Summary:

I do not have an answer to the question "What is an appropriate gain for the IF amplifier stage in the D0902745 FET demod boards?", because of the following problems.

Deatils:

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.

Summary:

A reasonable level of RF beatnote power for operating within the specs of the demod board is 17dBm arriving at the input to the power splitter just before the delay line.

Details:

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:

1. The overall conversion factor from RF to IF is ~2.3V IF per volt of RF.
   • 50ohm source connected to RF input of demod board, level = 10dBm on Marconi screen, consistent with inferred value from RF mon output.
   • LO driven at 14dBm by Fluke function generator.
   • The ratio was calculated for IF voltage input into a High-Z load.
   • So let's say we want to run at half the ADC full range of 10Vpp into the whitening board - this means we need to keep the RF input to <=11dBm. 
2. The Teledyne amplifier has a rated maximum input voltage of 17dBm. If we want to stay 3dB below this, we can send in 14dBm into the LO input of the demod board, which is what my characterizations were done with.

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.

After taking the measurements, calibrating them (approximately), and filterting them, I created the following plot. The exponential fit is quite good, as the error is not more than 0.03 C. I used the python function curve_fit in order to get this, and it gave me the time constant as well, which came out to 0.357 hr. From my previous calculations here, I plugged in the values we have (m = 12.2 kg, c = 500 J/kg*k, d = 0.0762 m, k = 0.26 W/(m^2*K), A = 1 m^2), and got that

This is a bit off, but it's probably due to the parameters not being exactly what I supposed them to be, and heat losses through the bottom of the can.

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:

• Check +/-15V supplies, power reg board.
• Check DC offset on I and Q front panel output with LO driven at +10dBm, RF input terminated. Found it to be 0.
• Checked calibration of back-panel DSUB connector monitors for LO and RF powers. Data to be uploaded, looked quite linear.
• Checked conversion gain from RF input to IF output for two sets of LO/RF powers.
• Measured conversion gain as a function of the IF frequency (i.e. frequency offset between LO and RF inputs, out to ~700kHz, 8 datapoints)
• Checked orthogonality and gain balance of the I and Q outputs.
• Measured the noise of the I and Q outputs in the audio frequency range using the SR785.

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:

• Whitening (z,p) at (4Hz,40Hz) or (15Hz,150Hz) so that we have frequency dependent gain that can handle the large DC signal level expected. Need to measure noise of the actual IR beat signal to determine what the appropriate whitening shape is.
• Low-pass above a few 100kHz to cut out 2f modulation product
• Low-passing at input of AD829 (or just use OP27?)

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. 

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.

Tip-Tilt Suspension CAD:
Discussed with Koji about motivation to simplify the design of this assembly, which has many unnecessary over-constraints. I have started to cad alternate parts with the aim of removing these over-constraints.

40m Lab CAD:
Acquired a stack of original engineering drawings of the vacuum chambers from Steve which I will take home, get scanned, and then use as reference for the cad i'm working on.

Other:
Started paperwork at west bridge office to get paid as an "occasional employee". Hopefully I receive old money.

I've packaged an AP1053 in a Thorlabs box. The gain and the input noise level were measured. It has the gain of ~10 and the input noise of ~0.6nV/rtHz@50MHz~200MHz.

Details

AP1053 was soldered on Thorlabs' PCB EEAPB1 (forgot to take a picture). The corresponding chassis is Thorlabs' EEA17. There is a 0.1uF high-K ceramic cap between DC and GND pins. The power is supplied via a DC feedthru capacitor (Newark / Power Line Filter / 90F2268 / 5500pF) found in the WB EE shop. The power cable has a connector to make the long side of the wires detachable. Because I did not want to leave the RF signal path just mechanically touched, the SMA connectors were soldered to the PCB. As the housing has no access hole, I had to make it at one of the sides.

