. Do you even need a specific clock? Since your frequencies (and presumably bandwidth) are so low, could you work with say a 384kHz clock requirement?

author: maxwell3e10
author_rank: Frequent Contributor
timestamp: December 29, 2020, 05:14:18 am
content: Yes, I've already come across Audio Science Review and Amir's measurements, its quite impressive. They are mostly done above 20 Hz, but hopefully someone would be willing to look at lower frequencies. The sample rate is not so important, I can just sample necessary waveforms at 44.1 or 384 kHz and make an audio file. External clock is kind of funny, it seems that as far as audio performance it's not really necessary, so its kind of a fad. So the DACs that have external clock are usually bad in analog domain, while the best ones in the analog domain don't bother with the clock. But I think there are some reclocking options one can figure out.

author: nctnico
author_rank: Super Contributor
timestamp: December 29, 2020, 11:28:26 am
content: Quote from: maxwell3e10 on December 28, 2020, 02:48:30 pmAt low frequencies wander of the DAC's clock is likely to become an issue as well. You'll probably want a good OCXO as a clock source.

author: Sighound36
author_rank: Frequent Contributor
timestamp: December 29, 2020, 01:14:41 pm
content: nctnico observation is a good one, a quality clock will be required for this project one which I agree withI have never measured anything below 16hz on a real high end dac (£10K+) before both with an R&S UPV 66or a FSW43 (though that starts @ 2Hz!) and that was using a ultra low phase noise reference clock -122dbc @ 1hz as 10Mhz master clock

author: maxwell3e10
author_rank: Frequent Contributor
timestamp: December 29, 2020, 02:17:55 pm
content: Well, for short-term measurements the clock quality is not so crucial at low frequency, its less important than for higher frequency measurements. But for long-term operation it is important. That is what I meant by the need for external clock. Not necessarily the sample clock, but a reference 10 MHz clock input.

author: Sighound36
author_rank: Frequent Contributor
timestamp: December 29, 2020, 02:49:51 pm
content: Quote from: maxwell3e10 on December 29, 2020, 02:17:55 pmMaxwell,That is what I was discussing a 10Mhz master ref significantly better better than any Rh clock (best one I have used was -101dbc @1hz topping out at -156dbc @ 10Mhz) I have tried this far though I suspect its long term stability will be still good in 10,000 years or so in fairnessThe one I use in the lab (and in the audio system are identical) with 10Mhz you are looking @ -174dbc

author: nctnico
author_rank: Super Contributor
timestamp: December 29, 2020, 03:39:24 pm
content: Quote from: Sighound36 on December 29, 2020, 02:49:51 pmIf you need short and long term stability then don't rule out a relatively simple GPSDO. Short term an OCXO can get near Cesium clock stability levels. Rubidium clocks have poor short term stability so these will need an extra cleanup oscillator to improve short term noise.Still I'm wondering what kind of device the OP is going to drive which requires such precise signal generation.

author: Sighound36
author_rank: Frequent Contributor
timestamp: December 29, 2020, 03:55:52 pm
content: Hi nctnicoI would agreed a quality GPSDO for the shorter term is a great suggestion I have a few around the lab they work well and can be put together for sensible sums.Have to say the purpose was intriguing myself as well

author: maxwell3e10
author_rank: Frequent Contributor
timestamp: December 30, 2020, 04:58:53 am
content: Maybe there is something I am missing here... Say I have a reference clock at 10 MHz with noise of -100dBc/Hz at 1 Hz. Now I divide it to generate a signal at 10 Hz. That should reduce phase noise by 20 log (10^6), giving -220 dBc/Hz -totally negligible. For higher frequency signals, like 10 kHz, one gets -160 dBc/Hz, so it starts to become an issue.

author: 1audio
author_rank: Frequent Contributor
timestamp: December 30, 2020, 06:29:36 am
content: There are several prosound audio interfaces that may well do what you want. The DC response is not necessarily there (most are AC coupled) but I suspect either RMEhttps://www.rme-audio.de/home.htmlor Lynxhttps://www.lynxstudio.com/may be able to help with simple mods. The prosound units do support external sync. And you can get a Rubidium stabilized clock fromhttps://en.antelopeaudio.com/products/10mx/and some others. You want a clock at an audio master clock frequency so everything works as expected.Another option would be a QuantAsylum QA401 (or QA402)https://quantasylum.com/products/qa401-audio-analyzerOne problem you will find is that the AKM factory fire has essentially stopped production of a lot of the premium DAC products. ESS is another DAC chip vendor and some pro products use theirs.You may be able to find a used pro interface that will do what you need. I have a Lynx 1 PCI interfacehttps://www.bhphotovideo.com/c/product/232262-REG/Lynx_Studio_Technology_LYN_L1_LynxONE_Sound_Card.html/specsthat is limited to 96 KHz sampling but otherwise offers excellent performance I'll pass along for a pretty nominal price. PM me if interested.

author: nctnico
author_rank: Super Contributor
timestamp: December 30, 2020, 10:25:26 am
content: Quote from: maxwell3e10 on December 30, 2020, 04:58:53 amIt doesn't work that way. And the divider will also add noise.Can you elaborate a little bit on what you are trying to achieve? Maybe there is a better way.

author: Sighound36
author_rank: Frequent Contributor
timestamp: December 30, 2020, 10:35:07 am
content: Quote from: nctnico on December 30, 2020, 10:25:26 amAbsolutely without question nctnico, have been trying to crack that nut for around 15 years

author: _Wim_
author_rank: Super Contributor
timestamp: December 30, 2020, 05:11:51 pm
content: Quote from: nctnico on December 30, 2020, 10:25:26 amCan you also explain which phenomenon causes that? I cannot seem to wrap my head around why this would be. Intuitively is seems because the 10 MHz pulse to pulse variation has a normal distribution and the divider takes a very long average of all those pulses, noise should go down as was stated by maxwell3e10.I am sure you and Sighound36 are correct, I just do not understand why...

author: nctnico
author_rank: Super Contributor
timestamp: December 30, 2020, 05:34:05 pm
content: Quote from: _Wim_ on December 30, 2020, 05:11:51 pmI'll take a stab at it: noise has an infinite frequency spectrum and a divider just counts edges until it reaches the division factor. What you basically get is a decimated signal. IOW: If you divide the 10MHz signal then the edges of the resulting wave will still wander around with the same amount of noise (assuming the divider is perfect and doesn't add any noise).If you want to reduces the noise by averaging cycles then you can use an oscillator with a good stability which slowly follows the incoming clock signal.Another option is to use several parallel clock sources and then combine these into a single clock signal which is an average of all sources. This is how UTC time is made. They have a whole bunch of high quality clocks which are then combined into a single clock output.

author: _Wim_
author_rank: Super Contributor
timestamp: December 30, 2020, 07:45:55 pm
content: Thanks for your explanation. Together with that and this document (https://www.ti.com/lit/an/slaa566/slaa566.pdf?ts=1609353715993&ref_url=https%253A%252F%252Fwww.google.be%252F) I think I now understand.What I did not realize before is that the typical phase noise of a 10MHz clock already had plenty of low frequency content by itself. I understood before that the divided clock would wander around by the same amount as the original clock, but thought this amount would now be insignificant. That was not correct, because this amount also already has low frequency content also.

author: maxwell3e10
author_rank: Frequent Contributor
timestamp: January 02, 2021, 07:18:02 pm
content: The notes that _Wim_ referenced clearly state in the conclusion: "The divider can lower noise from the clockregarding as the ratio of clock frequency to DAC output frequency". One can also see it from time jitter equation t_j=sqrt(Integral(phase noise))/(2 Pi f). So, one can write Integral(phase noise_DAC)=Integral (phase noise_clock)*(f_DAC/f_Clock)^2. So, even if the DAC simply uses 10 MHz clock to trigger directly, the phase noise is much reduced. In practice a proper design would use a low-phase noise local oscillator with a slow PLL loop to 10 MHz clock. But at 1 Hz offset it will be limited by the 10 MHz clock with the estimates shown before.

author: _Wim_
author_rank: Super Contributor
timestamp: January 02, 2021, 08:21:54 pm
content: The way I understood the paper and the explanation from nctnico is that in the case were pulses are counted, the jitter remains identical (last counted clock cycle still moves around like all other clock cycles).In this paper however, the divider represents the oversampling of the DAC, and oversampling is like averaging, hence also reducing the noise. So I think both "dividers" are not equal, but it could be I misunderstood.

author: maxwell3e10
author_rank: Frequent Contributor
timestamp: January 02, 2021, 10:22:36 pm
content: The jitter in time units is identical, but it represents a smaller fraction of the period for a lower frequency sine wave and hence a lower phase noise.

author: Sighound36
author_rank: Frequent Contributor
timestamp: January 03, 2021, 01:24:29 pm
content: Looking forward to seeing your solution in action Maxwell