Global damping screens are in progress for the new global damping infrastructure Jamie discussed in log #8159. The main overview screen is /opt/rtcds/caltech/c1/medm/c1sus/master/C1SUS_GLOBAL.adl. The overview screen links to a few sub-screens in the same directory called C1SUS_GLOBAL_DAMPFILTERS.adl, C1SUS_GLOBAL_GLOBALTOLOCAL.adl, and C1SUS_GLOBAL_LOCALTOGLOBAL.adl.

This global damping is in intended to damp the 4 test masses along global interferometer degrees of freedom that are orthogonal to the cavity signals. Ideally the result will be that OSEM sensor noise from the damping loops is invisible to the cavity signals. Mismatches in the suspensions' dynamics and gains will cause some noise to leak through anyway, but we should be able to tune some of this out by carefully scaling the drives to each suspension.

 I made a minor modification to install some output filters in the new global damping GLOBAL box in c1sus.mdl. These will be needed for tuning the suspension drives to compensate for mismatches in the pendulums.

I recompiled and installed the model, but did not start it. Basically same as Jamie left it in 8159. Interestingly, I did not see the new POSOUT that was put in before the SUSPOS DOF filter. I made sure to reopen the .mdl file fresh before making more mods, but for some reason I do not see that update...

While doing initial measurements for the new global damping infrastructure I discovered that the ETMY loop between the OSEM actuation and the OSEM sensors has a gain that is 2.5 times greater than the ITMY.  The result is that to get the same damping on both, the damping gain on the ETMY must be 2.5 times less than the ITMY. I do not know where this is coming from, but I could not find any obvious differences between the MEDM matrices and gains.

I uploaded a screenshot of measured transfer functions of the damped ITMY and ETMY sus's. Notice that the ETMY measurement is 2.5 times higher than the ITMY. The peak also has a lower Q, despite having the same damping filters running because of this mysterious gain difference. Lowering the damping gain of the ETMY loop by this 2.5 factor results in similar Q's.

The global damping input and output matrices were installed to run for the Y-arm. Since we are using just one arm for now, only the DARM and CARM DOFs were entered into the matrices.

The input matrix was set to have elements with magnitudes of 0.5 while the output matrix was set to have elements with magnitudes of 1. The input matrix gets the 0.5 because the sensor signals must be avergaed for each global DOF, to make an 'equivalent sensor' with the same gain. The output matrix gets magnitudes of 1 so that the overall gain of the global loops is the same as the local loops. A transfer function was measured on the CARM loop to check that the overall gain is in fact the same as the measured ITMY and ETMY loops.

Simple damping filters were installed for the ITMY and ETMY as well as the global y arm CARM and DARM loops.

The ETMY output tuning filter ETMY_GLOBPOS was set to have a gain of 0.4 because there is an extra gain of 2.5 relative to ITMY in some mysterious place as discussed in log 8172.

 New excitation points were added after the global damping loops for more testing options. The updated c1sus.mdl model was re-committed to the svn. Two interesting simulink 'requirements' were found during this minor modification. First, excitation points must be placed on the top level of the diagram. If they are in a subsystem you will get compiling errors. Second, the excitation name must end in _EXC. It will compile OK if you don't do this, but the excitation points will not put out any excitations.

To do further investigation on the mysterious gain factor of 2.5 between the ETMY and ITMY POS damping loops, I measured TFs in the POS direction to the locked YARM signal for each. This provides an additional sensor, common to both, so we can see if the gain is coming from the actuation side or sensing side of the damping loops. The difference in these TFs is about 

2.895

So it seems the majority of the damping gain difference is on the actuation side with some small difference on the sensing side. In order to allow for the later splitting of YARM LSC control between ITMY and ETMY (global damping and the cavity control must be along the same coordinate system), I placed this gain of 2.95 in ITMY_LSC.

To get a first measure of the relative performance of global damping to local damping I measured some TFs between the sensor signal inputs and YARM. So first, while the cavity was still locked with just ETMY, I measured a TF between C1:SUS-ITMY_SUSPOS_EXC and C1:LSC-YARM_IN1. Second, I split the cavity control evenly between the ETMY and ITMY by adjusting C1:LSC-OUTPUT_MTRX. I turned off the local damping and turned on the common DOF global damping (called CARM at this point despite being on just one arm). I then repeated the same TF but driving from C1:SUS-GLOBAL_CARMDAMP_EXC.