The gain of the unit was measured using the setup shown in the upper figure of Attachment 2. When the unit was energized, it drew the current of about 0.1A. The measued gain was compensated by the pick off ratio of the coupler (20dB). The gain was measured with the input power of -20, -10, 0, 10, and 15dBm. The measurement  result is shown in Attachment 3. The small signal gain was actually 10dB and showed slight degradation above 100MHz. At the input of 10dB some compression of the gain is already visible. It looks consistent with the specification of +26.0dBm output for 1dB compression above 50MHz and +24.0dBm output below 50MHz.

The noise level was characterized with the setup shown in the bottom figure of Attachment 3. The noise figure of the amplifier is supposed to be 1.5dB above 200MHz and 3.5dB below 200MHz. This is quite low and the output noise of AP1053 can not be measured directly by the analyzer. So, another LN amplifier (ZFL-500HLN) was stacked. The total gain of the system was measured in the same way as above. The measured noise level was ~0.7nV/rtHz between 50MHz and 200MHz. Considering the measurement noise level of the system, it is consistent with the input referred noise of 0.6nV/rtHz. I could not confirm the advertized noise figure of 1.5dB above 200MHz. The noise goes up below 50MHz. But still 2nV/rtHz at 3MHz. I'd say this is a very good performance.

I could not understand why 'netgpibdata' scripts are missing in "scripts/general" folder on pianosa... Where did they go???

Also, I found that the PROLOGIX GPIB-LAN controller for crocetta (192.168.113.108) is no longer working. I need to reconfigure it with "telnet"...

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.

Attached the program I used to create the plot

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:

• The problem is reminiscent of the one I experienced with the SR785 not too long ago - there the culprit was a switching power supply used for the Prologix GPIB-ethernet box.
• Then I remembered sometime ago rana and i had identified the power supply for the Fibox at the LSC rack as a potential pollutant. But today, I confirmed that this power supply is not to blame, as I unplugged it from its powerstrip and the spectrum didn't change.
• There are a couple of Sorensens in this LSC rack, from what it looks like, they supply power to a BIO interface box in the LSC rack. I thought we would want to keep this rack free of switching power supplies? Wasn't that the motivation behind keeping the (linear) power supplies for all the LSC rack electronics in a little separate rack along the east arm?
• I confirmed that when the D990694 input is terminated, these harmonics are no longer present. 
• I plugged the output of the SOS dewhite board to an oscilloscope - there is a ~20mVpp signal there even when the DAC output is set to 0, but this level seems too small to explain the ~1000 ct-pp signal that I was seeing. The whitening gains for these channels are set to 0dB.
• I also looked at the signal in the time domain using DTT - indeed the peak-to-peak signal is a few thousand counts.
• This isn't a problem with the particular input channel either - the behaviour can be reproduced with any of the 4 ALS input channels.

Am I missing something obvious here? I think it is impossible to do a THD measurement with the spectrum in this condition...

crochetta was reconfigured to have 192.168.113.108. It was confirmed that it can be used with netgpibdata.py

Configuration: I have connected my mac with the unit using an Apple USB-Ethernet adapter. The adapter was configured to have a manual IP of 192.168.113.222/255.255.255.0. "netfinder.exe" was run to assign the IP addr to the unit. It seemed that NVRAM of the unit evaporated as it had the IP of 0.0.0.0. Once it was configrued, it could be run with netgpibdata as usual.

Did some quick additional checks to figure out what's going on here.

1. The SOS/dewhite board for which I didn't have a DCC number is D000316. It has a single ended output.
2. I confirmed that the origin of this noise has to do with the ground of the aforementioned D000316 - as mentioned in my previous elog, having one end of a BNC cable plugged into the whitening board D990694 and the other end terminated in 50ohms yields a clean spectrum. But making the ground of this terminator touch the ground of the SMA connector on the D000316 makes the harmonics show up.
3. Confirmed that the problem exists when using either the SMA or the LEMO monitor output of these D000316 boards.

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.

