I am facing two problems:
c1psl, c1susaux, and c1auxey today
MC autolocker got stuck (judging by wall StripTool traces, it has been this way for ~7 hours) because c1psl was unresponsive so I power cycled it. Now MC is locked.
I swapped the inputs to the ZHL-3A at the PSL table - so now the X beat RF signals from the beat mouth are going through what was previously the Y arm ALS electronics. From Attachment #1, you can see that the Y arm beat is now noisier than the X. The ~5kHz peak has also vanished.
So I will pursue this strategy of switching to try and isolate where the problem lies...
Somebody had forgotten to turn the HEPA variac on the PSL table down. It was set at 70. I set it at 20, and there is already a huge difference in the ALS spectra
[rana, kevin, udit, gautam]
quick notes of some discussions we had today:
RXA: 0805 size SMD thin film resistors have been ordered from Mouser, to be shipped on Monday. **note that these thin film resistors are black; i.e. it is NOT true that all black SMD resistors are thick film**
We use D990694 in various places. Today, Rana alerted me to an important consideration to be kept in mind when we use this board, which I found quite interesting. I still don't understand the problem at the BJT level, but I think one can appreciate the problem without going to the transistor design of the LT1125. I'm attaching an annotated schematic of the whitening section in question. If the following assumptions are valid, then I think my picture is valid.
Then, as one can see in the attached schematic, when we set the gain of any input to <24dB, we must ensure that the input voltage is less than approximately 2V. Otherwise, by asking too much of the first stage op-amp in the quad IC LT1125, we may be messign around with all the 4 op amps in the quad! Even the 0dB setting is not immune to this problem, as it uses one of the 4 op amps.
Now that I think about this a bit more - this problem shouldn't be significant for the usual LSC degrees of freedom when in lock, as the huge DC gain of the loop should squish large DC values of the error signals, and so there shouldn't be any danger of overloading the LT1125. But I don't know if we are being hurt by this effect when flashing through resonances, when the PDH horn-to-horn voltage can be quite high (which is in principle a good thing?). I don't know if there is any "hysterisis" effect where the overloaded quad IC has some relaxation time before it returns to normal operation, and if we are being limited in our ability to catch lock because if this effect.
The concerns remain valid for th ALS demodulated error signals though, for which the signals will remain large throughout.
After discussing with Koji, we looked at the aLIGO incarnation of this board. Interestingly, it too has a similar topology of 4 switchable gain stages with gains of 24, 12, 6 and 3dB. The main differences are that they use single Op27 ICs instead of the quad LT1125s, and also, they use a different combination of feedback resistors to realize the various gains.
We considered upping the feedback resistance (R15, R143) on the 24dB gain stage of our boards from (1k, 66.5ohms) to (3k, 200ohms) as on the aLIGO boards - but this doesn't really help? Because KCL demands that the same current flow in R15 and R143, and so the output Vsat of the op amp and its max current driving capabilities in combination determine if the inverting input can follow the non inverting input?
As Hartmut points out in his note, he was able to access the full range of ADC voltages when the gain was set to 3dB, despite the fact that the LT1125 was still getting internally saturated. Operating with minimum 24dB whitening gain doesn't really solve the problem either because the problem just gets shifted to the next gain stage in the chain, and we still have saturation. I also don't have a feeling for how much differential voltage these LT1125s can sustain before they are damaged - I guess the planned THD check will reveal if they are okay or not.
It seems to me like the only way to truly fix this problem of one stage saturating and screwing up the others is to use single Op27s (or equivalent) in place of the quad LT1125s. The aLIGO design also has a series resistance to the non-inverting input - this can help prevent current overdraw from the previous stage (due to a lowered input impedance of the OpAmp - but I wonder how low this can go?).
this is the note from Hartmut Grote on this topic from 2004
I have acquired 5 pieces of the Teledyne AP1053 from Koji - these are now at the 40m. I will determine an appropriate location for storage of these and update. We are also looking to acquire 5 more of these. The combination of high power output (26dBm), low gain (10dB), and low noise figure (1.5dB) are quite uncommon in an amplifier and so these should be used only when such properties are required simultaneously.
*Steve informs me that these amps have been stored in the RF cabinet E6 along the east arm.
Steve's note: Teledyne rf amp product selection guide
Teledyne rf low noise amp guide
I did some work on the PSL table today. Main motivations were to get a pickoff for the BeatMouth PSL beam before any RF modulations are imposed on it, and to improve the mode-matching into the fiber. Currently, we use the IR light reflected by the post doubling oven harmonic separator. This has the PMC modulation sideband on it, and also some green leakage.
So I picked off ~8.5mW of PSL light from the first PBS (pre Faraday rotator), out of the ~40 mW available here, using a BS-80-1064-S. I dumped the 80% reflected light into the large beam dump that was previously being used to dump this PBS reflection. Initially, I used a R=10% BS for S-pol that I found on the SP table, but Koji tipped me off on the fact that these produce multiple reflected beams, so I changed strategy to use the R=80% BS instead.
The transmitted 20% is routed to the West edge of the PSL table via 2 1" Y1-1037-45S optics, towards the rough vicinity of the fiber coupler. For now it is just dumped, tomorrow I will work on the mode matching. We may want to cut the power further - ideally, we want ~2.5mW of power in the fiber - this is then divided by 4 inside the beat mouth before reaching the beat PD, and with other losses, I expect ~500mW of PSL power and comparable AUX light, we will have a strong >0dBm beat.