The resulting TFs are displayed in the attached figure. The blue curve is then the TF from local damping sensor noise to YARM. The green is global damping sensor noise to YARM. The suppression between local to global is in red. The global damping curve is about 50 to 100 times lower (better) than local damping. This can probably be improved with further tuning to account for remaining differences between the ITMY and ETMY.

Note, the damping loop used in the filter modules for all of these is zpk(0,[15 15],1), with a gain of 30. This purposely has little high frequency filtering so it is easier to see the influence on YARM.

Brett and Kamal

The global damping testing for the week is now complete. The c1sus.mdl simulink diagram settled on the attached screenshot. The top level of c1sus.mdl is shown on the left zoomed in over the new global damping block. The right shows the inside of that block. Also attached in the second screenshot are two of the modal damping MEDM screens. The left shows the main overview screen, the right shows the global damping filters. The overview screen is called C1SUS_GLOBAL.adl and is found in ...medm/c1sus/master/.

We have measured transfer functions and power spectra that show that global damping, with just a moderate amount of tuning (30 minutes of work) reduces the OSEM damping noise seen by YARM_IN1 by a factor between 50 and 80. Log 8193 highlights the transfer function measurements. The power spectra directly measure the noise in the cavity. I am not putting that data here because I have to catch. I will process the data and post it here later.

Overall the global damping tests appear to have been successful, isolating (not removing) the test mass damping noise from the cavity by almost 2 orders of magnitude. Presumably even more isolation is possible with more tuning.

Here is an amplitude spectrum plot of y-arm cavity noise with a 50 Hz cutoff damping filter of the form zpk(0,[50;50],1). The low passing of this filter was intentionally extremely poor in order to see the damping noise in the cavity. The blue trace is the noise with no damping, which may be considered the 'best case' scenario from a noise point of view. The green has regular local damping on the ITMY. The ETMY has no damping for this measurement because the cavity control feedback to the ETMY takes care of its control when the cavity is locked. Notice the the large increase in noise from 40 Hz to 250 Hz, up to 1 order of magnitude. This noise is from the OSEM sensors passing through the damping loops. The red curve shows the y-arm noise with the exact same damping, except it is now applied in the global scheme. In this case, the damping noise falls completely below the baseline level of the cavity and becomes indistinguishable from the 'no damping' case.

If the damping injected enough noise I'd expect we would see a drop of 50 to 80 times switching from local to global. That is, the same factor measured in the transfer functions listed in log entry 8193.  However, the damping noise is only at most 1 order of magnitude above the baseline in this measurement. We would have to increase the damping noise by about another order of magnitude before we could expect to see the global damping noise in the cavity measurement.

The units of the cavity displacement in the plot were calculated using the 1.4e12 counts per meter calibration in log 6834. The measured UGF of the LSC loop at the time was 205 Hz. The peak in the plot above 200 Hz appears to be from this unity crossing. Moving the UGF also moves this peak.

Moral of the story: global damping can isolate the damping noise pretty well from the cavity signal.

The new laptop for the Beam Profiler is ready for usage.

From the last meeting, we came to the idea of having 2 laptops that are designated for the beam profiler specifically. We now one laptop over at WestBridge, and one here at 40m. This makes it easier to travel with just the Beam Profiler heads, rather than carrying the Profiler heads + laptop. This machine has been named Stella by ChatGPT. Feel free to use it! (Only runs Windows 10)

We aligned and locked x and y arms.

MCL loop makes arms lock unstable, adds a lot of noise at frequencies 60-100 Hz. We'll fix it.

At some point we were not able to lock because of ADC overflows of PO signals. They happened if whitening filters were enabled. So we reduced the gain of POX whitening filters down to 36 dB and POY - to 39 dB. Now cavities can be locked with whitening filters.

Also we changed the pedestal of the lens in the beam path to the POX because the beam was too high.

We aligned and locked xarm for green.

I will do some experiments on acoustic noise canceling during my stay.
Now I am planning to cancel acoustic noise from PMC and see how the acoustic noise work and how we should place microphones.

First, I measured the noise in microphones and its circuit.

-blue, green, red, solid lines; microphone signals
-blue, green, red, dashed lines; un-coherent noise in signals
-yellow, black, solid lines; circuit noise (signal input is open, not connected to the microphones)

We can see the acoustic signal above 1 Hz, and the circuit does not seem to limit its sensitivity. But I do not know why yellow and black is so different. I will check it tomorrow.

 Sorry for my poor explanation.

I measured this by the same way as you measured the instrumental noise of seismometers.
I put the three microphones at the same place so that the three can hear the same sound. I did not make any sounds, just put them in the lab.
The signals from microphones are all amplified by the circuit.
And I took the correlations of each signals and two others and got the noise (dashed lines) by subtracting the correlated signal from the original signal.