• I pulled the board out and looked at the output of Ch8 on the oscilloscope with the board powered by a bench power supply - signal looked clean, no evidence of the noisy ~20mVpp signal I mentioned in my previous elog.
• Put the board back into a different slot in the eurocrate chassis and looked at the signal from Ch8  - looked clean, so the ground of the eurocrate box itself isn't to blame.
• Put the board back in its original slot and looked at the signal from Ch8 - the same noisy signal of ~20mVpp I saw yesterday was evident again .
• Disconnected the backplane connector which routes signals from the DAC adaptor box to D000316 board - noisy signal vanished .

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?

I decided to plot the temperatures measured over two days for the sensor inside the can and inside the lab just to see if there was any significant difference between the two, and obtained the following plot. This shows that there is a difference in measurements of a few 0.01 C. The insulated seismometer can didn't change temperature as much as the lab did, which is as expected. I'll work on properly calibrating the sensors sometime in the future so that we can use the sensor that's just in the lab as an accurate thermometer.

[koji, gautam]

We discussed possible solutions to this ground loop problem. Here's what we came up with:

1. Option #1 - Configure the DAC card to receive a ground voltage reference from the same source as that which defines the LSC rack ground.
2. Option #2 - construct an adapter that is differential-to-single ended receiving converter, which we can then tack on to these boards.
3. Option #3 - use the D000186-revD board as the receiver for the DAC signals - this looks to have differential receiving of the DAC signals (see secret schematic).  We might want to modify the notches on these given the change in digital clock frequency 

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.

I subtracted out the lab temperature change during the period of cooling to see if it would have a significant effect on the time constant, but when I fit the new data, the time constant came out to 0.355 hr, which is not a significant change from the value of 0.357 that I got earlier.

Hornet cold cathode gauge analoge output  [ DSub9 pin 3 and 7 ] are wired to go ETMX Acromag. It was reading 4.9V at 7.8e-6 Torr [ 3,110 V  8.35e-5A ] at the end of a 24ft BNC cable. Now it has to be hook up to an Acromag channel.

This will replace the not functioning C1: Vac-CC1_pressure

gautam: the motivation behind hooking this gauge up to our DAQ system is that non-vacuum-system-experts have a quick diagnostic to make sure everything is in order. This gauge is physically placed adjacent to V1, and so if something goes wrong with our vacuum pumps, we would see the effect here immediately. we did note that occassionally, the reading fluctuated by ~1V on the DMM used to check the voltage output at the end of the BNC cable, so we still need to run some long-term stability analysis once this channel is hooked up to the Acromag.  For future reference, in order to make this gauge work, we need to check that

1. Error flag has been cleared.
2. HV is ON (state has to be manually toggled).

Summary of my tests of the demod boards, post gain modification:

• DC tests (supply voltage, DC offsets at I and Q outputs, power LEDs etc)
• RF tests
  • Back panel RF and LO power level monitor calibration
  • Coupling factor from RFinput to RFmon channel
  • Conversion loss as a function of demodulated beat frequency
  • Orthogonality and gain balance test
  • Linearity of unit as a function of RF input level
  • Electronics oise in the 1-10kHz band at the IF outputs.

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...

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.

The ETMX  Sorrenson power supply -15V was running at -13.9V

[kira, gautam]

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

• +/- 15V for the temperature sensors
• +/- 20V, 5A for heater circuit (to be confirmed by Kira after looking at voltage regulator datasheet for dropout voltage)
• 2 ADC channels for temperature sensing (one inside can and one outside)
• 1 DAC channel for controlling MOSFET

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?

If this looks okay (after pics are uploaded), we will implement these changes (hardware + software) tomorrow.

-----

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.

1. The heater circuit should get its own dedicated supply.
2. We do not want to ever use the old Ref Cav heater for anything. Its unreliable and noisy.
3. Steve should update all the sticky labels on all the power supplies in the lab to indicate what voltage they should be set at. Even if the label is correct, the date should be updated.

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.