Attachment #1 is a picture of my modifications. For this work, I
I plan to do some characterization of this problem. The plan is to use THD as a metric for whether we are having hidden saturations. Pg 9 of the LT1125 datasheet tells us what fraction of THD to expect. I will use one of the several unused DAC channels available at the LSC rack to drive a 100Hz sine wave into one of the inputs of the whitening chassis, and measure the THD up to a reasonable harmonic number (will probably be set by the ADC noise) for (i) various whitening gain settings and (ii) various input signal amplitudes.
The motivation is to attempt to quantify the problem better:
Then we can decide what, if anything, to do about this issue.
On Friday, while Udit and I were doing some characterization of the EX+PSL IR beat at the LSC rack, I noticed that there were sidebands around the main beat peak at 20dBm lower level. These were offset from the main peak by ~200kHz - I didn't do a careful characterization but because of the symmetric nature of these sidebands and the fact that they appeared with the same offset from the main peak for various values of the central beat frequency, I hypothesize that these are from the modulation sidebands we use for PDH locking the EX laser to the arm cavity. So we can estimate the modulation depth from the relative powers of the main beat peak and the ~200kHz offset sidebands.
Since the IR light is used for the beat and we directly couple it to the fiber to make the beat, there is no green or IR cavity pole involved here. 20dBm in power means . And so the modulation depth, . I will do a more careful meaurement of this, but this method of measuring the modulation depth can give us a precise estimate - for what it's worth, this number is in the same ballpark as the measurement I quote in elog12105.
What is the implication of having these sidebands on our ALS noise? I need to think about this, effectively the phase noise of the SR function generators we use to do the phase modulation of the EX laser is getting imprinted on the ALS noise? Is this hurting us in any frequency range that matters?
I was looking into the physics of polarization maintaining fibers, and then I was trying to remember whether the fibers we use are actually polarization maintaining. Looking up the photos I put in the elog of the fibers when I cleaned them some months ago, at least the short length of fiber attached to the PD doesn't show any stress elements that I did see in the Thorlabs fibers. I'm pretty sure the fiber beam splitters also don't have any stress elements (see Attached photo). So at least ~1m of fiber length before the PD sensing element is probably not PM - just something to keep in mind when thinking about mode overlap and how much beat we actually get.
I was looking at this a little more closely. As I understand it, the purpose of the audio differential IF amplifier is:
Attachment #1 shows, the changes to the TF of this stage as a result of changing R19->50ohm, R17->500ohm. For the ALS application, we expect the beat signal to be in the range 20-100MHz, so the 2f frequency component of the mixer output will be between 40-200MHz, where the proposed change preserves >50dB attenuation. The Q of the ~500kHz resonance because of the series LCR at the input is increased as a result of reducing R17, so we have slightly more gain there.
At the meeting yesterday, Koji suggested incorporating some whitening in the preamp itself, but I don't see a non-hacky way to use the existing PCB footprint and just replace components to get whitening at audio frequencies. I'm going to try and measure the spectrum of the I and Q demodulated outputs with the actual beat signal to see if the lack of whitening is going to limit the ALS noise in some frequency band of interest.
Does this look okay?
The demod circuit board is configured to have gain of x100 post demod (conversion loss of the mixer is ~-8dB). This works well for the PDH cavity locking type of demod scheme, where the loop squishes the error signal in lock, so most of the time, the RF signal is tiny, and so a gain of x100 is good. For ALS, the application needs are rather different. So we lowered the gain of the "Audio IF amplifier" stage of the circuit from x100 to x10, by effecting the resistor swaps 10ohms->50ohms, 1kohm->500ohms (more details about this later).
I tried to couple the PSL pickoff into the fiber today for several hours, but got nowhere really, achieved a maximum coupling efficiency of ~10%. TBC tomorrow... Work done yesterday and today:
I think part of the problem was that the rejected beam from the PBS was not really very Gaussian - looking at the spot on the beam profiler, I saw at least 3 local maxima in the intensity profile. So I'm now switching strategies to use a leakage beam from one of the PMC input steering optics- this isn't ideal as it already has the PMC modulation sideband on it, and this field won't be attenuated by the PMC transmission - but at least we can use a pre-doubler pickoff. This beam looks beautifully Gaussian with the beam profiler. Pics to follow shortly...
I tried to couple the PSL pickoff into the fiber today for several hours, but got nowhere really, achieved a maximum coupling efficiency of ~10%. TBC tomorrow... Work done yesterday and today
I've often gotten confused by the labeling on the SUS MEDM screens about the coil "Vmon" fields - they're labelled as "30 Hz HPF", and indeed this is one of the many readbacks available on the coil driver board. But the actual EPICS channel that is being displayed in this field is from the "EPICS VMON" monitor point on the coil driver board. It has a gain of 1/2, so the actual voltage going to the coil is twice the channel value. Today, I fixed the SUS master screen to avoid this confusion - new labeling is shown in Attachment #1.
Attachment #1 shows the current situation of the PSL table IR pickoff. It isn't the greatest photo but it's hard to get a good one of this setup. Now there is no need to open the Green PSL shutter for there to be an IR beat note.
All this lead me to conclude that I have reached at least some sort of local maximum. The AR coating of the lens has ~0.5% reflection at 8 degrees AOI according to spec, and EricG mentioned today that the fiber itself probably has ~4% reflection at the interface due to there not being any special AR coating. There is also the fact that the mode of the collimator isn't exactly Gaussian. Anyways I think this is a big improvement from what was the situation before, and I am moving on to debugging the ALS electronics.