So,
-The signal is the acoustic sound in the lab, amplified by the circuit.
-Three lines are from three different microphones.
-Dashed lines are subtraction of coherent signal from the original.
-Yellow and black lines are from different amplifiers in the same circuit box. The circuit has 6 channels.
-I did not calibrate the signals I got by DTT since I do not know the calibration factor now. It is just the number I got from the real time system.

Previous results
I am measuring the noise level of the microphones. The circuit does not seems to limit their sensitivities but the circuit's noise seems to be different from other channels.

Measurement
I measured the circuit noise of all 6 channels. (input open)
(mic_open.png)
The noise level is about 10 times different from the others.

Comparing the acoustic signal, microphone+circuit noise, and ADC noise;
(mic_noise.png)
- blue; acoustic signal
- green; microphone+circuit noise
- red; circuit (the data was not took simultaneously.)
- sky blue; ADC noise

To do
I will remake the circuit though the circuit does not limit the sensitivity. I would like to make sure that the circuit does not affect badly and to make the circuit noise level the same.
At the same time, I will get the PMC control signal and see coherence between it and acoustic sound.

 Thank you for changing the sample rate!
Also we have to change the Anti-Aliasing filter, as Jamie said.

Now my question is, whether S/N ratio is enough at high frequencies or not. The quality of EM172 microphone is good according to the data sheet. But as you can see in previous picture, the S/N ratio around 1kHz is not so good, though we can see some peaks, e.g. the sound that a fan will make. I have to check it later.
And, is it possible to do online adaptive noise cancellation with a high sampling rate such that computationally expensive algorithms cannot be run?

[Raji, Ayaka]

Thanks to Den, power supplies for microphone circuit are changed.
So I measured the microphone noise again by the same way as I did last time.

  solid lines: acoustic noise
 dashed lines: un-coherent noise
black line: circuit noise (microphone unconnected)

The circuit noise improves so much, but many line noises appeared.
Where do these lines (40, 80, 200 Hz...) come from?
These does not change if we changed the microphones...

Anyway, I have to change the circuit (because of the low-pass filter). I can check if the circuit I will remake will give some effects on these lines.

 I do not think they are acoustic sounds. If so, there should be coherence between three microphones because I placed three at the same place, tied together. However, there are no coherence at lines between them.

 I am trying to find what limits the reduction rate with wiener filtering.
I did some calculations below:

Reduction rate estimation by microphone noise

When the instrumental noise (noise in microphone) and noise injected to signal after the acoustic signal is injected exist, the noise cancellation rate is limited. (I will write a short document about it later.) I assumed that there is only instrumental noise and that the other noise in PMC is below enough, and calculated the cancellation rate. The instrumental noise is modeled according to the measurement before (ELOG).
The green line is the original PMC signal, the red one is PMC residual error, and the blue one is PMC residual error estimated by the cancelling rate.
Around 30 - 80 Hz, the wiener filtering seems to be already good enough. However, I do not know what limits the cancellation rate (such as 100 - 200 Hz).

Filtering signals

I hypothesized that the wiener filter is not good because of some peaks or other noise. So I filtered the PMC signal and mic signal to see the difference.
The red line is wiener filter with no filters, the blue one is with filters (low pass, high pass, and notch).
The wiener filter seems to get smoother but the PMC residual error did not change at all.

I will attach a document which describes how the noise affect the wiener filter and the noise cancellation ratio.

And I re-estimate the SN ratio in the microphone (but still rough):

The yellow line is modeled signal level, and cyan line is modeled noise level.

Then, the estimated filtered residual noise is:

The noise is already subtracted enough below 80 Hz even though there is still coherence.
Above 300 Hz, the residual error is limited by other noise than acoustic noise since there is no coherence.
I am not sure about the region between 100-300 Hz, but I guess that we cannot subtract the acoustic noise because primary noise (see the document), such as a peak at 180 Hz, is so high.

[Rana, Ayaka, Jenne]

We aligned the REFL beam to the center of PD.
Also we removed the small black parts from mirror holder so that the beam is not clipped. They are originally for holding the mirror, but the mirror should be held by the small screw on the side of the mirror mount. This screw was hidden by the label, so we moved the label on the right hand side of the mirror mount (See a picture below).  

Also we removed the half-wave plates and PBS so that laser power is increased.

Then I aligned the beam for PMC, locked MC, and centered the beam spots.

