Control frequency of autosave

Isn't it possible to control how often InDesign autosaves your documents? Obviously my version, CS4, saves my 40 book documents every second minute and then I have to wait at least one minute before I can go on working. Maybe every 5 minutes would be sufficient!
I can't find any control feature anf InDesign help doesn't mention it at all.

There are no controls that you can access for writing the recovery files (which is the equivalent of an autosave, sort of). I doubt that the recovery file write is the problem, though, if you are having performance issues.
On very big preformance drag is cross-references, especially in books if they are inter-document references, and even more so if some of the files are closed.

Similar Messages

Table for mitigation control frequency

Hi,
We are trying to build a program to be able to send notifications to mitigation monitors. For this I am trying to find the tables where the relevant details of the mitigation controls are located.
I need the following information from the tables:
1. Mitigation ID - I have found this in HRP5354
2. Mitigation Name - I have found this in HRP1000
3. Mitigation Monitor - I have found this in HRT5320
4. Tcode in Reports - I have found this in HRT5320 + GRACACTION
5. Frequency - I have NOT found this.
The #5 - frequency of the report action is something that I am still missing. Please help me with the info, also suggest if there is a better way to get this information. Thanks!!
Thanks,
Sammukh

Hi Sammukh,
you can find the frequency in table GRFNCNREPORT.
In general you can easily find the tables with SE11 to check where the object is used.
Hope this helps.
Regards
Alessandro

Control frequency and number of cycles

I am trying to use a stepper motor and a motion controller to control a push rod. The push rod will fatigue sutures that are attached to a valve. My question is if there are any sample codes in labview that will help me control the frequency and the number of cycles. I will also have a load cell to give me a force profile.

Hello,
Have you looked at the Motion shipping examples for LabVIEW? There is a particular example, One-Axis Move (Accel - Max Velocity - Decel).vi, which is located at:
C:\Program Files\National Instruments\LabVIEW 7.1\examples\Motion\FlexMotion\One Axis.llb
This example allows you to control the acceleration, deceleration, velocity, and target position (in counts) of the movement. There are a number of other shipping examples located in the C:\Program Files\National Instruments\LabVIEW 7.1\examples\Motion\FlexMotion directory that may be of additional assistance in developing your code. I would suggest starting with the One-Axis Move example first, though.
Take a look at that example and let me know if you have any additional questions on this issue.
Regards,
Scott R.
Applications Engineering
National Instruments

How to apply notch filter in the PID control

Dear all,
I am using PID control in my positioning stage but the output is every noisy with some fixed frequencies(30,50,150Hz). I want to add a notch filter to take off those noise. However, I do not know how to apply the notch filter to the PID module. Could anyone tell me how and show me example if you have?
Thanks
best
Attachments:
PID_notch filter.png ‏21 KB
PID_notch filter.png ‏21 KB

Hi LYnn,
I am sorry for my late reply. My stage is driven by a voice coil actuator (VCA). The VCA as a resolution of 1 um and range of 6 mm. The NF of the stage does shift when the load is changing. Do you have any advice for controlling such kind of stage?
BTW, as far as I see, the control frequency is determined in the DAQ Assistant settings and its value is equal to "Sampling rate"/"Samples to read", am I right?
Thanks
best
Attachments:
VCA stage.png ‏160 KB

Torque Control on componet at different temperatures

Greetings,
I am in the process of building an application with Motion Control with change control modes from position control to Torque Control. Here is my situation. I am going to be twisting a rubber component from +- X Nm's at a particular frequency. However the component will be under temperature extremes and as the component heats up and cools down the stiffness changes therefore it may take more or less angle to achieve programmed torque. I still have to keep my frequency unchanged during the transition though. Is there automatic error compensation for velocity to keep the frequency unchanged even though the distance required to achieve Torque setpoints will change? I hope this is Clear enough to get a handle on what I am trying to achieve. I have no problem with Position Control, it is essentially the Torque Control frequency compensation that is what I need advice on.
Regards,
Wade

Wade,
for this purpose you will have to change the feedback for your motor to a signal that is proportional to torque. One option is to use a torque sensor with an analog output signal in the range of +/- 5 V or +/- 10 V. You can use an NI-Motion control device in analog feedback mode then and instead of defining a position profile you could define a torque profile then (e. g. by using contouring mode). With such a setup the motor will travel as far as it is needed to generate the torque profile, regardless of the travel distance.
To learn more about analog feedback, please have a look at this thread.
Regards,
Jochen

Can anyone help me with this- recording at audio signal in the upper frequency region of the audio s

Using Audition, how would I go about recording an audio signal (voice) at
14,500Hz? I know this would be outside the frequency range for the human ear to hear, but that is the point of my question. Any help would be very much appreciated
Thanks!!!!