1. Checked for ground loops - seem to be fine, see black trace on Attachment #1 which was taken with the FnGen hooked up to the input, but not putting out any signal
2. Spectrum with 1Vpp sine wave @ ~103Hz. The various harmonic peaks are visible, and though I've not paid attention to bin width etc, the largest harmonics are ~1000x smaller than the main peak, and so the THD is ~1ppm, which is in the ballpark of what the datasheet tells us to expect around 100Hz for a gain of ~10 (=20dB). The actual gain was set at 0dB (so all opAmps in the quad bypassed)

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:

1. The "Variable Gain" feature on the D990694 boards is achieved by 4 single gain stages cascaded together in series, with the ability to engage/bypass each stage individually.
2. The 4 gain stages are constructed using the 4 OpAmps in a quad LT1125 IC, each in standard non-inverting configuration.
3. The switches unfortunately are on the output side of each op-amp. This means that even if a stage is bypassed, the signal reaches the input pin of the OpAmp.
4. For proper operation, in closed-loop, the differential voltage between the input pins of the OpAmp are 0.
5. But this may require the OpAmp to source more current than it can (just using Ohms law and the values of the resistors in the feedback path).
6. As a result, a large differential voltage develops between the input pins of the OpAmp.
7. The LT1125 is not rated to operate in such conditions (this is what Hartmut was saying in the ilog linked earlier in this thread).
8. Part of the internal protection mechanism to prevent damage to the IC in such operating conditions is a pair of diodes between the input pins of the OpAmp.
9. When a large differential voltage develops between the input pins of the OpAmp, the diodes act to short the two to bring them to the same potential (minus whatever small drop there is across the diodes). Actually, according to the datasheet, when the differential voltage between the input pins exceeds 1.4V, the input current must be limited to 25mA, to avoid damaging the protection diodes? If so, we may already have damaged a bunch of these amplifiers.
10. While the LT1125 IC is protected in this condition, the infinite input impedance of the OpAmp is reduced to the resistance between the inverting input and ground. The output voltage may still be saturated, but the output current draw is within what the IC can supply.
11. As a result, Ohms law means that the preceeding stage is overdrawn for current. This is clearly not ideal.
12. Another possible problem is that there is some sort of interaction between the 4 opamps in the quad IC, which means that even if one stage is overdrawn for current, all of them may be affected.
13. The Advanced LIGO version of this board addresses #11 and #12 by (i) placing a series resistor between the input signal and the non-inverting input of the opamp, and (ii) using single opamp ICs instead of a quad, respectively.

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.

> 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?

GO A HEAD

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.

Some points before we can set up the can:

1. Cable length and type
   • For the DAC, we can use the LEMO outputs and change it to BNC, then have a long BNC cable running over the top of the lab and to the can 
   • ADC is also LEMO, which we can convert to BNC and have a cable run from that to the temperature sensors
   • Sorensens use plain cables, so we need to find ones that can take a few amps of current and have them be long enough to reach the can and temperature sensors
2. Making sure that there is enough space for the can
   • Can measures about 59 cm in diameter, which does fit in the space we chose
3. Finding Sorensens that work and can provide +/-24V to the heater circuit (since Rana said we want the heater to have its own supply)
   • Found two Sorensens, but only one works for our purpose (update: found a second one that works)
   • The other can only proviide up to 20V before shorting and has been labeled
   • Grounding (see point 5) - we want to have these power supplies be independent, but we must still specify a ground
   • There is exactly enough space to fit in the two Sorensens below the ones that are currently there
4. DIN fuses for 15V and 24V
   • 15V fuses can be easily installed since we don't need a very high current for the temperature sensors
   • the 24V fuses seem to be able to handle 6.3A according to the datasheet, but it only says 4V on the fuse itself. Not sure if this is the wrong darasheet...
5. Connecting the crcuit to the DAC and what connectors to use
   • Using the rightmost DAC because there's less important things connected to it, and use the LEMO conncectors to provide the input
   • Connect the grounds of the DAC and the new Sorensens that we're going to install to the grounds of the rest of the Sorensens
   • *confirm that this setup will work and if not find an alternative
6. Which channels to use for the ADC
   • channels 29, 30, 31 are available, so we can use any two of those (one for each sensor)

Also, I need to eventually remake the connections on my circuit board because they are all currently test points. I also need to find a box for the heater circuit and figure out what to do with the MOSFET and heat sink for it. This can either be done before setting everything up, or we can just change it later once we have the final setup for the can ready.