The MC2 pitch is a little bit high but still close enough to the center.

Jenne had also centered the beam spots on QPDs for WFS.

 In order to perform acoustic noise cancellation with MCL signal, I am trying to find sweet spots for microphones.

I set microphones at various places around MC chambers, and see how coherent microphones and MC signals are.
I had checked the half part of MC.

• data set #1
  place where I set the microphones (left), MCL signal (blue) and its error (green) (right top), and coherence between microphones (original: fine lines, error: thick lines) (right bottom).
  
• data set #2
   
•  data set #3
  
• data set #4
  

The acoustic noise around the MC2 chamber is most critical so far. I could subtract the signal and the sensitivity got 2 times better.
I will see the acoustic coupling from the other side of MC.

 I got two seismometers and one microphone back from Tara.

They are now near the Gurlap under the MC.

Motivation

In order to estimate whether we can see acoustic coupling in arms or not, we have to calibrate signals to phase noise.

Method

I used the same method as Yuta and Jenne did (6834).
I switched from ETM locking to ITM locking since only ITM actuators are calibrated (5583), and measured the open loop transfer function and the transfer function from ITM excitation to POX/POY error signal. Then I can estimate the calibration value H [counts/m] from POY/POX error signal to displacement.

Results

Yarm; H = 9.51 x 10¹¹ counts/m
 

Xarm; H = 6.68 x 10¹¹ counts/m

Phase noise in arms:

blue; Xarm, green; Yarm

Next Step

I will calibrate the acoustic signal and see if it is reasonable that we can see the acoustic coupling signal in the arms.
But I guess it is difficult. Actually I have not seen coherence between ETM feedback signals and acoustic sounds yet. (I measured acoustic noise near POX and in PSL table.)

If I find that it is hopeless, I will create some sounds and try to measure transfer function from acoustic sound to arm cavity signals.
I am interested in how the transfer function calculated by wiener filtering is different from the measured transfer function.

Note

I found that we do not have enough phase margin. This is why the arm locking is not so stable.

I uploaded a zip file that contains data files used for the calibration.

OLTF_x/y.txt: the open loop transfer function (measured by IN1/IN2 in arm servo filter bank).
coh_x/y.txt: coherence of OLTF. I used the data where coherence > 0.98.
ext_err_x/y.txt: the transfer function from ITM excitation signal to POX/POY error signal.
coh_x2/y2.txt: coherence of ext_err. I used the data where coherence > 0.98.

The LSC whitening filter was off because the xarm was unlocked when the POX Q whitening filter was turned on. (We have to study what was wrong.)
The SUS whitening filters were on.
The all digital filters except +6dB filter were on.

[Rana, Jamie, Ayaka]

We could not lock the arms with MC_L loop on. Therefore we measured the change in YARM error signal when the MC_L is turned on.

(data; POYerr_MCF.xml in zip file)

Green line; POY error signal when MCL loop was on and the YARM loop gain (0.5) was so high that the saturated control signal made funny peak around 250 Hz.
Blue line; POY error signal when MCL loop was off and the YARM loop gain was low (0.2).
Pink line; POY error signal when MCL loop was on (the gain was -300) and the YARM loop gain was low (0.2).
Red line; POY error signal when MCL loop was on, another low pass filter (2nd order, cut off frequency of 55Hz) was added to MCL loop and the YARM loop gain was low (0.2).

We also changed the filter trigger in order to lock YARM. The FM7 and 8 trigger was turned off. It means that spectrum above was took with FM2,3,4,5,6,9,10 on. Whitening filters were also on.

MCL control signal makes the arm spectrum bad because the MCL control signal moves MC2 mirror additionally and adds extra frequency noise.
Ideally, error signal should be the same at higher frequencies and go down at the lower frequencies when the MCL loop is on because MCL signal should suppress the seismic noise.

Before we added the LPF, MCF/MCL loop cross over (which was taken with the template /users/Templates/MC/MCL-MCF_xover-2012-8-23.xml) is below;

(MCL-MCF_xover.xml in zip file)

After the LPF is added, the cross over has been changed as below;

(MCL-MCF_xover2.xml in zip file)

For now, I will just turn off the MCL loop for the acoustic noise experiments.

 Last Thursday, I put the speaker and my laptop in the PSL table, and make triangular wave sound with the basic frequency of 40Hz, and Gaussian distributed sound.
(I create the sounds from my laptop using the software 'NHC Tone Generator' because I could not find the connector from BNC to speaker plug.)
And I measured the acoustic coupling in MCF signal. The all the 6 microphones were set in PSL table around PMC and PSL output optics. 