You are thinking about it absolutely the wrong way if you think you can "convert" a voice file to be only 14.5kHz. You need to think of a control frequency (14.5kHz) being modulated by the voice line you record. You can't just change the pitch of the voice file or use the FFT filter to eliminate everything under 14.5kHz. It won't work that way.
Try it yourself. Record a voice line. Use Pitch Shifter or pitch strech to change the pitch on the voice. Crank it up all the way to highest possible pitch. Listen to it. Painful isn't it? Could you understand anything? Thought not. Then go to the FFT filter. Reset so it's flat. Add a break point (click on the line) at 14.5kHz. Then click and drag the line below 14.5kHz and bring it to zero. Listen to it. You get the occasional movment if you watch the Frequency Analysis window, but certainly nothing audible to my burned out ears.
The direction I would look at is using the Vocoder effect in MT. insert your voice line into the MT window. In EV, create a new file using Generate Tones at 14.5kHz of the length you need. Insert the tone file into MT ABOVE the voice file. Select and highlight them both. Still in MT, go to Effects>Vocoder. Make your Control Wave the voice file and your Process Wave the tone file. Hit OK. A new file (that looks blank) will be created. Play it in EV with the Frequency Analysis window open and you will see activity at 14.5kHz. This is amplitude modulation (AM). Will it result in anything subliminal? I doubt it, but it would depend on how much you belive subliminal messaging works.
Didn't an earlier version of Audition have some subliminal effect? Could it have been as far back as CEP 1.2?

How can I control a stepper motor drive using a DAQ card?

I need to control a simple CW/CCW stepper motor drive using a DAQ
card. I simply need to output a 5 volt signal and then a drop in
the signal, and so on 200 times or steps for 1 revolution. I have
desiigned the basic program, but do not know how to output the
appropriate signal.
Cheers,
Matt