If all of this looks good then we can begin the setup.

gautam:

1. I recommend using a DAC output from the rightmost AI board because (i) only the green steering mirror PZTs are hooked up to it while the other has ETMX suspension channels and (ii) the rightmost AI board has differential receiving from the DAC, and in light of the recent discussions about ground loops, this seems to be the way to go. Outputs 5-8 are currently unused, while outputs 1-4 are used for the EX green input steering mirror control.
2. Converters required:
   • 2 pin LEMO to BNC --- 2pcs for each temp sensor.
   • Single pin LEMO to BNC --- 1pc for AI board to heater circuit input (readily available)

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.

[Kira, Steve]

We set up a new rail for the Sorensens (attachment 1) and placed one of them down on this new rail (attachment 2). Unfortunately the older rail that had been used to support the other Sorensens (the top one in attachment 1) is thick and does not allow another one of the Sorensens to slide in between the current ones. So we will have to support all the ones on top with a temporary support, take out the old rail, and then insert the new ones before letting the new bottom rail carry the weight of all of the Sorensens. We will do that tomorrow.

In addition, we have to figure out how to lead all the cables to the can, but there are no holders on the side of the lab to do so. So, we decided that we would have a new one installed on the side shown in attachment 3 so that we wouldn't have to place the wires along the floor.

Also, there has been some space made for the can along with the new insulation. The stuff mounted on the wall was removed and will be reattached tomorrow so that it doesn't get in the way of the can anymore.

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).

Algorithm is:

1. Get data (for now, offile file, but can be readily adapted to download data live or from NDS).
2. Compute power spectrum using scipy.signal.periodogram. I use a Kaiser window with beta=38 based on some cursory googling, and do 10 averages (i.e. nfft is total length / 10), and set the scaling to "spectrum" so as to directly get a power spectrum as opposed to a spectral density.
3. Find fundamental (assumed highest peak) and N harmonics using scipy.signal.find_peaks_cwt. I downsample 16k data to 2k for speed. A 120second time series takes ~5 seconds.
4. Compute THD as where V_i denotes an rms voltage, or in the case of a power spectrum, just the y-axis value.

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.

[Kira, Steve]

We installed and labeled the Sorensens today.

Our 3 cranes passed  professional  inspection. Fred Goodbar of Konacrane with 450 lbs load at full extension.

Certificates will be posted in 40m wiki as they arrive.

Gautam and Steve,

The "called 225 lbs" steel crane load measured right on 102 kg

The trick to the measurment to maintain 1 mm gap to the central cilynder of the load cell.

The lead plate stabilized the large load.

gautam: some additional notes:

1. the wiring on the Omega controller unit as given to us was wrong - I had to fix this on the D-sub connector in order to get the load cell to work. something to check for the other units.
2. the main difficulty in doing this calibration run was that the readback is very sensitive to tilts of the load relative to the sensor.
3. the problem is complicated by the fact that the load cell itself does not have a flat surface - it has a ring that protrudes above the flat face of the cylindrical load cell by a few mm as Steve mentioned.
4. so in order to measure the weight of our stacks, we have to mitigate this problem and ensure that the full load of the stack is normally incident on the load cell - if the load cell itself is somehow torqued during the measurement because of the distribution of the load on it being uneven, we get an inaccurate measurement.
5. In this calibration measurement, we think the error is <1% (true mass is 102kg, we measure 104kg on the meter which seems reasonable as the sum of the donut + lead plate)

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:

1. Fix the AOM driver power cabling that I broke.
2. Make the AOM +24VDC power supply independent - right now it is shared between the AOM driver and the two ZHL-3-A amplifiers.
3. Tap an independent +24VDC power supply from 1X1 for the ZHL-3A amplifiers (I guess one power supply and fuse is sufficient for both amplifiers since they are in the same box).
4. Tap an independent +15VDC power supply for the Teledyne box.
5. Tap an independent +15VDC power supply for the little fan on the back of the PSL controller, that is currently powered by a bench supply (+12VDC, but it's just a fan, so +15VDC or +10VDC will do just fine, and these are the Sorensen levels we have).
6. Tap an independent +/-24VDC power supply for the FSS summing box. Right now it is being powered by a bench supply under the PSL table. The indicated supply voltage on the box is +/-18V. But according to the schematic, this +/-18V get regulated down to +/- 15V, so we may as well use +/-24V which is available from the Sorensens in 1X1 (there is no +/-18VDC Sorensen there). The datasheets for the 7815 and 7915 ICs suggest that this will be just fine.
7. Where possible, make at least 1 spare outlet for each supply voltage available at 1X1, such that in future, tapping extra supply points won't be such a huge pain.

So in summary, we will need, at 1X1, (at least, including 1 spare for future work):

• New +24VDC connections ------ 3x
• New -24VDC connections ------- 2x
• New +15VDC connections ------ 3x

[steve, gautam]

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

1. FSS Summing box
2. Teledyne amplifier
3. ZHL-3A amplifiers

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 checked channels 6 and 7 on the ADC and they have long wires leading to BNC ends and are currently not being used, so we could probably just attach the temperature sensors to those channels.

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:

1. Mode-matching into the BeatMouth PSL light fiber had deteriorated dramatically - it was ~1mW out of 4.4mW. I spent 5 mins getting it back to 3.2mW (72% efficiency) and then moved on... I am a little surprised the drift was so large, but perhaps, it's not surprising given that there has been a lot of work on and around the PSL table in the last couple of weeks. There is a 300mm focusing lens after the last steering mirror so the effect of any alignment drifts should be attenuated, I don't really understand why this happened. Anyways, perhaps a more intelligent telescope design would avoid this sort of problem.
2. I removed the ND filter in the PSL pickoff to BeatMouth path (this was not responsible for the reduced power mentioned in #1). I verified that the total power reaching the photodiode was well below its rated damage threshold of 2mW (right now, there is ~620uW). I will update the BeatMouth schematic accordingly, but I think there will be more changes as we improve mode matching into the fibers at the end.
3. Hooked up the output of the fiber PD to the Teledyne amp, routed the latters output to the LSC rack. Measured RF electrical power at various places. In summary, ~6dBm of beat reaches the splitter at the LSC rack. This is plenty.
4. The main finding tonight was discovered by accident.
   • For the longest time, I was scratching my head over why the beat note amplitude, as monitored on the control room SA (I restored it to the control room from under the ITMX optical table where Koji had temporarily stored it for his tests on the PSL table) was drifting by ~10-15dB!
   • So each time, having convinced myself that the power levels made sense, I would come back to the control room to make a measurement, but then would see the beat signal level fluctuate slowly but with considerable amplitude.
   • The cause - See Attachment #2. There is a length of fiber on the PSL table that is unshielded to the BeatMouth. While plugging in RF cables to the BeatMouth, I found that accidentally brushing the fiber lightly with my arm dramatically changed the beat amplitude as monitored on a scope.
   • For now, I've "strain relieved" this fiber as best as I could, we should really fix this in a better way. This observation leads me to suspect that many of the peaky features seen in Attachment #1 are actually coupling in at this same fiber...
   • The beat note amplitude has been stable since, in the ~90 mins while I've been making plots/elogs.
   • Surely this is a consequence of differential polarization drift between the PSL and EX beams?
5. There are prominent powerline harmonics in these signals - how can we eliminate these? The transmission line from PSL table to LSC rack already has a BALUN at its output to connect the signal to the unbalanced input of the demod board.
6. Not sure what to make of the numerous peaks in the LO driven, RF terminated trace.
7. The location of the lowest point in the bucket also doesn't quite match previous measured out-of-loop ALS noise - we seem to have the lowest frequency noise at 150-200Hz, but in these plots its more like 400Hz.

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...