The performance of the offline noise cancellation with wiener filter is below.
(The target signal is MCF and the witness signals are 6 microphones.)

• With Gaussian sound (Sorry for wrong labeling 'XARM' and no calibration)
  
• With 40Hz Triangular sound (Sorry for no calibration again)
  

I can see some effects on MCF due to the sound on PSL table. Though I can subtract some acoustic signal and there are no coherence between MCF signal and mic signals, still some acoustic noise remains.
This is maybe because of some non-linearity effects or maybe because we have other effective places for acoustic coupling measurement. More investigations are needed.

Also, I compared the wiener filter and the transfer function from microphones signal to MCF signal. They should be the same ideally.

(Left: Wiener filter, Right: Transfer function estimated by the spectrum. They are measured when the Gaussian sound is on.)

These are different especially lower frequencies than 50 Hz. The wiener filter is bigger at lower frequencies. I guess this adds extra noise on the MCF signal. (see the 1st figure.)
The wiener filter can be improved by filterings. But if so, I want to know how can we determine the filters. It is interesting if we have some algorithms to determine the filters and taps and so on.
The more investigations are also needed.

 I have been searching for the way we can subtract signal better since I could see the acoustic coupling signal remains in the target signal even though there are no coherence between them.

I changed the training time which is used to decide wiener filter.
I have total 10 minutes data, and the wiener filter was decided using the whole data before.

(Right: the performance with the data when the triangular sound was created. Left: the performance with the data when the gaussian sound was created.)

I found that the acoustic signal can be fully subtracted above 40 Hz when the training time is short. This means the transfer functions between the acoustic signals and MCF signal change.
However, if the wiener filter is decided with short-time training, the performances at lower frequencies get worse. This is because wiener filter do not have enough low-frequency information.

So, I would like to find the way to combine the short-time training merit and long-time training merit. It should be useful to subtract the broad-band coupling noise.

[Den, Ayaka]

We found that seismometer was working and the calibration in the filter banks should have been wrong.
We turned off the all FM2 filter in RMS filter banks.

We also installed STS seismometer. It is under the BS. Now we have spectrum of three seismometers.

RA: the above plot is kind of unreadable and useless. Please replace with something legible and put in some words about why there is a wrong filter, what exactly it is, etc., etc. etc. And why would you leave in a filter which is not supposed to be on? We might as well leave a few secretly broken chairs in the control room...

In order to calibrate MC_F signal, I need to know the calibration value from thorlab's PZT driver to laser frequency.
The calibration value should be ~ 15MHz/V (the PZT driver has 15 gain, and the laser has the calibration value of ~ 1MHz/V according to the laser spec sheet), but I want to confirm it.
This can be measured by sweeping the input voltage of the PZT driver and see the transmission signal from unlocked PMC.
 

1. Response of PMC transmission when the signal is inputted to laser PZT

I inputted 0.2 Hz triangular wave with 5Vamp and 2.5V offset into the PZT driver and see the transmission signal from PMC. After the PZT driver and before the laser, there is an analog low pass filter but its cut off frequency is 1 Hz so I did not take it into consideration. 
(TEK00000.CSV, TEK00001.CSV in the zip file)

I could not the side-band resonances. I guess it was because the generated signal is not big enough (but still the maximum range of the signal generator.)
Therefore, in order to calibrate the input voltage to the frequency, I need to know finesse or FWHM frequency.
 

2. Responce of PMC transmission when the voltage of PZT on the PMC is swept

In order to measure the finesse and FWHM frequency, I also swept the PMC PZT voltage with the DC offset slider at the FSS.adl and tried to measure the finesse of PMC. (reference: elog #904)
(PMC-PZTcal_121203.xml in the zip file)

The result of fitting:

V_FSR (the PZT voltage difference between the 2 resonances) ~ 63 +/- 7 V (= 731MHz (given))
V_FWHM (the PZT voltage to sweep FWHM) ~  0.32 +/- 0.04 V (~ 3.7 MHz)
Finesse ~ 200 +/- 30

However, this finesse value is much smaller than the value on the Wiki, 800. (Manasa showed me.)
V_FSR is comparable to the result Rana got at the referenced elog. But I am not sure about the V_FWHM because it is hard to figure out how large the PZT voltage changed from the template file (PMC-PZTcal.xml).
Are those mode wrong? But if so, where is the correct mode resonances? I think they should be visible...
 

3. Calibration value 

When I know the FWHM frequency, I can calibrate the input on the PZT driver into laser frequency.