What DAQ card do you have? Ideally, one with a counter output to generate the steps, where you can control frequency. If you try to start the motor too fast (acceleration or abrupt application of fast pulses) it may stall out. Search in Help/Search for Examples for 'frequency' and 'pulse' for example of frequency generation. You can also use a digital output on the DAQ card to control direction.
If DAQ card does not have a counter, you coul do a software timed loop to toggle a digital output to generate the pulse train. It may not be fast enough nor accurate for your application, do not know from info you have given
~~~~~~~~~~~~~~~~~~~~~~~~~~
"It’s the questions that drive us.”
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Follow Up...Vocal Mixing Approach

I've been kicking around possibilities in a different thread with some very helpful people
http://discussions.apple.com/thread.jspa?threadID=2386425&tstart=0
Now, I'd like to implement what AT THIS POINT, I think may be my best way to approach the issue of getting my vocal mixes good.
First, I am going to go back to using my mic the way I had in previous recordings. That is, using the large windshield that came with my SM7, that helped with a lot of plosives etc. Also, I will apply the bass roll off and NOT the presence boost. Presence boost seemed to give too harsh highs. I can EQ later if need be. Flat will be easier to start with. As far as pre tracking, this is about all i can do, since I know my recording levels and all are good, and i don't have any other equipment besides the SM7 and the DUET.
Next, I would like to know y'alls take on "NORMALIZE". I usually have not ever normalized my vocal tracks. The last one I had issues with a lot, I actually DID normalize, So, perhaps I WON'T in the future. I'm gonna open up a meter for monitoring. Prolly the Multimeter. I'm going to go in and for the sake of time, only do some manual edits to the volume envelope for the REAL bad guys, those that stand out clearly in my ears. Maybe do a quick check for anything that is REAL quiet too, boost that a little manually. Problem is I don't know how much to boost or lower. Can I see with the meter what the overall average db is, and see if I have a loud part that hits, say -6db and the average is around -12db, so I'd just lower that part by 6db?? and vice versa for quieter parts.
Now, the fader is controlled by the volume envelope so if I do automation I will lose control of the fader for my levels, so should I insert GAIN plug or just use an output gain on one of the inserts I'm gonna use anyways. Either way, at this point I will set the relative levels. Just a quick setting to start off with.
Now I believe I can apply this plugin
http://sounds.wa.com/audiounits.html
Thanks Ericksimon!!
Looks like a nice tool, similar to what I was looking for but not too cumbersome. I have not gotten a chance to try it out or see how to use it properly.
So at this point, I think I should have a really steady volume level going on.
Now I am not sure about the whole compression thing. I have come across several options. First though, do I want to do any EQing at this point??? before I add compression or after?? There are two points I would want to use EQ, one is to tame any out of control frequencies and 2 would be to shape the overall tone of the vocal. I suppose I could EQ for any bad frequencies first, then do compression and whatnot, then EQ at the end to slightly shape the sound if need be.
Anyways, compression. The Multipressor worries me a bit. I've used it with nice results on instruments and sampled audio, but not on vocals. Also, the downward expansion looks cool. I don't usually have any problem with too much noise, but I guess that's a problem i just never knew I had!!! I just try to track with as little gain as possible while still achieving a usable recording level. Then I get increase the gain later without having a lot of noise on the recorded track. Also, the multipressor would address my Highs in my voice jumping out at times. If I could just compress the Highs and not so much the lower frequencies. That might give me better results than JUST compression the whole spectrum equally.
The OTHER option, maybe, would be to use the DeEsser. I can compress a specific frequency range, but I don't think it has multiple bands. So, I couldn't really get any compression going on for the other bands. But I was thinking about the DeEsser in tandom with a Noise Gate instead of the Mulitpressor. This may be a little easier, but I think I might want to have a full spectrum compression, but be able to compress certain "trouble" frequencies more or less.
So I think I should be done at this point as far as levels and all that go, perhaps if I was to use EQ to color my track a bit, but that is gonna fall in with applying REverb and Delay or whatever else I want to add flavor. WHICH IS A WHOLE nother thread. After mixing about 13 tracks in logic now, I still can't find a setting that I'm comfortable with for my voice. Every recording seems different so every time i have to do something quite different. And every time i feel like it's the first time!!!
Anyways, how does my approach at this point look??
Holy crap,.....I didn't realize I had written that much. My god, If anyone replies you are truly a saint.

Bee Jay wrote:
Next, I would like to know y'alls take on "NORMALIZE".
Never do it. Absolutely pointless.
I think if it doesn't hurt anything I WILL because I think it helps me when doing manual edits so that I have a point of reference that is somewhat consistent throughout multiple audio regions
Now, the fader is controlled by the volume envelope so if I do automation I will lose control of the fader for my levels,
This is a good point - in general, send the output of a channel to a bus/aux channel. You can now automate one, and use the fader on the other to offset the levels - doesn't really matter which, although I prefer to automate the channel fader to smooth out the signal before hitting the aux wher the compressor is.
I am not real familiar with aux channels, I've tried using them but didn't understand the send amounts and how it was affecting the signal. If I put the send amount at 100 does 100% of the signal get passed to the fader then. So in your example, I would definitiely want to set at 100. As of now I simply used the gain control on the compressor or whatever plugin was last in the chain, I think that should not be any different in the sound....
The Multipressor worries me a bit. I've used it with nice results on instruments and sampled audio, but not on vocals
I wouldn't use it unless there is a definite reason to use it. At this point, you don't sound sure why you are using, and thus are unlikely to use it effectively. So don't over-complicate and use something simpler - there are plenty of good sounding compressors around, if you don't like Logic's.
well what I did is looked at the problem frequencies that I had corrective EQ'd and made that one band,,,,from say 700-1200, then made the other two bands as 0-700 and 1200-18000 or whatever. I didn't touch the low band, but I applied compression slightly different for each one of the other bands. I think it turned out much better than I expected, but I really don't know that I needed much compression anyways......the manual automation, rider plug and EQing had the signal going to the compressor pretty good already. It's amazing how much better a compressor sounds when the signal coming in isn't pure garbage!!!
I just try to track with as little gain as possible while still achieving a usable recording level.
What levels do you record at?
I believe the settings on my Duet are at about 56-58 usually when I record. This puts the input signal usually bouncing around between -24 to -12 db. I heard that was a good range and leaves lots of headroom. I use Ind. Monitoring level to crank it up so I can record at this level, otherwise it's too quiet to hear over the beat. I like to perform with lots of volume. These settings seem to work well, and NO i don't have a noise problem. Actually after alll the compression and everything was said and done. The "silent" parts between phrases or words were actually pretty much dead silent. So that seems to tell me my gainstaging is correct?
Also, the multipressor would address my Highs in my voice jumping out at times. If I could just compress the Highs and not so much the lower frequencies. That might give me better results than JUST compression the whole spectrum equally.
Maybe, but I still think it's overcomplicating things, although it's hard to say without knowing the recordings. if you have particular sections which are harsh, the common technique employed by the big guys is to split the vocal ("mult") into multiple channels for the appropriate parts, so you can EQ and treat the parts separately. Some producers will even go so far as to automate or mult individual syllables in a vocal phrase, although that is extreme.
yeah, haha, It already took something like 4 hours last night to do what I did. I broke down and manually automated nearly every phrase/syllable in the song!!! Boy, that was work. But, the results were quite apparent. Even taming the out of control stuff, it gave life to the stuff that used to get kind of lost in the mix. I been reading the manual to get some easier shortcuts and stuff for editing automation, so that should cut some time a bit. Also, I'm not gonna be recording much more backing tracks I don't think. If I do, I will probably just copy the automation from the first "Lead" track to the backing track. I have gotten a little more comfortable using reverb to help fatten up my vocals.
So I used manual automation edits. Rider Plugin. Corrective EQing. Multipressor. "Color" Eqing. And Finally SilverVerb.
Sounded much better than before. And really I only spent about an hour or 2 more than what I had originally spent battling with plugins that couldn't correct a bad input. So good in good out and you know the rest. And like I said. I accidentally erased most of the automation I did, then saved then quit. Didn't realize I did that. Oh well. I'm learning

I want to use the Function Generator VI to send command signals through the NI 7344 motion controller. This will be a closed loop servo valve system. I want to be able to change from say a square wave to a sine wave on the fly. Idea's?

I am going to run tests that require an actuator to move using various types of arbitrary waveforms such as sine or square. The NI 7344 is hooked to the UMI that is going through a driver for a servo valve. The loop is analog and it is closed. I have played with some of the examples but can't get it to work. I have used the function generator VI to generate a signal but I think I am using the wrong input VI to the motion control board. When I use what I have it moves the servo and then stops. It doesn't continually generate the signal.
I would love to use the controls on the function generator vi to control frequency and amplitude ect. Any help or pointer would be helpful. Thank you in advance.

Hello,
I'm not clear on exactly how you want to use the generated data but I'm assuming they will be used as your target points.
There's a built-in example for motion called 'One-Axis Contour Move.vi'. This example demonstrates how to provide your target points as 1-D array. All you need to do is to replace the input array with the output of the function generator. In order to have it run continuously, use a while loop. You can further program your application so that it'll change the waveform on the fly by monitoring the user interface but this might be little little tricky as you will need to reset the move and load the new generated points while keeping track of your current position.
I hope this helps. Let me know if you have further questions regarding this
application.
Best regards,
Yusuf C.
Applications Engineering
National Instruments

Mail Error messages..please help

Ok, so recently, my .mac mail account keeps giving me this error.
+Some actions taken while the account “[email protected]” was offline could not be completed online.+
+Mail has undone actions on some messages so that you can redo the actions while online. Mail has saved other messages in mailbox “On My Mac” in “On My Mac” so that you can complete the actions while online.+
+Additional information: The attempt to send data to the server “mail.mac.com” failed.+
It seems as if it is no longer receiving or sending mail, but I recently switched my smtp from port 25 to 587 to see if that helped any, but it did not. It keeps putting email I sent last night into the On My Mac portion of the drafts, even when I go through and delete them. Anyone else experience these problems? Help would be greatly appreciated.

Yep, same problem. Started a few days ago after I tried to send a video attachment. Have been looking for answers on all the forums but the solutions that I've seen wont work on my system. Seems to be related to Mail's cache but I think there is a bug in Mail that is related to Drafts. My Drafts folder is filling up with stuff that I have no control over. Autosaving of drafts is great but they dont get erased after you send the mail... Now Mail has frozen and I'm using the glacially slow web Mail... This seems to be a recent problem and so I suspect a glitch on the Mac servers is the real issue. Anyone have any answers? (short of going into my system and hacking stuff via console .... please there must be a better way!!!)

CANOpen Communication NI cRIO 9022 (9853 CAN module) - MAXON EPOS2

I'm implementing a velocity control loop on two Maxon EPOS2 70/10 control units, interfacing them with a NI cRIO 9022 (CAN module 9853). To this aim, I'm using the libraries from NI and they work quite well: for each EPOS2 I can read current, position and velocity using 2 PDO messages and set a desired velocity using 1 PDO message, all in a single timed loop on Labview Real-Time. My VI is based on a NI example included in the mentioned library. Even though the sampling time of the loop is set to 10 ms (as in the example file), I verified that all the operations were executed in about 30 ms.
My target is 1 ms sample time and I'd like to know if there is any chance to get this; are there some limitations in the software/communication protocol I'm using or is it possible to achieve 1 kHz control frequency? Based on the documentation, the EPOS2 CANopen driver is split as follows:
1.      Hardware Layer: VIs which interact with the CAN hardware (FPGA Interface folder).
2.      CAN Layer: VIs which construct/deconstruct CAN packets and handle communication between the Hardware Layer and the top level application (CAN folder).
3.      CANOpen Layer: Low level VIs which simplify the generation of the SDOs and PDOs used to communicate over the CAN bus (CANopen folder).
4.      EPOS2 Driver Layer: High level wrappers over the CANOpen Layer simplifying the control of EPOS2 drives (main folder).
It seems that the Layer that construct/deconstruct CAN packets is impelemented on the RT level. Is it correct? Do I need to move it to the FPGA level to improve performances?
I look forward to hearing any suggestions from you.
Best,
Nevio

Hi Nevio,
I am trying to simply read encoder values from my EPOS2 24/2 control units into Lavbiew. I have been through all the documentation however I am unable to establish communication between the EPOS2 and labview.
I am connecting the two using a USB connection. I can connect to and control the motors in EPOS Studio, but nothing in labview.
I am very new to Labview so I am probably not putting the correct blocks down for initialization.
Could you please help me by suggesting some good examples to look at or linking me to labview sample of initializing communication with the EPOS2 units.
Any help is greatly appreciated
Regards,
Kris

Two channels needed with pulse delay. Do you know of a VI for this?

I want two channels, one with a given number of pulses of controlable length. On a second channel I want a controlable frequency during time the first channel pulse. I want to control the time after the pulse on the first channel goes high that the frequency on the second channel goes on.

I'm a little confused as to exactly what you want. First, I'm going to assume that you are talking about TTL level signals, so the counters will be used. I'm also assuming that you have a DAQ card available to you. You can control output pulses on one counter, and set up a frequency on the other counter, but I'm unclear as to the relationship between the two.
Are my assumptions correct, and what is the relationship (in clearer terms)?
Mark

Motion 2.1 Problems

Hi:
I am having problems with Motion software
When I am working on it for example working with the particles system the program just close without asking me to save, it just close and a popup window say that it has been a problem or error and it ask me if a want to report it to apple.
If any body can help me with this...
Thanks!

Probably not what you want to hear, but Motion is buggy. Just make sure you save often. Increasing the frequency of autosaves in preferences also helps.
The next time it happens, post the crash report here (but include only the thread that crashed), and we can see if we can suss it out.

I want two channels, one with a giv

en number of pulses of controlable length. On a second channel I want a controlable frequency during time the first channel pulse. I want to control the time after the pulse on the first channel goes high that the frequency on the second channel goes on.

en number of pulses of controlable length. On a second channel I want a controlable frequency during time the first channel pulse. I want to control the time after the pulse on the first channel goes high that the frequency on the second channel goes on.You can use counters to generate your pulse trains. The E Series Multifunction IO boards have two DAQ-STC counters. The 660x Counter/Timer boards have at least four NI-TIO counters. The following applies to either of these type counters.
Programmatically, you can set up the pulse width or frequency on the fly. There is an example called Pulse Width Modulator on the National Instruments website that shows how one can change knobs on the user interface to affect the pulse train parameters to illustrate the idea of changing the counter settings programmatically.
You can find the example by going to the http://www.ni.com/support page and choosing Example Programs under the Technical Resources pull-down menu. Then, type in keywords to search for, such as "pulse width
and frequency".
Regards,
Geneva L.
Applications Engineering
National Instruments
http://www.ni.com/ask

FCP said it couldn't save a project, then crashed, all changes lost.???

What was I doing? Nothing unusual.I'd brought in some files, had removed some others, all working fine when played in the canvas.
It is kind of driving me crazy.

Hi -
Open your project in FCP, go to the menu File > Restore Project. A window will open that will allow you to select the last (or earlier, if you wish) autosaved copy of you project. You will lose the work you have done since the last autosave.
You can set the autosave duration by going to the menu Final Cut Pro > User Preferences. On the General pane you can set the frequency of autosaves in the lower left corner.
I keep mine set to every two minutes, so if I have a crash, I never lose more than two minutes of work.
Hope this helps.
Mtd

Control frequency of autosave

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