The forline pressure of TP3 was 399 mTorr

It was replaced this morning at  TP3 controller 134,638hrs with the "failed TP2 station" drypump. The foreline pressure now at TP3 is 100 mTorr at 6 hrs of operation.[ at day 3  63 mT ]

IFO pressure at CC Hornet 7.9e - 6 Torr

Valve configuration: vacuum normal as TP3 is the forepump of the Maglev & the annuloses are not pumped

40m Lab Cad:
Updated the dimensions of and fleshed out the chambers in greater detail, by referring to the engineering drawings that Steve gave to me. I have scanned and uploaded most of these drawings to Dropbox in [40mShare]>[40m_cad_models]>[Vacuum Chamber Drawing Scans]. The excel file "LIGO 40m Parts List" in the [40m Lab CAD] folder also lists the Steve drawings I referenced for dimensions of each part.

Next steps:
1. Finish details of all chambers.
2. Start placing representative blocks on the optical table.

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...

Summary:

Current configuration of PSL free-space to fiber coupling is:

• 3.25 mW / 4.55mW (~71%) coupling efficiency, both numbers measured with Ophir power meter, Filter OFF
•  (I choose to define it in this way as opposed to the reciprocal) of 75 (~19dB). The uncertainty in this number is large (see discussion), but I am confident that we have >10dB, which while isn't as good as can be, is sufficient for the main motivation behind this work.

Motivation:

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:

1. General writeup about how PM fibers work and PER. Gives maximum achievable PERs for a given misalignment of incident beam relative to one of the two birefringent axes.
2. Another similar writeup. This one put me onto the usefulness of the alignment keys on the fibers.
3. Thorlabs PM980 specs - this tells us about the orientation of the two axes for the kind of fibers we use.

Procedure and details:

• The principle of operation behind polarization maintaining (PM) fibers is that intentional birefringence is introduced along two perpendicular axes in the fiber.
• As a result of which light propagates with different phase velocity along these axes.
• For an arbitrary incident field with E-field components along both axes, it is almost impossible to predict the output polarization as we do not know the length of propagation along each axis to sufficient precision (it is also uncontrolled w.r.t. environmental fluctuations). So even if you launch linear polarization into the fiber, it is most likely that the output polarization state will be elliptical.
• But if we align the incident, linear polarization along one of the two axes, then we can accurately predict the polarizaiton at the output, to the extent that the fiber doesn't couple power in the two axes during propagation. I cant find a spec for the isolation between axes for the fiber we use, but the specs I could find for other fiber manufacturers suggest that this number is >30dB, so I think the assumption is a fair one.
• A useful piece of information is that the alignment key on the fibers gives us information about the orientation of the birefringent axes inside the fiber. For the Thorlabs fibers, it seems that the alignment key lines up with the stress-inducing rods inside the fiber (i.e. the slow axis). I confirmed this by looking at the fiber with the fiber scope.
• The PSL pickoff beam I am using for this setup is from the transmission of the PBS after the Faraday. So this field should have relatively pure P-polarization.
• The way I have set up the fiber on the PSL table, the fast axis of the fiber corresponds to P-polarization (i.e. E field oscillates parallel to the plane of the optical table). Actually, it was this alignment that I tweaked in this work.
• Using the information about the alignment key defining polarization axes on the fiber, I also set up the output fiber coupler such that the fast axis lined up as near parallel to the plane of the optical table as possible. In this way, the beam incident on the PBS at the output of my setup should be pure P-polarizaiton if I setup my input alignment into the fiber well.
• I tweaked the rotation of the Fiber mount at the input coupler to maximize the ratio of P_p / P_s, as measured by the pair of PDs at the output. 
• As #1 of my listed references details, you need to align the incident linear polarization to one of the two birefringent axes to closer than 6 degrees to achieve a PER of >20dB. While this sounds like a pretty relaxed requirement, in practise, it is about as good as we can hope to achieve with the mounts we have, as there is no feature that allows us to lock the rotational degree of freedom once we have optimized the alignment. Any kind of makeshift arrangement like taping the rotating part to the mount is also flaky, as during the taping, we may ruin the alignment.
• Attachment #1 shows the result of my alignment optimization - the ratio P_p / P_s is about 75.
• The uncertainty on the above number is large. Possible sources of error:
  • Output coupler is not really aligned such that fast axis corresponds to P-polarizaiton for the output PBS.
  • The two photodiodes' gain balance was not checked.
  • The polarization content of the input beam was not checked.
  • The PBS at the output could be slightly misaligned relative to the S/P polarization directions defined by the tabletop.
  • The PBS extinction ratio was not checked.
• But anyways, this is a definite improvement on the situation before. And despite the large uncertainty, I am confident that P_p / P_s is better than 10dB.
• Moreover, Steve and I installed protective tubing on the lengths of fiber that were unprotected on the PSL table, this should help in reducing stress induced polarization drifts in the fiber, at least in these sections of fiber.
• So I think the next step is to monitor the stability of the RF beatnote amplitude after these improvements. At some point, we need to repeat this procedure for the EX and EY fibers as well.
• If the large drifts are still seen, the only thing we can exclude as a result of this work is the section of fiber from the PSL light coupler to the beat mouth.