The results are:

if I take the finesse of 800 and FSR of 731 MHz (the values on the Wiki): ~5.0 MHz/V
if I take the finesse of 200 and FSR of 731 MHz (the measured value): ~20.0 MHz/V

Actually, the measured value is closer to the value calculated from the spec sheet.

Hmm... Does anyone find falses in my measurement?
If not, the finesse can be 4 times smaller than the value which was 5 years ago?

BS chamber seemed to be kicked again around 10:00 am today.

I moved PZT mainly in YAW and locked both arms. I adjusted the beam to be almost on the center of both ETM by sights.

 In order to see the acoustic coupling on arm signals, I set 6 microphones and the speaker on the AP table. The microphones are not seismically isolated for now.
I have a signal generator under the AP table.

When I played the 43 Hz triangular wave sound, I could see some coherence between POY error signal and microphones even though there is no peak in POY.

To Do:

• Try to subtract the acoustic signal and see with which microphone the acoustic signal can be subtracted best. But how can I find whether the signal is subtracted or not? Is coherence information enough?
• Make circuits for microphones to come to 40m.
• Make suspension systems for microphones. One idea is that the microphones should be suspended from bridges which is to be put around at the top of the tables since there is no space for stacks for each microphones.
• Prepare a new ADC.
• Perform the same measurements at the other tables, such as POX and POY.

I will just leave the picture of spectrum that shows the injected acoustic sound effects due to the oplevs.

red line: POY error without oplev feedback nor acoustic noise
blue line: POY error without oplev feedback but with acoustic noise
brown line: POY error with oplev feedback but without acoustic noise
green line: POY error with oplev and acoustic noise

You can see there is noise only at green line around 70 - 100 Hz. And it does not look like the acoustic signal is injected directly to the arms but the acoustic sound couples to the original noise source.

 Motivation

We observed that oplev servos affect the arm spectra badly (elog #7798). Some of them are fixed, but still they inject noise into the arms.
So I tried to turn the oplevs off and to see the acoustic noise effect. However, the mirrors moves so much that the signal does not seem to be linear any more, and the noise spectrum of arms changes especially around 60 - 100 Hz as you can see the spectrogram of YARM error signal below. This makes it difficult to find acoustic coupling noise. Therefore, I tried to fix the oplev servos so that the noise spectra do not get worse when the oplev servos are on.

Checking oplev UGFs

I checked the oplev open loop transfer functions. The UGFs of oplevs are all around 1-3Hz and phase margin looks enough except the BS oplev.
The gain of the BS oplev OLTF has so low that the signal is not fed back. Moreover, there is much phase delay in the BS feedback loop than the others'.
The counts of BS oplev sum is not changed so much for this 4 months, so the oplev beam seems to hit the BS correctly.
I am not sure what makes difference.

Clipped oplev beam fixed

Den and I found the output beam of ETMY oplev was clipped the other day. Also I found the scattered beam of ITMY oplev was on the edge of the mirror inside the vacuum and it made more scattered lights.
(before) -> (after)

  I fixed both of the clipped beam. But still the oplev feedback inject the noise into the arm. (red: oplev off, blue: oplev on)
   

 Sorry for the mess. I fixed it.

[Rana, Ayaka]

The BS oplev pitch feedback came back.

The problem was that 300^2:0 filter was off. And I turned on all the low pass filters (ELP35), then the oplev servo does not seem to inject big noise into the arms as long as I see the spectra of POY and POX. These low-pass filters will be modified tomorrow so that the acoustic coupling noise is minimized.

[Koji, Ayaka]

Last night, I injected acoustic noise at POX table and AS table with oplev controls on (LPF is on).

1. acoustic noise at the POX table

I set the microphones and speakers at the POX table and see the acoustic coupling.

I could see slight change around 40 Hz. This can be caused by the oplev feedback loop because the speaker was on the same table as the ITMX oplev.

2. acoustic noise at the AS table

I controlled XARM with AS error signal and set the microphones and speaker on the AS table.

The resonance a 200 Hz seemed to be enhanced. But still we are not sure that it is caused by acoustic noise. Because this resonance is enhanced when the OL gain is high, and the gain adjustment was so critical that this resonance was easily enhanced even when the acoustic noise is not injected. And sometimes it has gone away.

 I calibrated MC_F signal into Hz/rtHz unit using the transfer function from MC_F to PMC feedback signal.

Here is the diagram:
  

n_mcf is MC_F signal we can get at dtt. I measured n_pmc/n'_mcf using SR.