There is 3.65mW of power coupled into the fiber - our fiber coupled PDs have a damage threshold of 2mW, and this 3.65mW does get split by 4 before reaching the PDs, but good to keep this number in mind. For a quick measurement of the PMC and X end PDH modulation depth measurements, I used an ND=0.5 filter in the beam path.
I used the Beat Mouth to make a quick measurement of the PMC and EX modulation depths. They are, respectively, 60mrad and 90mrad. See Attachments #1 and #2 for spectra from the beat photodiode outputs, monitored using the Agilent analyzer, 16 averages, IF bandwidth set to resolve peaks offset from the main beat frequency peak by 33.5MHz for the PMC and by ~230kHz for the EX green PDH.
For this work, I had to re-align the IFO so as to lock the arms to IR. c1susaux was unresponsive and had to be power-cycled. As mentioned in the earlier elog, to avoid saturating the Fiber Coupled beat PDs, I placed a ND=0.5 filter in the fiber collimator path, such that the coupled power was ~1mW, which is well inside the safe regime.
For the EX modulation depth, I could have gotten multiple estimates of the modulation depth using the higher order products that are visible in the spectrum, but I didn't.
We installed some rails to mount the 2U chassis containing ~100m of delay line cabling, and the 1U chassis containing the FET demodulators for the ALS signals in the LSC rack. This has made it MUCH easier for a single person to work there and remove/reinstall these chassis. The delay line box has 100m of cable inside it, and so was rather heavy (~8kg) - previously, it was being supported only by a pair of brackets on the front, so the new arrangement is much more robust. Steve is looking into acquiring plastic spacers of the appropriate width, so that we can secure the units to the rack using usual rack mount screws (but the material of the newly installed rails and the screw heads holding them in place necessitate this plastic spacer).
Delay line box has been re-installed, demodulator chassis has been removed by me for characterization. Steve will put up photos once the units are re-installed.
For this work, I had to disconnect a bunch of cabling, but only those connected to ALS. All cables were labelled, and I will re-connect them once I am done with the demod chassis.
Anyways I think this is a big improvement from what was the situation before, and I am moving on to debugging the ALS electronics.
The plan is to lower the gain of the IF amplifier stage on the FET demodulator board from 100 to 10. As per Attachment #1, this will make the overall gain from RF beatnote from the Beat Mouth to the signal input to the D990694 whitening board +19dB, assuming "typical" values for the conversion loss of the mixer, and the various other passive components on the FET demod board. I've used numbers I measured a couple of weeks ago for the delay line loss and the cabling loss from the PSL table to the LSC rack. This in turn will set a limit on how much RF beat power we can handle, from the Beat Mouth. According to this power budget, if we have -5dBm of beat, we will have an input to the whitening board of ~6Vpp, which is about half its full range. The trouble is, I don't know what the transimpedance gain of the Fiber Beat PDs are. The datasheet suggests a "maximum gain" of 5e4 V/W, which presumably takes into account the InGaAs responsivity and the actual transimpedance gain. However, according to the last power budget I did inside the Beat Mouth, I had -8dBm of beat for a combined 400uW of PSL+EX light, which definitely does not add up. I've emailed the company to ask about the spec, haven't gotten anything useful yet...
The problem is further complicated by the fact that the fiber inside the Beat Mouth is NOT polarization maintaining, and so the actual relative polarizations of the arm IR light and the PSL IR light is unpredictable, and also uncontrolled. I suppose we could simply place a HWP before the fiber collimator at either end, and rotate the polarization until we get a desired amount of beat, but this still does not solve the problem of the polarization being uncontrolled.
I am going to characterize the demod board using E1100114. I am unsure as to the conversion loss of the mixer - the datasheet suggested a number of 8dB, but T1000044 suggests that the conversion loss is actually only 4dB. I figure it's best to just measure it. Would also be good to verify that the overall transfer function and noise of the IF amplifier stage match my expectation from the LISO model.
Option #1: Rana ordered 50ohm and 500ohm SMD resistors of the 0805 package size, I asked Steve to get a few more values just in case we want to twiddle with the gain of this stage further (specifically, I asked for values such that we can set it to x5, x3 and x1). But changing the feedback resistors modifies the overall TF shape - see e.g. Attachment #2. Need to also look at how the noise performance varies.
Another possibility is to turn down the gain of the IF amplifier stage to x10, retire the ZHL-3A, and use a lower gain amplifier in its place. We do have the recently acquired Teledyne amplifiers, but we would have to package it in such a way that it can be integrated into the existing Fiber ALS signal chain. This would allow us to handle significantly larger RF beatnote powers, which I expect we will have if we improve the mode matching into the fibers (provided the aforementioned polarization drift possibility doesn't hurt us too much).
A third possibility is to attenuate the power coupled into the fibers to lower the RF beatnote amplitude. I don't like this option so much because placing an ND filter or a PBS+HWP combo in the beam path is likely to screw up the mode-matching into the fiber collimator, which I have already spent so many hours trying to improve, but if it must be done, it must be done.
The correct option is of course the one that gives us the lowest ALS noise. It is not clear to me which one that is at this point.
I effected the change to the Audio IF preamp stage on channels 3 and 4 (Xarm and Yarm respectively) using the resistors Steve ordered (the ones Rana ordered don't have any labeling on them, and I couldn't tell the 50ohm and 500ohm ones apart except by looking at the label on the ziplock bag they came in, so I decided against using them). I've started a DCC page to collect photos, characterization data, and marked up schematic etc for this part. Characterization is ongoing, more to follow soon. Note that for the photo-taking, I disconnected all the on-board SMA connectors so that the cabling wouldn't block components. I have since restored them for testing purposes, and was careful to use the torque-limited SMA tightening tool when restoring the connections.