Last night I worked at the PSL table for the modulation depth measurement for an aLIGO EOM. Let me know if the IFO behavior is unusual.

What I did was:

• Cranked up the HEPA speed to 100
• Placed an aLIGO EOM in the AUX beat path (south side of the PSL laser). (It is still on the PSL table as of Feb 20, 2018)
• Closed the PSL shutter
• Turned off the main Marconi forr 11MHz. The freq and output power of this marconi have not been touched.
• Turned off the freq generation unit
   
• Worked on my measurement with the spectrum and network analyzers + aux marconi.
   
• Turned down the HEPA speed to 30
• Turned on the freq generation unit
• Turned on the main Marconi
• Opened the PSL shutter => IMC locked

I wanted to lock the single arm POX/POY config to do some tests on the BeatMouth. But I was unable to.

• I tracked the problem down to the fact that the TRX and TRY triggers weren't getting piped correctly to the LSC model
• In fact, all RFM channels from the end machines were showing error rates of 16384/sec (i.e. every sample).
• After watchdogging ETMX, I tried restarting just the c1scx model - this promptly took down the whole c1iscex machine.
• Then I tried the same with c1iscey - this time the models restarted successfully without the c1iscey machine crashing, but the RFM errors persisted for the c1scy channels.
• I walked down to EX and hard rebooted c1iscex.
• c1iscex came back online, and I ssh-ed in and did rtcds start --all.
• This brought all the models back online, and the RFM errors on both c1iscex and c1iscey channels vanished.

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

1. I have tried to align the input polarization axis (p-pol) to the fast axis of the fiber, and believe I have done it to ~75dB.
2. Steve and I installed some protective tubing for the vertical lengths of fiber going into the beat mouth.
3. Today, I decided to measure the noise at the differential rear panel outputs rather than the single-ended front panel outputs. For the test, I used a DB25 breakout board and some pomona mini-grabber to BNC clips to connect to the SR785.

For comparison, I have plotted alongside today's measurement (left column) the measurement from last week (right column).

Conclusions:

• The clear daylight between red and green traces in the left column give me confidence that I am measuring real laser frequency noise in the red trace. It even has the right shape considering the bandwidth of the EX PDH servo.
• The installation of protective tubing doesn't seem to have reduced the heights of any of the peaks in the red traces. I hypothesize that some of these are acoustic coupling to the fiber. But if so, either the way we installed the protective tubing doesn't help a whole lot, or the location of the coupling is elsewhere.
• Judging by the control room analyzer, there doesn't seem to be as large drifts in the RF beat amplitude tonight () as I saw the last couple of times I was testing the BeatMouth®. For a more quantitative study, I'm gonna make a voltage divider so that the ~10V output I get at the rear panel power monitor output (for a LO level of ~0dBm, which is what I have) can be routed to some ADC channel. I'm thinking I'll use the Y ALS channels which are currently open while ALS is under work.
• 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...

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.