Other information I used:

G_out = 2.49/123.49 (see the document D980352-E01-C)

Fout has 1 pole at 10 Hz (see the document D980352-E01-C)

A_pzt = 371e+6/63 [Hz/V] (see elog)

F_wt has 1 pole at 100 Hz and 1 zero at 10 Hz.

Then, calibration transfer function of H is fitted as 1e+9/f [Hz/V]:

Then, the calibrated spectrum of MC_F is below:

This calibration have about 20 % error.
Compared to the spectrum in Jenne's paper (elog), above 20 Hz it seems to be laser frequency noise. But now we have extra unknown noise below 10 Hz.

Note: calibration value of laser's PZT is ~ 1MHz/V. This is reasonable compared to the data sheet of the laser. (This is calculated by combining result of H and transfer function of the circuit box1 and FSS.)

 Yesterday, I made new mounts for microphones.

I glued a microphone on a pedestal. The cables are attached loosely so that its tension does not make any noise.
At the bottom of the mount, I attached the surgical tube forming a ring by double-side tape so that it damps the seismic vibration.

I made 6 mounts and these are all on the AS table now.
I took some data of XARM signal controlled by AS.

My plan is to find/set an upper limit on acoustic coupling noise in AS signal.
The acoustic noise can be estimated by the Wiener filter, but it is not accurate because it may see residual correlation between AS and microphone signals that should be 0 when the data is long enough.
I will find/set an upper limit by the analysis based on Neuman-Pearson criterion, that is analog of a stochastic GW background search.
If I can find the acoustic coupling noise should be below the shot noise, I am happy. If not, some improvements may be needed someday.

I calibrated the AS error signal into the displacement of the YARM cavity in the same way as I did before (elog).

The open loop transfer function is:

The transfer function from ITMX excitation to AS error signal is:

Then I have got the calibration value : 5.08e+11 [counts/m]

The calibrated spectrum in unit of m/rtHz is

REF0: arm displacement
REF1: dark noise + demodulation circuit noise + WT filter noise + ADC noise (PSL shutter on)
REF2: demodulation circuit noise + WT filter noise + ADC noise (PD input of the circuit (at 1Y2) is connected to the 50 Ohm terminator)
(The circuit and WT filter seem to be connected at back side of the rack. Actually there is a connector labelled 'I MON' but it is not related to C1:LSC-ASS55_I_ERR)

Also we changed the AS gain so that ADC noise does not affect:

However, this did not make big change in sensitivity. I guess this means that circuit noise limits the sensitivity at higher frequencies than 400 Hz.
I tried to adjust the AS gain carefully but I could not do that because of the earthquake. Further investigation is needed.

There was an earthquake around 2:30 am. Now all the mirrors except SRM are damped.

No, I did not apply open loop TF to it (actually I could not measure the open loop TF because of the earthquake last night). So I should not have said it was the displacement.

Also I changed the AS gain with whitening filter on and xarm locked. Still it does not make any change.

Since I found that the the AS sensitivity seems to be limited by circuit noise, I inserted a RF amplifier just after the AS RF output.
Now, the sensitivity is improved and limited by the dark noise of the PD.

(Note: I did not apply the open loop TF on this xml file.)
REF3: dark noise + circuit noise + WT filter noise + ADC noise
REF4: circuit noise + WT filter noise + ADC

With this situation, I injected the acoustic noise:

REF5, 6, 7: with acoustic excitation
no reference: without acoustic excitation

We could see the coherence only at the same frequencies, around 200 Hz as we saw before (elog).

I am currently looking at the acoustic noise around both arms to see if there are any frequencies from machinery around the lab that stand out and to see what we can remove/change.

• Attachment 1 is a picture of the microphone and suspension system (bungee cords) that hangs from the cable trays to isolate it from vibrations.
• To record data, I used both the microphone (attachment 1) attach it its preamp connected to a spectrum analyzer in order get a graph of power spectral density, recording from 0-10k Hz and 10-100kHz. I started recording data at the furthest end of the x arm and worked towards the center taking measurements every couple of feet (ten rungs on the cable tray). 
• The second attachment is the first 5 psd I got from the furthest end of the x arm going 10 rungs on the cable tray closer each measurement.
• Going forward, I am going to take more measurements with greater resolution at the lower frequencies from 0-200 and stepping up from there by factors of 2.

