Mr Fred Goodbar of Konacrane was in the lab 830am-1130am today. All three cranes in the VEA were inspected, loaded with 450lb test weights, and declared in good working condition and safe to use.

1. Apparently, the clackity noise heard when running the crane at the south end is a known problem - the crane was opened up and inspected sometime in the past, and no obvious cause was found. This is not expected to affect the usability of the crane.
2. The travel speed of the cranes is slow - but this is apparently intentional, on the request of Steve V.

The interferometer subsystems appear normal after the inspection. 

Jenne, Koji

Tonight we found that the wireless for Martian network was down.
We inspected the router and found the power was down. The power of the weather station was also down.

By touching the power outlet which they are connected, the power changes on and off.
This problematic power outlet has a label "L#17" just below the photograph of the mk I (1989).
The plug was connected to the left one.

As it was scary, we moved the power plug to the next one (L#19).
The wireless router and the weather station were powered now,
though the weather station is showing a wrong time in its clock.

Elog crashed a couple times, restarted it a couple times.

I created the sus model, which is the suspension controller for ITMX, ITMY, BS, PRM, SRM.  I also created sup, which is the suspension plant model for those same optics.

Updated /cvs/cds/caltech/target/fb  master and daqdrc files to add SUS, SUP models.  Megatron's /etc/rc.d/rc.local file has been updated to include all the necessary models as well.

The suspension controller needs the Binary IO outputs need to be checked and corrected if wrong by changing the constant connected to the exclusive or gates. Right now its using the end suspension binary output values which may not be correct.

The CRYO pump cooled down and VC1 was opened. This valve configuration is Cryo-only

PSL output shutter opened at 4pm 

PZT HV power supplies turned on for OMC and IOO steering mirrors.

There positions were not corrected to strain gauge values.

Ben helped us to conclude that the FAILURE led indicator is working correctly and

has nothing to do with the one lose, dangling wire#258 in the side connects of the vac rack.

I reset the CSB switch inside the Maglev controller and tried to start accelerating.

It fails after 2-3 sec and failure led light comes on.

Now we can say the MAGLEV 360 is DEAD and the new OSAKA TG420M can be swapped in.

However it requires new interface with our epics based MEDM or better...?

The comparison between the new and old MC servo (FSS part) was attached.

- The new servo has the same DC range as before.
  Even though there is 1/2 gain in the chain now, the previous range of the FSS box was 0 to 10V.
  Now it is +/-10V. So we did not lose the range.

- The new servo has x3.2 larger range above 100Hz.

- x1.6 enhancement of the FSS Box output noise above 10Hz.

- The noise of the HV amp (and the summing amp) is x300 and x2600 more filtered at 10kHz and 100kHz respectively.

Current OSEM sensor values with all the suspensions aligned are attached.
For 'BS','ITMX','ETMX','ITMY','ETMY','PRM','SRM','LO1','LO2', the ones out of the range [200,800] are marked, and for 'PR2','PR3','SR2','AS1','AS4', the ones out of the range [6000,24000] are marked.

Hi everybody,

Last night I (with the help of Jenne and Jenne's advice - not to implicate her in this or anything) changed the filters for GUR1, GUR2, and STS in C1:PEM-RMS, adding a butterworth bandpass filter at each corresponding frequency band as well as a gain to convert from counts to micros/sec, and then adding a low pass filter in case of aliasing upon squaring.

Currently the seismic signals are going crazy, and producing "Nan" output on the strip graph (which leads to the instantaneously sharp spikes - which leads to the entire signal being filled on the visualizer on the wall). I checked the DataViewer output and the tdsdata output using both grep and wc, and it seems that both every single signal point is present and is a real number (also not a small real number, thereby debunking floating-point error). I'm currently not sure why seismic-strip reads out 'Nan' - perhaps because it's taking the log of 0, taking a negative log, taking the root of a negative number, or dividing by zero.

Does anyone know if the seismic-strip Nan issue is a program bug? If it's not (and therefore a filter bug), please let me know as well.

I'll be in lab for the rest of the night changing the butterworth filters to odd-order elliptic filters (at Rana's suggestion), as well as changing the cut-off frequency for the low-pass filters.

I'll E-log about it when I'm done.

Just to be sure that my numbers are correct - The STS, GUR1, and GUR2 channels all have gain 10, right? (I parsed through the e-log, and these seem to be the most recent numbers

Thanks for your help,

Masha

UPDATE: I changed all of the GUR1Z channels to order-5 elliptic filters. I approximated the attenuation for each one by setting the integral from _CutoffFrequency to 10^3 Hz of 10^(-Percent(f)/20) df to 0.01, where Percent(f) is a linear approximation of the relationship between the log of the frequency and the dB level (with the attenuation defining one of the points). Right now the Nan problem continues to persist, even after I loaded the coefficients. In Dataviewer, the channels look relatively normal for the past 10 minutes, as does the data when viewed with tdsdata.

 FIGURED IT OUT - THERE WAS A PROBLEM WITH THE LOW PASS FILTERS (TOO HIGH ORDER). FIXING IT NOW, SHOULD BE GOOD IN AN HOUR. 

For your planning:

The X and Y directions in the STACIS still both oscillate uncontrollably in closed loop, so I'll be doing my testing in Z for now. If I need to use the other axes I'll lower their gain with the pots and add weight to the STACIS platform to try to make it more stable.

Measurements I've taken for Z:

--Open loop gain, taken by driving the PZTs with a swept sine signal and measuring with both internal geophones and external accelerometers. These measurements look a lot like the plots supplied by the STACIS manufacturer, with a resonance at 15-16 Hz (X and Y also look good). Figure below was taken with geophones:

--Open loop gain, where the input is ambient seismic noise measured by one set of accelerometers on the floor and one set on top of the STACIS:

--Closed loop gain, where the input is ambient seismic noise, and feedback is supplied by the geophones (like normal STACIS operation). There's a definite drop in the transfer function, as expected:

--Open and closed loop transfer functions superimposed (the higher one is open):

I am currently working on using the less-noisy accelerometers to provide feedback instead of the geophones. I have found the right point before the extension board to input the accelerometer signal which is NOT the same as the Signal IN/OUT cables- those are at the end of the board, after amplifying and filtering. I want the accelerometer signal to go through the same circuitry as the geophone signal so that the noise of the sensors themselves can be compared.

Problem: Coherence isn't great between the accelerometer sets at low frequencies, which leads to a not very smooth transfer function. I might try using the shaker, because the larger motion may lead to better coherence between the accelerometers on top of the STACIS and at its base.

 We've set up a preliminary test bed for the phase camera. It simply uses a HeNe that is split into two beams. One is frequency shifted by an AOM by -40 MHz - df, where df is some acoustic frequency. The second beam is transmitted through a 40MHz EOM to get sidebands. The two beams are recombined and are, currently, incident on a photodetector, but this can be replaced by the phasecamera.

We turned everything on with df = 1kHz and confirmed that a 1kHz signal is visible on the output from the photodetector (PD). The signal looks to be about 1:300 of the DC level from the PD.

1. In terms of the AOM:

How much beam power is incident on the AOM? How much is the deflection efficiency?
i.e. How much is the power lost by the crystal, deflected in the 1st order, and remaining in the oth order?

I am curious because I assume the AOM (which vender?) is designed for 1064nm and the setup uses 632nm.

2. In terms of the EOM:

How much sidebands do you expect to have?

I assume the EOM is designed for 1064nm, the only difference is the coating at the end. Is this right? 

3. Beating

How much beating strength do you expect?

Is your beating level as expected?

How much is the contrast between the PM sideband and the frequency shifted carrier?
This must include the consideration on the presence of the carrier and the other sidebands.

I attach the transfar function of the current shunt.
There is a little gap at 10 Hz for phase, but it is a ploblem of measurement and not real one.
 

Uploaded on T1000461 too.

I need some more hints to understand the improvement, although its generally good to re-build it considering the sad state of the assembly/installation that you found.

I see that the current design brings the 11 MHz signal to -2 dBm before intering the first ZHL-2+, but since that has a NF of 9 dB, that seems to only degrade the phase noise to -2 - (-174 +9) = -163 dBc. That seems OK since we only need -160 dBc from this system. Probably the AM noise is worse than this already (we should remember to hook up a simple AM stabilizer in 2016, as well as the ISS).

What else are the main features of this improvement? I can reward a good summary with some Wagonga.

I'm not claiming we need to modify the frequency source immediately as we are not limited by the oscillator amplitude or phase noise.
I just wanted to note something in mind before it goes away quickly.

Alberto's T1000461 tells us that the oscillator and phase noise are degraded by factor of ~3 and ~5 due to the RF chanin.
My diagram is possible removal of up-down situation of the chain.

Maybe more direct improvement would be:

- Removal of two amplifiers out of four. The heat condition of the box is touch thought it is not critical.

- The modification will allow us to have a spare 11MHz channel at 1X2 rack that would be useful for 3f modulation.

I, by chance, found  that my windows partition has Vision32 installed.
So I run my usual curvature characterization for the TT phasemaps.

They are found under this link
https://nodus.ligo.caltech.edu:30889/40m_phasemap/40m_TT/（requires: LVC credentials)
or
/cvs/cds/caltech/users/public_html/40m_phasemap/40m_TT

asc/ (ascii files) --> .asc files are saved in Wyko ascii format.
bmp/ (screen shots of Vision32)
mat/ (Matlab scripts and results)
opd/ (Raw binary files)

Estimated radius of curvature

Mirror / RoC from Vision32 / RoC from KA's matlab code
G&H "A" 0864 / -527.5 m / -505.2 m
G&H "B" 0884 / -710.2 m / -683.6 m
LaserOptik SN1 / -688.0 m / -652.7 m
LaserOptik SN2 / -605.2 m / -572.6 m
LaserOptik SN3 / -656.7 m / -635.0 m
LaserOptik SN4 / -607.5 m / -574.6 m
LaserOptik SN5 / -624.8 m / -594.3 m
LaserOptik SN6 / -658.5 m / -630.2 m

The aperture for the RoC in Vision32 seems a bit larger than the one I have used in the code (10mm in dia.)
This could be the cause of the systematic difference of the RoCs between these, as most of our mirrors
has weaker convex curvature for larger aperture, as seen in the figure. (i.e. outer area is more concave
after the subtration of the curvature)

I did not see any structure like Newton's ring which was observed from the data converted with SXMimage. Why???

Is it possible to calculate astigmatism with your tools?  Can we get curvature in X/Y direction, preferably aligned with some axis that we might align to in the vacuum?

We received the custom cables to test the new suspension electronics. They are under my desk. So we are ready.

This batch was a small one - the company says that they can make molded cables if we have a minimum order, something to consider I gues.s.

Update 1900 11 Feb: I verified that the pin outs of the cables are as we intended (for one set of each type of cable). Because this was a small order, the connectors have metal shells, and so for cable #2 (sat box to flange), the two shells are shorted to each other. I can't verify if the shield is isolated from the shell on J5 without cutting open the cable. One thing that occurred to me is that we should give pins 5,8,11 on J4 and 16,20,24 on J5 (respectively) unique identifiers. They should only be shorted to GND on the circuit board itself. To be fixed for the next iteration. I uploaded some photos here.

I was unable to measure the capacitance of the cable using the LCR meter, and didn't opt to try any other method.

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...

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 thought a little bit about the design of the preamp we want for the demodulated ALS signals today. The requirements are:

1. DC gain that doesn't cause ADC saturation.
2. Audio frequency gain that allows the measured beat signal spectrum to be at least 20dB the ADC noise level.
3. Electronics noise such that the measured beat signal spectrum is at least 20dB above the input-referred noise of this amplifier.
4. Low pass filtering at the input to the differential receiving stages, such that the 2f product from the demodulation doesn't drive the AD829 crazy. For now, I've preserved the second-order inductor based LPF from the original board, but if this proves challenging to get working, we can always just go for a first-order RC LPF. One challenge may be to find a 2.2uH inductor that is compatible with prototype PCB boards...
5. Differential sending, since this seems to be definitively the lower noise option compared to the single-ended output (see yesterday's measurement). The plan is to use an aLIGO AA board that has differential receiving and sending, and then connect directly to the differential receiving ADC.

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?

*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.

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.

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.

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?)

I picked up a unit of D1900068 SR785 accessory box from Dean's office at Downs. 

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.

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.

Suspension Actuator noise:

There are 3 main sources of electronics noise which come in through the coil driver:

1. Voltage noise of the coil driver.
   1. The input referred noise is ~5 nV/rHz, so not a big issue.
   2. The Johnson noise of the output resistor which is in series with the coil is sqrt(4*k*T*R) ~ 3 nV/rHz. We probably want to increase this resistor from 200 to 1000 Ohms once Gautam convinces us that we don't need that range for lock acquisition.
2. Voltage noise of the dewhitening board.
   1. In order to reduce DAC noise, we have a "dewhitening" filter which provides some low passing. There is an "antiDW" filter in the digital part which is the inverse of this, so that when they are both turned on, the result is that the main signal path has a flat transfer function, but the DAC noise gets attenuated.
   2. In particular, ours have 2 second order filters (each with 2 poles at 15 Hz and 2 zeros at 100 Hz).
   3. We also have a passive pole:zero network at the output which has z=130, 530 Hz and p = 14, 3185 Hz.
   4. The dewhitening board has an overall gain of 3 at DC to account for our old DACs having a range of +/-5 V and our coil drivers having +/- 15 V power supplies. We should get rid of this gain of 3.
   5. The dewhitening board (and probably the coil driver) use thick film resistors and so their noise is much worse than expected at low frequencies.
3. DAC voltage noise. 
   1. The General Standards 16-bit DACs have a noise of ~5 uV/rHz.
4. the satellite box is passive and not a significant source of noise; its just a flaky construction and so its problematic.

I've removed the SOS coil driver (D010001-B, S/N B151, labelled "SRM") + Universal Dewhitening Board (D000183 Rev C, S/N B5172, labelled "B5") combo for SRM from 1X4, for photo taking + inspection.

I first shutdown the SRM watchdog, noted cabling between these boards and also the AI board as well as output to Sat. Box. I also needed to shutdown the MC2 watchdog as I had to remove the DAC output to MC2 in order to remove the SRM Dewhitening board from the rack. This connection has been restored, MC locks fine now.

I believe the ETMs and ITMs are different from the others.

  Summary:

The pin-outs for the DAC interface board have been determined.

Details:

• I used a temporary break-out cable (pic attached) and connected the 40pin IDC connector on this to the DAC interface board at 1Y4.
• I had a hypothetical pin-out map which was to be verified. So I connected pairs of ribbon wire to an oscilloscope in the configuration which I believed to be correct, and then used awggui to send a 3Hz, 10000 count sine-wave to the corresponding channel via the excitation points set up earlier.
• I verified that the correct waveform showed up on the scope screen. I then tried sending the same signal to another DAC channel and verified that there were no accidental shorts/bad connections. The signal was fairly noisy, but this was probably because of the makeshift connections.
• Repeated the above for all 8 channels in the bank marked 9-16 on the DAC interface board.

Turns out that my deductions using the D0902496 wiring diagram, a spare D080303 DAC to IDC adaptor and a multimeter were correct! The pin outs as determined by this test are sketched in the graphic below.

To Do:

•  Now that the pin-outs have been determined, I need to go about making the custom ribbon that will connect the 40pin IDC on the DAC interface board to the 10-pin IDC on the PZT driver board. Because there is a pair of wires that will have to be 'skipped' while going from the 40-pin to the 10 pin IDC (corresponding to the pair not-connected between two DAC channels on the 40-pin IDC), this may be tricky.

Misc:

The excitation points added to the simulink model are still there, I plan on keeping it as such till I finish installation of the boards as they will be useful for testing purposes.

Pin-Outs of the DAC to IDC Adaptor (D080303) inside the "DAC Interface Box at 1Y4":

Makeshift break-out ribbon cable:

 I did just this, and it looks okay to me:

 I redid the measurement with the appropriate units set on the SR785. Power spectral density plots for no output (top), 500Hz, 1000 counts amplitude sine wave (middle) and 2000Hz, 1000 counts amplitude (bottom) are attached, with the right unit on the Y-axis.

Those 'peaks' for the oscillations seem ridiculously broad.  I think you should look again, really quickly, with smaller bandwidth at, say, the 2kHz oscillation, to make sure it looks reasonable.

 Summary:

I measured the maximum output of the DAC at 1Y4 as well as its power spectrum. The results are as follows (plots below):

• Maximum amplitude of differential output: + 10V.
• Power spectrum has a peak at 64 kHz.

Therefore, the gain of the high-voltage amplification stage on the PZT driver boards do not need to be changed again, as the required output range of 0-100V from the DAC board was realised when the input voltage ranged from -10V to +10 V w.r.t ground. The AI board converts the differential input to a single ended output as required by the driver board.

I will now change some resistors/capacitors on the AI board such that the position of the notches can be moved from 16k and 32k to 64k and 128k.

Procedure:

 Max. amplitude measurement

My previous measurement of the maximum output amplitude of the DAC was flawed as I made the measurement using a single channel of the oscilloscope, which meant that the negative pin of the DAC channel under test was driven to ground. I redid the measurement to avoid this problem. The set up this time was as follows:

• Positive pin of DAC connected to channel 1 of oscilloscope using break out cable and mini-grabber probe
• Negative pin of DAC connected to channel 2 of oscilloscope
• Grounds of channels 1 and 2 connected (I just hooked the mini-grabbers together)
• Measurement mode on oscilloscope set to channel 1 - channel2
• Used excitation points set up earlier to output a 3 Hz sine wave with amplitude of 32000 counts from channel 9 of the DAC. 

The trace on the oscilloscope is shown below;

So with reference to ground, the DAC is capable of supplying voltages in the range [-10V 10V]. This next image shows all three traces: positive and negative pins of DAC w.r.t ground, and the difference between the two.

 Power spectrum measurement

I used the SR785 to make the measurement. The set up was as follows:

• Positive pin of DAC to A-input of SR560
• Negative pin of DAC to B-input of SR560
• A-B output to Channel 1 input A of the SR785
• SR785 configured to power spectrum measurement

Initially, I output no signal to the DAC, and obtained the following power spectrum. The peak at 65.554 kHz is marked.

I then re-did the measurement with a 200 Hz (left) and 2000 Hz(right), 1000 counts amplitude (I had to change the Ch1 input range on the SR785 from -18dBm to -6dBm) sine wave from channel 9 of the DAC, and obtained the following. The peaks at ~64 kHz are marked.

Now that this peak has been verified, I will work on switching out the appropriate resistors/capacitors on the AI board to move the notches from 16k and 32k to 64k and 128k. 

We need the unit of the voltage power spectrum density to be V/sqrt(Hz).
Otherwise we don't understand anything / any number from the plot.

The first picture shows that there is indeed a DAC next to the ADC in the LSC IO chassis. The second picture shows how there are two cables, each one carrying 8 channels of DAC. The third one shows how these come out of the coil drivers to handle the Tip/Tilt mirrors which point the beam from the IMC into the PRC. It should be the case that the second Dewhitening filter board can give us access to the next 8 channels for use in driving an audio signal into the control room or an ISS excitation.

The DAC was fine.  I realized tonight that the digital filter bank outputs were off, so I wasn't actually sending signals out.  Oooops. 

Gautam and I looked into the DAC noise contribution to the noise budget for homodyne detection at the 40m. DAC noise is currently the most likely limiting source of technical noise.

Several of us have previously looked into the optimal SRC detuning and homodyne angle to observe pondermotive squeezing at the 40m. The first attachment summarizes these investigations and shows the amount of squeezing below vacuum obtainable as a function of homodyne angle for an optimal SRC detuning including fundamental classical sources of noise (seismic, CTN, and suspension thermal). These calculations are done with an Optickle model. According to the calculations, it's possible to see 6 dBvac of squeezing around 100 Hz.

The second attachment shows the amount of squeezing obtainable including DAC noise as a function of current noise in the DAC electronics. These calculations are done at the optimal -0.45 deg SRC detuning and 97 deg homodyne angle. Estimates of this noise are computed as is done in elog 13146 and include de-whitening. It is not possible to observe squeezing with the current 400 Ω series resistor which corresponds to 30 pA/rtHz current noise at 100 Hz. We can get to 0 dBvac for current noise of around 10 pA/rtHz (1.2 kΩ series resistor) and can see 3 dBvac of squeezing with current noise of about 5 pA/rtHz at 100 Hz (2.5 kΩ series resistor). At this point it will be difficult to control the optics however.

So it seems reasonable to reduce the DAC noise to sufficient levels to observe squeezing, but we will need to think about the controls problem more.

Gautam and I redid our calculations, and the updated plot of squeezing as a function of DAC current noise per coil is shown in the attachment. The current noise is calculated as the maximum of the filtered DAC noise and the Johnson noise of the series resistor. The total noise is for four optics with four coils each.

The numbers are worse than we quoted before: according to these calculations we can get to 0 dBvac for current noise per coil of about 2.4 pA/rtHz at 100 Hz.