In order to test various things like conversion loss etc, I figured it would be useful to have two RF signal sources, so I scavenged the Fluke RF generator that Johannes was using from under the PSL table. In the process, I accidentally bumped the PSL interlock on the southeast corner of the PSL table. I immediately turned the NPRO back on, and relocked PMC/IMC. Everything looks normal now. Acromag may even have caught my transgression.
Stuff is beginning to look clearer now that I've done some initial characterization of the demod boards. I will upload a more detailed report of the characterization on the DCC page, but important findings are:
The delay line has a loss of ~3dB. The power splitter has a loss of 3dB. So putting everything together, 17dBm at the input of the power splitter gives us just the right amount of RF power to have the LO input driven at 14dBm, and the IF output be ~5Vpp into a High-Z load, which is about half the ADC full range.
I saw some interesting behaviour of the Audio IF amplifier stage on the demod board today, by accident. I was testing the board for I/Q orthogonality and gain balance, when I noticed a large gain imbalance between the I and Q channels for both Board #3 and #4, which are the ones we use for the IR ALS demodulation. This puzzled me for some time, but then I realized that I had only reduced the gain of this stage from x100 to x10 for the I channel, and not for the Q channel! The surprising thing though was that the output waveform still looked like a clean sinusoid on the o'scope, and there was no evidence of the voltage clipping that is characteristic of an op-amp being driven beyond its voltage rails. The conversion factor with a preamp gain on x10 was measured today to be 2V IF / 1V RF. But this means that for a preamp stage gain of x100, we expect 20V IF / 1V RF, which is well in the saturation regime of the AD829, since the Vcc is only +/-15V. I'm guessing the diodes D2 and D3 are for overvoltage protection, but given that the pre-amp gain is x100, the input signal at the inverting input of the AD829 is only 0.2V at DC, which isn't above the forward bias voltage for the switching diode BAV99. Perhaps there is some interaction between the pre-amp and the FET demodulator that I dont understand, or I am missing something about the differential to single-ended topology that would explain this behaviour.
I found it puzzling why the large preamp stage gain didn't hurt us with the green beat - even though the green optical beat signal was smaller than the current IR beat, a back-of-the-envelope calculation suggested that it would still have saturated the ADC with a x100 gain on the preamp. Perhaps this observation is part of the story, and there is also the unpredictable behaviour of the D990694 board for an input signal with large DC levels...
I did the following tests on this board today:
I didn't really measure the transfer function of the preamp stage after the modification because there wasn't a convenient test point and I couldn't find the high impedance FET probe for the Agilent - I wonder if somebody in WB has it? Anyways, all the tests suggested the board is operating as expected, and I now have calibrations for the back panel DSUB for LO/RF power levels, and also the conversion gain from RF to IF. I will put together a python notebook with all my measurements and upload it to the DCC page for this part. I need to double check expected noise levels from LISO to match up to the measurement.
I will now proceed to the next piece (#3?) of this puzzle, which is to understand how the D990694 which receives the signals from this unit reacts to the expected DC voltage level of ~4Vpp.
After discussion with Koji, I have also decided to look into putting together a daughter board for an alternative Audio IF preamp stage. The motivation is that for the ALS application, we expect a high DC signal level all the time (because the loop does not suppress the beat note amplitude). So we would like for the preamp stage to have the usual shape of some zero around 4Hz, a pole around 40Hz, and then the LowPass profile of the existing preamp stage (to cut out the 2f frequency product, but also to minimize the possibility of the fast AD829 going into some unpredictable regime where it oscillates). So, the desired features are:
While setting up for this measurement, I noticed something odd with the whitening switching for the ALS channels. For the usual LSC channels, the whitening is set up such that switching FM1 on the MEDM screen changes a BIO bit which then enables/disables the analog whitening stage. But this feature doesn't seem to be working for the ALS channels - I terminated all 4 channels at the LSC rack, and measured the spectrum of the IN1 signals with DTT in the two settings, such that I expect to see a difference in the spectra if the whitening is enabled or disabled - FM1 enabled (expected analog whitening to be engaged) and FM1 disabled (expected analog whitening to be bypassed). But I see no difference in the spectra. I confirmed that the BIO bit switching is happening at least on the software level (i.e. the bit indicator MEDM screens indicate state toggling when FM1 is ON/OFF). But I don't know if something is amiss in the signal chain, especially since we are using Hardware channels that were previously used for AS_165 and POP_55 signals.
Is the whitening shape such that we expect the terminated noise level to be below ADC Noise even when the whitening is engaged? I just checked the shape of the de-whitening filter, and it has -40dB gain above 150Hz, so the inverse shape should have +40dB gain.
I will now proceed to the next piece (#3?) of this puzzle, which is to understand how the D990694 which receives the signals from this unit reacts to the expected DC voltage level of ~4Vpp
gautam 2.15pm: This was a FALSE ALARM, with the inputs terminated, the electronics noise really is that low such that it is buried under ADC noise even with +40dB gain. I cranked up the flat whitening gain from 0dB to 45dB for the X channels (but left the Y channels at 0dB). Attachment #2 is the comparison. Looks like the switching works just fine.
The netgpibdata scripts are now under git version control at /opt/rtcds/caltech/c1/scripts/general/labutils/netgpibdata. I think the idea was to make this directory a collection of useful utilities that we could then pull at various labs / at the sites.
I could not understand why 'netgpibdata' scripts are missing in "scripts/general" folder on pianosa... Where did they go???
I've been trying to setup for the THD measuremetn at the LSC rack for a couple of days now, but am plagued by a problem summarized in Attachment #1: there are huge harmonics present in the channel when I hook up the input to the whitening board D990694 to the output of a spare DAC channel at the LSC rack. Attachment #2 summarizes my setup. I've done the following checks in trying to debug this problem, but am no closer to solving it:
Am I missing something obvious here? I think it is impossible to do a THD measurement with the spectrum in this condition...
Did some quick additional checks to figure out what's going on here.
So either something is busted on this board (power regulating capacitor perhaps?), or we have some kind of ground loop between electronics in the same chassis (despite the D990694 being differential input receiving). Seems like further investigation is needed. Note that the D000316 just two boards over in the same Eurocrate chassis is responsible for driving our input steering mirror Tip-Tilt suspensions. I wonder if that board too is suffering from a similarly noisy ground?
I think I've narrowed down the source of this ground loop. It originates from the fact that the DAC from which the signals for this board are derived sits in an expansion chassis in 1Y3, whereas the LSC electronics are all in 1Y2.
Looking at Jamie's old elog from the time when this infrastructure was installed, there is a remark that the signal didn't look too noisy - so either this is a new problem, or the characterization back then wasn't done in detail. The main reason why I think this is non-ideal is because the tip-tilt steering mirrors sending the beam into the IFO is controlled by analogous infrastructure - I confirmed using the LEMO monitor points on the D000316 that routes signals to TT1 and TT2 that they look similarly noisy (see e.g. Attachment #1). So we are injecting some amount (about 10% of the DC level) of beam jitter into the IFO because of this noisy signal - seems non-ideal. If I understand correctly, there is no damping loops on these suspensions which would suppress this injection.
How should we go about eliminating this ground loop?
We discussed possible solutions to this ground loop problem. Here's what we came up with:
Why do we care about this so much anyways? Koji pointed out that the tip tilt suspensions do have passive eddy current damping, but that presumably isn't very effective at frequencies in the 10Hz-1kHz range, which is where I observed the noise injection.
Note that all our SOS suspensions are also possibly being plagued by this problem - the AI board that receives signals is D000186, but not revision D I think. But perhaps for the SOS optics this isn't really a problem, as the expansion chassis and the coil driver electronics may share a common power source?
gautam 1530 7 Feb: Judging by the footprint of the front panel connectors, I would say that the AI boards that receive signals from the DACs for our SOS suspended optics are of the Rev B variety, and so receive the DAC voltages single ended. Of course, the real test would be to look inside these boards. But they certainly look distinct from the black front panelled RevD variant linked above, which has differential inputs. Rev D uses OP27s, although rana mentioned that the LT1125 isn't the right choice and from what I remember, LT1125 is just Quad OP27...
After emailing the technical team at Menlo, I have uploaded the more detailed information they have given me on our wiki.
The trouble is, I don't know what the transimpedance gain of the Fiber Beat PDs are. The datasheet suggests a "maximum gain" of 5e4 V/W, which presumably takes into account the InGaAs responsivity and the actual transimpedance gain.
Summary of my tests of the demod boards, post gain modification:
Everything looks within the typical performance specs outlined in E1100114, except that the measured noise levels don't quite line up with the LISO model predictions. The measurement was made with the scheme shown in Attachment #1. I didn't do a point-by-point debugging of this on the board. I have uploaded the data + notebook summarizing my characterization to the DCC page for this part. I recommend looking at the HTML version for the plots.
*I'd put up the wrong attachment, corrected it now...
I will put together a python notebook with all my measurements and upload it to the DCC page for this part. I need to double check expected noise levels from LISO to match up to the measurement.
gautam 9 Feb 2018 9pm: Adding a useful quote here from the LISO manual (pg28). I think if I add the Johnson noise from the output impedance of the mixer (assumed as 50ohms, I get better agreement between the measured and observed noises (although the variance between the 4 channels is still puzzling). The other possible explanation is small variations in the voltage noise at the various mixer output ports. Could we also be seeing the cyclostationary shot noise difference between the I and Q channels?
In any case, I am happy with this level of agreement, so I am going to stick this 1U chassis back in its rack with the primary aim of measuring a spectrum of the beatnote, so that I have some idea of what kind of whitening filter shape is useful for the ALS signals. May need to pull it out again for actually implementing the daughter board idea though... I have updated DCC page with LISO source, and also the updated python notebooks.
We did a survey of the lab today to figure out some of the logistics for the PID control test for the seismometer can. Kira will upload sketches/photos from our survey. Kira tells me we need
There are no DAC channels available in the c1ioo rack. In fact, there is a misleading SCSI cable labelled "c1ioo DAC0" that comes into the rack 1X3 - tracing it back to its other end, it goes into the c1ioo expansion chassis - but there are no DAC cards in there, and so this cable is not actually transporting any signals!
So I recommend moving the whole setup to the X end (which is the can's real home anyways). We plan to set it up without the seismometer inside for a start, to make sure we don't accidentally fry it. We have sufficient ADC and DAC channels available there (see Attachments #1 and #2, we also checked hardware), and also Sorensens to power the heater circuit / temperature sensing circuit. Do we want to hook up the Heater part of this setup to the Sorensens, which also power everything else in the rack? Or do we want to use the old RefCav heater power supply instead, to keep this high-current draw path isolated from the rest of our electronics?
I have attached the sketch of the whole system (attachment 3) with all the connections and inputs that we will need. Attachment 4 is the rack with the ADC and DAC channels labeled. Attachment 5 is the space where we could set up the can and have the wires go over the top and to the rack.
I was poking around at the LSC rack to try and set up a temporary arrangement whereby I take the signals from the DAC differentially and route them to the D990694 differentially. The situation is complicated by the fact that, afaik, we don't have any break out boards for the DIN96 connectors on the back of all our Eurocrate cards (or indeed for many of the other funky connecters we have like IDE/IDC 10,50 etc etc). I've asked Steve to look into ordering a few of these. So I tried to put together a hacky solution with an expansion card and an IDC64 connector. I must have accidentally shorted a pair of DAC pins or something, because all models on the c1lsc FE crashed. On attempting to restart them (c1lsc was still ssh-able), the usual issue of all vertex FEs crashing happened. It required several iterations of me walking into the lab to hard-reboot FEs, but everything is back green now, and I see the AS beam on the camera so the input pointing of the TTs is roughly back where it was. Y arm TEM00 flashes are also seen. I'm not going to re-align the IFO tonight. Maybe I'll stick to using a function generator for the THD tests, probably routing non AI-ed signals directly is as bad as any timing asynchronicity between funcGen and DAQ system...
I decided to try doing the THD measurement with a function generator. Did some quick trials tonight to verify that the measurement plan works. Note that for the test, I turned off the z=15,p=150 whitening filter - I'm driving a signal at ~100Hz and should have plenty of oomph to be seen above ADC noise.
I'm going to work on putting together some code that gives me a quick readback on the measured THD, and then do the test for real with different amplitude input signal and whitening gain settings.
**Matlab has a thd function, but to the best of my googling, can't find a scipy.signal analog.
To remind myself of the problem, summarize some of the discussion Koji and I had on the actual problem via email, and in case I've totally misunderstood the problem:
So my question is - should we just cut the PCB trace and add this series resistance for the 4 ALS signal channels, and THEN measure the THD? Since the DC voltage level of the ALS signal is expected to be of the order of a few volts, we know we are going to be in the problematic regime where #11 and #12 become issues.
Correcting a mistake in my earlier elog: the D990694 is NOT differential receiving, it is single ended receiving via the front panel SMA connectors. The aLIGO version of the whitening board, D1001530 has an additional differential-to-single-ended input stage, though it uses the LT1125 to implement this stage. So the possibility of ground loops on all channels using this board will exist even after the planned change to install series resistance to avoid current overloading the preceeding stage.
After labeling all cables, I pulled out one of the D990694s in the LSC rack (the one used for the ALS X signals, it is Rev-B1, S/N 118 according to the sticker on it).
Took some photos before cutting anything. Attachments #1-3 are my cutting plans (shown for 1 channel, plan is to do it for both ALS channels coming into this board). #1 & #2 are meant to show the physical locations of the cuts, and #3 is the corresponding location on the schematic. These are the most convenient locations I could identify on the board for this operation.
I don't know what the purpose of resistors R196, R197, R198 are. I'm assuming it has something to do with the way the ADG333ABR switches. The aLIGO board uses a different switch (MAX4659EUA+), and doesn't have an analogous resistor (though from what I can tell, it too is a CMOS SPDT switch just like the ADG333ABR, just has a lower ON resistance of 25ohm vs 45ohm for the ADG333ABR).
As for the actual resistance to be used: Let's say we don't have signals > 5V coming into this board. Then using 301ohms (as in the aLIGO boards) in series means the peak current draw will be <20mA, which sounds like a reasonable number to me. Larger series resistance is better, but I guess then the contribution of the current noise of the OpAmp keeps increasing.
This is proving much more challenging than I thought - while Cut #1 was easy to identify and execute, my initial plan for Cut #2 seems to not have isolated the input of the second opamp (as judged by DMM continuity). Koji pointed out that this is actually not a robust test, as the switches are in an undefined state while I am doing these tests with the board unpowered. It seems rather complicated to do a test with the board powered out here in the office area though - and I'd rather not desolder the 16 and 20 pin ICs to get a better look at the tracks. This PCB seems to be multilayered, and I don't have a good idea for what the hidden tracks may be. Does anyone know of a secret place where there is a schematic for the PCB layout of this board? The DCC page only has the electrical schematic drawings, and I can't find anything useful on the elog/wiki/old ilog on a keyword search for this DCC document number. The track layout also is not identical for all channels. So I'm holding off on exploratory cuts.
*I've asked Ben Abbott/Mike Pedraza about this and they are having a look in Dale Ouimette's old drives to see if they can dig up the Altium/Protel files.
I quickly put together some code that calculates the THD from CDS data and generates a plot (see e.g. Attachment #1).
I conducted a trial on the Y arm ALS channel whitening board (while the X arm counterpart is still undergoing surgery). With the whitening gain set to 0dB, and a 1Vpp input signal (so nothing should be saturated), I measure a THD of ~0.08% according to the above formula. Seems rather high - the LT1125 datasheet tells us to expect <0.001% THD+N at ~100Hz for a closed loop gain of ~10. I can only assume that the digitization process somehow introduces more THD? Of course the FoM we care about is what happens to this number as we increase the gain.
The main motivation for this work is that I want +15VDC power available on the PSL table to hookup the Teledyne box that Koji made a week ago and do some noise measurements on my revised IR ALS signal chain. But I think this is a good opportunity to effect a number of changes I've been wanting to do for a while.
Tomorrow, Steve and I will do the following:
So in summary, we will need, at 1X1, (at least, including 1 spare for future work):
We completed this work today. Need to clean up a little (i.e. coil excess cable lengths, remove unused cables etc), which we will do tomorrow. All connections have been made at the DIN rail end, but the fuses have not been inserted yet, so there is no voltage reaching the PSL table on any of the newly laid out cables. We also need to establish two +15VDC connections at the DIN rail side. I may establish this later in the evening, as the main point of this work was to get the Teledyne signal path operational. Setting up these DIN connectors is actually a huge pain, we tried to setup a few extra ports for the voltages we used today so that in future, life is easier for whoever wants to pipe DC power to the PSL table. The rule is, however, to re-establish the same number of open ports for each voltage as was available when you started.
For the ZHL-3A, Teledyne, and AOM driver cables, we used 18AWG, 2 conductor, twisted wire, while for the PSL fan we used 20AWG. For the FSS box, we decided to use the 3 conductor 24AWG twisted wire. I believe that these wire gauge choices are appropriate given the expected current in each of these paths.
Pictures + further details tomorrow.
gautam @ 1030pm: there was some mistake with the +15V wiring we did in the evening (the PSL fan and Teledyne cables were plugged into the wrong DIN terminal blocks). I fixed this, and also routed +15VDC to the newly installed set of terminal blocks for this purpose (since we had run out of +15VDC ports at 1X1). After checking voltages at both 1X1 and on the PSL table, I hooked up
to their newly laid out power supplies. IMC locks so looks like the FSS box is doing fine . So we can recover one bench power supply from under the PSL table on the east side. I didn't hook up the AOM driver just now because of some accessibility issues, and I'd also like to do an ALS beat spectrum measurement if possible.
I have been puzzled as to why the duty cycle of the EX green locks are much less than that of the EY NPRO. If anything, the PDH loop has higher bandwidth and comparable stability margins at the X end than at the Y end. I hypothesize that this is because the EX laser (Innolight 1W Mephisto) has actuation PZT coefficient 1MHz/V, while the EY laser (Lightwave 125/126) has 5MHz/V. I figure the EX laser is sometimes just not able to keep up with the DC Xarm cavity length drift. To test this hypothesis, I disabled the LSC locking for the Xarm, and enabled the SLOW (temperature of NPRO crystal) control on the EX laser. The logic is that this provides relief for the PZT path and prevents the PDH servo from saturating and losing lock. Already, the green lock has held longer than at any point tonight (>60mins). I'm going to leave it in this state overnight and see how long the lock holds. The slow servo path has a limiter set to 100 counts so should be fine to leave it on. The next test will be to repeat this test with LSC mode ON, as I guess this will enhance the DC arm cavity length drift (it will be forced to follow MCL).
Why do I care about this at all? If at some point we want to do arm feedforward, I thought the green PDH error signal is a great target signal for the Wiener filter calculations. So I'd like to keep the green locked to the arm for extended periods of time. Arm feedforward should help in lock acquisiton if we have reduced actuation range due to increased series resistances in the coil drivers.
As an aside - I noticed that the SLOW path has no digital low pass filter - I think I remember someone saying that since the NPRO controller itself has an in-built low pass filter, a digital one isn't necessary. But as this elog points out, the situation may not be so straightforward. For now, I just put in some arbitrary low pass filter with corner at 5Hz. Seems like a nice simple problem for optimal loop shaping...
gautam noon CNY2018: Looks like the green has been stably locked for over 8 hours (see Attachment #1), and the slow servo doesn't look to have railed. Note that 100 cts ~=30mV. For an actuation coefficient of 1GHz/V, this is ~30MHz, which is well above the PZT range of 10V-->10MHz (whereas the EY laser, by virtue of its higher actuation coefficient, has 5 times this range, i.e. 50MHz). Supports my hypothesis.
Having implemented the changes to the audio amplifier stage, I re-installed this unit at the LSC rack, and did some testing. The motivation was to determine the shape of the ALS error signal spectrum, so that I can design a whitening preamp accordingly. Attachment #1 is the measurement I've been after. The measurement was taken with EX NPRO PDH locked to the arm via green, and Xarm locked to MC via POX. Slow temperature relief servo for EX NPRO was ON. Here are the details:
Conclusion: In the current configuration, with x10 gain on the demodulated signals, we barely have SNR of 10 at ~500Hz. I think the generic whitening scheme of 2 zeros @15Hz, 2poles@150Hz will work just fine. The point is to integrate this whitening with the preamp stage, so we can just go straight into an AA board and then the ADC (sending this signal into D990694 and doing the whitening there won't help with the SNR). Next task is to construct a test daughter board that can do this...
c1mcs had died for some reason. Looking at dmesg, I see:
None of the other EPICS processes died. Not sure what to make of this. I was at the PSL table working, and had closed the PSL shutter to avoid MC autolocker trying to keep the MC locked while I was mucking about, but this shouldn't have had any effect on an EPICS process?
Anyway, I just logged into c1sus, stopped and restarted the model. IMC locks fine now.
After discussing with Koji, I decided to try and align the input beam polarization at the PSL fiber coupler to one of the special axes of the PM fiber. The motivation is to try and narrow down the source of the large RF beatnote amplitude drift I noticed and reported last night.
The setup for doing so is shown in Attachment #1 - essentially, I setup one of the newly purchased couplers in a mount, set up a PBS, and placed two photodiodes at the S and P ports of the PBS. The idea is to rotate the input coupler in its mount, thereby maximizing the PER (monitored on two Thorlabs PDA520s - I didn't check the gain balance of them).
I spent ~30mins doing some preliminary trials just now, and, I was able to achieve a PER of ~1/20. But I think much better numbers were reported in this SURF project (although I'm not entirely sure I understand that measurement). I will spend a little more time tweaking the alignment. The procedure is tricky as at some point, simply rotating the mount reduces the mode-matching efficiency into the fiber so much that it is not possible to get a meaningful PER measurement from the photodiodes. I'm adjouring for now, more to follow...
Current configuration of PSL free-space to fiber coupling is:
I had noticed that the RF beat amplitude was fluctuating by up to 20dBm as viewed on the control room analyzer. As detailed in my earlier elog, I suspected this to be because of random polarization drift between the PSL and EX fields incident on the Fiber coupled PDs. Since I am confident the problem is optical (as opposed to something funny in the electronics), we'd like to be able to isolate which of the many fiber segments is dominating the contribution to this random polarization drift.
Some useful references:
Procedure and details:
I wanted to lock the single arm POX/POY config to do some tests on the BeatMouth. But I was unable to.
Not sure what to make of all this, but I can lock the arms now.
Attachment #1 shows the ALS noise measurement today. Main differences from the spectrum posted last week is that
For comparison, I have plotted alongside today's measurement (left column) the measurement from last week (right column).
I made a voltage divider using a 20.47kohm and 1.07kohm (both values measured with a DMM). The whole thing is packaged inside a Pomona box I found lying around on the Electronics bench. I have hooked it up to the ALSY_I channel and will leave it so overnight. The INMON of this channel isn't DQed, but for this test, the 16Hz EPICS data will suffice. I've locked the EX laser to the arm, enabled slow temperature servo to allow overnight lock (hopefully) and disabled LSC mode (as locking the arm to the MC tends to break the green lock)
To convert the INMON counts to RF power, I will use (based on my earlier calibration of this monitor channel, see DCC document for the demod chassis).
1AM update: Attachment #1 shows that the RF amplitude has been relatively stable (less than 10% of nominal value variation) over the course of the last hour or so. Even though there is some low frequency drift over timescales of ~20mins, no evidence of the wild ~20dB amplitude changes I saw last week. The signs are encouraging...
overnight update: See Attachment #2 - looking at the past 11 hours of second trend data during which the arm stayed locked, there actually seems to have been more significant variation in the beatnote amplitude. Swings of up to 6dBm are seen on a ~20min timescale, while there is also some longer term drift over 12 hours by a couple of dBm. There is probably a systematic error in the Y-axis, as I measured the RF power at the input of the power splitter at the LSC rack to be ~3dBm, so I expect something closer to 0dBm to be the LO input power which is what I am monitoring. So further debugging is required - I think I'll start by aligning the X fiber coupled beam to one of the fiber's special axes.
To make this setup more permanent, I modified the c1lsc model to pipe the LO power monitor signals from the Demod chassis to unused channels ADC_0_25 (X channel LO) and ADC_0_26 (Y channel LO) in the c1lsc model. I also added a couple of CDS filter blocks inside the "ALS" namespace block in c1lsc so as to allow for calibration from counts to dBm. I didn't add any DQ channels for now as I think the slow EPICS records will be sufficient for diagnostics. It is kind of overkill to use the fast channels for DC voltage monitoring, but until we have acromag channels readily accessible at 1Y2, this will do.
Modified model compiled and installed successfully, though I have yet to restart it given that I'll likely have to do a major reboot of all vertex FEs
I thought a little bit about the design of the preamp we want for the demodulated ALS signals today. The requirements are:
Attachment #3 shows a design I think will work (for now it's a whiteboard sketch, I''ll make this a computer graphic tomorrow). I have basically retained the differential sending and receiving capabilities of the existing Audio I/F amplifier, but have incorporated some whitening gain with a pole at ~150Hz and zero at ~15Hz. I've preserved the DC gain of 10, which seems to have worked well in my tests in the last week or so. Attachments #1 and #2 show the liso modelled characteristics. Liso does not support input-referred noise measurements for differential voltage inputs, so I had to calculate that curve manually - I suspect there is some subtlety I am missing, as if I plot the input referred noise out to higher frequencies, it blows up quite dramatically.
Next step is to actually make a prototype of this. I am wondering if we need a second stage of whitening, as in the current config, we only get 20dB gain at 150Hz relative to DC. Yesterday's beat spectrum measurement shows that we can expect the frequency noise of the ALS signal at ~100Hz to be at the level of ~1uV/rtHz, but this is is around the ADC noise level? If so, 20dB of whitening gain may be sufficient?
Still have to make preamp prototype daughter board with the right whitening shape... This test suggests to me that I should also make the output differential sending...
*Side note: I was wondering why we need the differential receiving stage, followed by a difference amplifier, and then a differential sending stage. After discussing with Koji, we think this is to suppress any common-mode noise from the mixer outputs.
Using one of the prototype PCB boards given to me by Johannes, I put together v1 of this board and tested it.
Attachment #1 - Schematic with stages grouped by function and labelled.
Attachment #2 - Measured vs modelled Transfer function.
Attachment #3 - Measured vs modelled noise. Measurement shown only between positive output and ground, the other port is basically the same. I will update this attachment to reflect the expected signal level in comparison to the noise, but suffice it to say that given the measured input referred noise, we will have plenty of SNR between 0.1Hz and 10kHz. The single stage of whitening should also be sufficient to amplify the signal above ADC noise in the same frequency band
Attachment #4 - Positive output as viewed on a fast (300 MHz) scope using a Tektronix x1 voltage probe.
Attachment #5 - Daughter board noise with measured ALS noise overlaid (the gain of x10 on the existing audio pre-amp has been divided out).
Given the overall good agreement between model and measurement, I am going to test this with the actual RF beat. For this test, we will need a differential receiving AA board to interface the output of the daughter board with the ADC input.
Next step is to actually make a prototype of this.