Brief Summary: I am currently looking at the acoustic noise around both arms to see if there are any frequencies from machinery around the lab that stand out and to see what we can remove/change. I am using a Bluebird microphone suspended with surgical tubing from the cable trays to isolate it from vibrations. I am also using a preamp and the SR875 spectrum analyzer taking 6 sets of data every 1.5 meters (0 to 200Hz, 200Hz to 400Hz, 400z to 800Hz, 800Hz to 3200Hz, 3.2kHz to 12kHz, 12kHz to 100kHz).

·                Attachment 1 is a PSD of the first 3 measurements (from 0 to 12kHz) that I took every 1.5 meters along the x arm with the preamp and spectrum analyzer

·                Attachment 2 is a blrms color map of the first 6 sets of data I took (from 2.4m to 9.9m) 

·                Attachmetn 3 is a picture of the microphone set up with the surgical tubing 

Problems that occurred: settings on the preamp made the first set of data I took significantly smaller than the data I took with the 0dB button off and the last problem I had was the spectrum analyzer reading only from -50 to -50 dBVpk

• Problems that have occurred since my last post: All of the sudden, I was getting very strange data that was very quiet and did not match the previous input range of my last locations (see attachment). After resoldering the custom bnc connection cables with Lydia, which were in disrepair, and checking almost everything we could think of, we found that the gain dial on the preamp was turned all the down. Immediately after it was fixed, the data returned to expected values (based on neighboring locations and data taken at the last location before the problem occurred). 
• Updates: Since my last post, I have created a normalized blrms color map in addition to the one I already have. Additionally, I have started working on plotting the color maps next to a labeled, to-scale drawing of the lab, but have yet to complete it. 
• Attachment 1: comparison of the psds
• Attachment 2: blrms color map
• Attachment 3: normalized color map

[Gautam, Arijit]

PMC and IMC re-aligned and re-locked. Both cavities are staying locked. Arm cavities are also locked.

[Annalisa, Terra, Koji, Gautam]

Summary: We find a configuration for arm scans which significantly reduces phase noise. We run several arm scans and we were able to resolve several HOM peaks; analysis to come.

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As first, we made a measurement with the already established setup and, as Jon already pointed out, we found lots of phase noise. We hypothesized that it could either come from the PLL or from the motion of the optics between the AUX injection point (AS port) and the Y arm. 

• We first characterized the PLL loop phase noise by comparing the beat signal against the Agilent reference signal, and we found that the beat had lots of phase noise with respect to the reference. Decreasing the PLL gain, we got rid of the phase noise in the beat signal.
• Next, for the optical path length induced phase noise, we took the transfer function between TransMon and REFL signal rather than TransMon and Agilent reference signal. This takes advatage of the fact that the TransMon and REFL both see optical path length phase noise, which therefore gets canceled out in the transfer function. 

In this configuration, we were able to do arm scans where the phase variation at each peak was pretty clear and well defined. We took several 10MHz scan, we also zoomed around some specific HOM peak, and we were able to resolve some frequency split. 

We add some pictures of the setup and of the scan.

The data are saved in users/OLD/annalisa/Yscans. More analysis and plots will follow tomorrow. 

Summary

We installed two heaters setup on the ETMY bench in order to try inducing some radius of curvature change and therefore HOMs frequency shift.

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We installed two heaters setup.

Elliptic reflector setup (H1): heater put in the focus of the elliptical reflector: this will make a heat pattern as descirbed in the elogs #14043 and #14050. 

Lenses setup (H2): heater put in a cylndrical reflector (made up with aluminum foil) 1'' diameter, and 2 ZnSe lenses telescope, composed by a 1.5'' and a 1'' diameter respectively, both 3.5'' focal length. The telescope is designed in such a way to focus the heat map on the mirror HR surface. For this latter the schematic was supposed to be the following: 

This setup will project on the mirror a heat pattern like this:

which is very convenient if we want to see a different radius of curvature for different HOMs. However, the power that we are supposed to have absorbed by the mirror with this setup is very low (order of 40-ish mW with 18V, 1.2A) which is probably not enough to see an effect. Moreover, mostly for space reasons (post base too big), the distances were not fully kept, and we ended up with the following setup:

In this configuration we won't probably have a perfect focusing of the heat pattern on the mirror.

In vacuum connections

See Koji's elog #14077 for the final pin connection details. In summary, in vacuum the pins are:

13 to 8 --> cable bunch 0

7 to 2 --> cable bunch 2

25 to 20 --> cable bunch 1

19 to 14 --> cable bunch 3

where Elliptic reflector setup (H1) is connected to cables 0 and 1, and the lenses setup is connected to cables 2 and 3.

Installed setup

This is the installed setup as seen from above: