Triggered dig. input, 1.st sample delay?

I find the "Start & Stop Trig Input.vi" useful, and want to use it in an application where I sample the state of five photo detectors. When a vehicle passes a detector, I want to time stamp the sample. I collect the samples in an array, and search the array after the acquisition ends. Sample index devided with sample rate is then the time. The problem is that I get a varying time offset from the start trigger (sample 0) to my time stamps. If I wire a bit directly to the trigger pin, I get a time difference from about 10ms to several hundred ms between the trigger and that bit. My DAQ is a DIO-32HS.
Gunnar

Hi Gunnar,
Let me make sure I understand you correctly. You are trying to perform pattern input at a specific rate and you want to be able to timestamp your inputs. You notice that when you apply a rising edge signal to both, the trigger and a bit, there is an acquisition difference of 10ms-100ms. What type of trigger are you using? Is it a pattern match? What is your sampling rate?
Are you using the control "pts after stop trigger"? Does your delay correspond to this number of points divided by your frequency? What I would try using is a simple "Buffered Pattern Input-Trig.vi" from the shipping examples with LabVIEW and connect your trigger to one of the bit lines. I've tested it on my PCI-6534 and I see my signal as the first data point in my arra
y. Anyway, let me know what specifics you are dealing with and I might be able to help you out. Have a good day.
Ron
Applications Engineer
National Instruments

Similar Messages

Sample Delay VI

Currently i'm using Labview 7.1 embedded together with speedy-33.
i had my audio input to speedy-33 from my PC sound card and i wish to delay the time of my signal by let say 1sec.
I had my left signal connect to the sample delay vi then from sample delay vi to audio output.
Audio in (only one channel,left)--> sample delay --> audio output
i should heard a delay of 1sec when a audio signal i played but there is no delay.
i use two waveform graph to observe the signal before and after the sample delay vi and indeed there is a delay as the waveform are different.

The file attach has a input sin wave connected to a sample delay vi and then to a analog output.
Waveform graph were ploted before and after the delay for comparsion.
Thanks for the help.
Attachments:
Sin sample.vi ‏36 KB

Avoiding phase cancellation with sample delay plug in

Hi
I had soem horns in a mix sounded flat so i used the sample delay plug in to slightly delay the right side then they sounded great , really big and sat in the mix but when witching to mono they nearly dissapear.
Is there any trix to avoid this phase cancellation when swithching to mono?

Data Stream Studio wrote:
1.-I think if you reverse polarity on 1 of the channels of your mix instead of just reversing polarity of one side of the problematic horns, you may well solve one phase issue and create a whole lot of new ones.
2.-This is a good idea, maybe there's a setting that sound as fat as the one you're using but will not phase cancel in mono.
3.-I'm not sure what audio file you'd move... could you explain?
If it's a Stereo file, split it into two mono tracks, and move on of the two up/down the timeline by a few samples, until you get the least phase combing effect.
Maybe the OP needs to go back and re-mix that song, and find the culprit horn out of the whole section... if he can...

Sample Delay - samples into millisecond delay time?

Hi guys
Hopefully a simple question and I'd love some info.
I'm mixing in 5.1 and want to simply put up a general 25 millisecond delay in the left rear channel and 35 millisecond in the right rear channel. Some room calculations for what I recorded require 25 - not 20 and 35 not 30 milliseconds...... the standard stereo Delay won't allow increments of 5 miliseconds.
I'm therefore thinking of using the SAMPLE DELAY ........ whilst I have a 24 bit 48 khz recording setting ....... is it possible that within the maximum 10000 samples I can get up to 35 milliseconds ....?
How do we make the calculation work ..... the manual says 10 centimetres mic distance is 13 samples at 44k
is it possible to get up to 35 milliseconds using this?
Many thanks
Dick

My dear Christian ....... that is just the quickest most wonderful solution to question I've ever had ..... thank you for that PERFECT answer ....... I'm a happy man!
All the very best
Dick

"fine/sample delay", mixing down live drums, esp for JustinC!

I'm curious about the "fine/sample delay" and "phase inversion" techniques briefly mentioned by JustinC in an older post regarding recording live drums. I understand using phase inversion on snare mics that are pointing up/down, but I'm not 100% sure about the fine delay adjustment. Is this actually shifting the audio to line up waveforms from mics that are different distances from the same source? I've done this with an upright bass before (F-hole mic and another a few feet away) and it seemed to get a much fatter, clearer sound. I really appreciate any tips you might have on how to pinpoint and solve the problems.

hey over-man,
I was prowling tonight and ran across your post. I don't know your reference specifically but I understand your question.
Sample delay regarding a multimic'ed source:
This is based on phase cancellation and reinforcement. It is initiated by everything in a multimic'ed environment. The times, distances, microphone choice, effects choice-it's everywhere.
The best approach for me (typically) is to get the most appropriate sound earlier on unless you can expect variables ahead of time and factor them in right away. We'll work with an example of simply getting a natural drum sound that will line up with existing tracks (bass, gtr, vox...).
1) You can set up the sound/mics as close as possible to fit the sound you're after (we'll breeze over these points not specific to your question).
2) Before destroying the sound with a handful of plugins adjust delays. Some assumptions can be made:
A: Overheads/Room Mics will typically be the reference for the reason that they are what you have least control over because they represent more sources (Drums in this case).
B: These Mics will often have the largest delay from the start for the reason that they are farthest from the source.
C: These Mics should be adjusted less than others, I would EQ them quite minimally. Solve any problems as early in the signal chain as possible and use only what you know you'll need. Do not apply FX you don't need and position mics to better suit your desired sound.
So your OHs/Rooms are as good as they can sound naturally and sit with the track as well as possible, you can now start adjusting delays/intrtoducing other elements.
The order can vary here but I tend to start with the tracks/mics which will be more prominent. I will use snare in this example.
Some useful tools before we resume: Logic's sample delay, Expert Sleepers' Latency Fixer, because Logic's Sample Delay will not predelay. Found here:
http://www.collective.co.uk/expertsleepers/latencyfixer.html
I would normally just load both of these up, I use full PDC and apply these AUs to each track-remember to turn off Software Monitoring!
Create an instance of Latency Fixer, set it to 2048 samples. You have now added 2048 samples of predelay to your track.
Create an instance of Sample Delay, set it to 2048 samples. You have now adjusted your track to play back at the original record position.
Copy both instances to each adjacent track.
So we now have Latency Fixer followed by Sample Delay on each track for the kit (not necessary for the OHs unless you want to shift everything a little, but I will often do this from the arrange by ms).
At this point you will not need to adjust Latency Fixer's settings and simply use sample delay to offset delay in either direction.
Bring the snare track up to a reasonable level (the closer you are the better). Adjust Sample Delay in either direction and you will notice the sound changes from phase cancellation/reinforcement. Certain frequencies will stand out more at different settings. It normally won't take much, we don't want it to sound like a delay. Someplace within +/- 300 samples is often best for a natural sound. It's a good idea to have a phase inverter (Gain PlugIn) handy for 2 reasons: It may sound better inverted. You can invert the phase to pronounce things you DON'T want to hear and then flip it around the right way after setting the delay to pronounce them. Volume has a huge bearing on interaction here, a good delay setting may sound bad after reducing the level 2dB so Sample Delay setting, volume and phase should all be updated as soon as possible.
By now you have noticed how adjusting the delay affects the sound, inverting the phase is an exaggerated example of this principle. Inverting the phase of two identical tracks cancels them entirely, applying fine delay to one will introduce some of the frequencies and leave some cancelled. You can hear it scan through which frequencies by adjusting the delay. With multiple sources, different mics and different distances these delays already exist. Delaying them so they match (based on distance) is not always the best sounding but try to start somplace close. After setting the delay and Volume as close as possible you should be able to get away with much less surgical types of PluggIng and ultimately have more natural sounding tracks. Will often comprise 75% of the sound (if you want natural tracks).
If there are dynamic effects you know you'll want to apply there are special considerations.
Many of Logic's "lookahead" paramaters delay the signal.
Compression/Gating, etc. (on a per track basis) will alter the volume significantly enough that if you know you want a track compressed... ahead of time you should probably set these as you would then adjust the delay.
I tend to use predelayed tracks as keys for the audible tracks, this allows you to get around the aforementioned lookahead problems and will provide you with more control over the "shape" or envelope.
Concerning phase inversion based on which head you mic: It is a generalized solution to mic setup, though not a hard and fast rule which applies in every case.
Delaying in this manner is aligning waveforms. Based on frequency you are causing some frequencies to stand out more or less since we are working with complex waveforms. Very small deviation from original position will affect higher frequencies most and as you get further away from the original postion it begins to affect mids and eventually low frequencies. Of course this is because the frequency/wavelength. Align and misalign the waveforms for your desired sound. It does require you listen very closely when setting it up but I really think the results are worth the effort. Hopefully you'll find this helpful, it's easier to hear the effect than to describe it, hopefully it is apparent once you hear it.
Finally, I don't have any Platinums. If it doesn't work better for you, don't believe it needs to. Hopefully others will chime in here so you can get some second opinions on the approach and/or general drum mixing techniques.
Cheers, J

Sample Delay

Hi,
with using ProTools there is a nice function that allows you to adjust the sample delay due to adding several plug-ins to a track.
this way you can compensate the physical micro-dealy created by the
poor/stressed DSP by processing that amount of data at the same time..
now, where can i find the same fuction here in LE?
how is ti called?
thanx

You're talking about plug-in delay compensation. It's built right into Logic (Express and Pro). You can switch its behaviour to act on All tracks, Audio Tracks and Instruments, or None in the Audio preferences (I'm not in front of Logic at the moment...can't remember the exact path).
You'll want to leave it on "Audio Tracks and Instruments" when you're tracking, and "All" when mixing. Other than that, it's automatic. No adjustments required.

Sample Delay bug

Does has anyone else noticed that Sample Delay doesn't remember its setting when a project is closed ? Or is this just a problem with my system ?
1) Insert a Sample Delay plug-in on any track and set the delay value to anything other than zero
2) Save your project, close it, re-open it
3) Check the delay value--it's reset to zero
Happens every time I close the project. Bummer.
James
[email protected]

To get rid of the playing through the channel strip thing, control-click (or right click) - it may be a long click, I don't remember - on the speaker/play icon in the sample editor and you have the option to listen through the pre-listen channel.

Triggered Analog Input

I want to trigger an analog input with a pulse generate
by gating counter 0 with the Analog Output UPDATE* signal.
any suggestion?
Thanks,
Erin

Response-For what it is worth this isn't a very straight forward thing to
achieve with LabVIEW VIs. I have been trying to do the same thing for a
month now with no luck. If it is any consolation, NI tech support hasn't
provided any working solution to this either. If I get it figured out in
a few days I'll try to remember to write back.
Try using the analog scan clock to trigger the analog read instead of the
traditional "triggering" schemes in the LabVIEW documentation. Their terminology
is misleading. Per LabVIEW the term "trigger" relates to the oscilloscope
concept where you specify voltage level and edge-transition to recognize
a waveform in order to high-speed sample the incoming waveform.
In the embedded system the term "trigger" refers a specifi
c point in time
(defined as an "event") usually represented by a signal transition (rising
or falling edge) at which you want something done such as read an analog
input or save a counter value. In microcontrollers this is generally handled
with a specialized hardware port called an input capture which is designed
to automatically recognized signal transitions and simultaneously fire off
an interrupt (read as "trigger") that can be used to start your unique action.
LabVIEW documentation doesn't seem to have a clue about this concept. Perhaps
you know all this already. Hope it helped in some manner anyway.
Frank
"Erin Ryan" wrote:
>>I want to trigger an analog input with a pulse generate>by gating counter
0 with the Analog Output UPDATE* signal.>>any suggestion?>>Thanks,>Erin

DATA_CHANGED event handler is not triggered after input.(OO ALV)

Dear Experts,
What I did:
I register Enter as the trigger event for data_changed event, and I put my checking logic in data_changed_handler which is a method of a local class. When the checking fails, I put messages using er_data_changed->add_protocal_entry to show messages.
My checking is simple, if the cell is empty, error messages pops up tells that this cell can not be blank for this row.
My problem:
When there is no entry in the alv list(Only this situation), I insert a new entry, the cell which is required is empty, I press Enter, data_changed event is triggered, the message pops up saying that the cell is required. Then close the pop up message window, I input something in the cell, and press enter. The data_changed event is not triggered.
Is there any1 has any ideas on this problem?
Vicnent
Edited by: Changjiu Tan on Sep 7, 2009 1:33 PM

METHOD HANDLE_DATA_CHANGED.                               "#EC NEEDED
*<<< BEGIN MODIFY EXAMPLE CODE
*   u4FDDu5B58u30C7u30FCu30BFu3092u7DE8u96C6u3059u308Bu3000START
    DATA: LW_INS      TYPE LVC_S_MOCE,
          LV_FNAME    TYPE LVC_FNAME,
          LW_MOD      TYPE LVC_S_MODI,
          LV_MAX      TYPE I,
          LV_RES(12) TYPE C,
          LW_LIST     TYPE ZTSRE012,
          LV_ROWID    TYPE LVC_S_ROID,
          LT_MOD      TYPE LVC_T_MODI,
          LV_COLID    TYPE LVC_S_COL,
          LV_ROW_ID    TYPE LVC_S_ROW,
          LV_COL_ID    TYPE LVC_S_COL.
*   u73FEu5728u306Eu30D5u30A9u30FCu30ABu30B9u4F4Du7F6Eu3092u53D6u5F97
    CALL METHOD GO_ALV_GRID7->GET_CURRENT_CELL
      IMPORTING
        ES_ROW_ID = LV_ROW_ID
        ES_COL_ID = LV_COL_ID.
    LOOP AT GT_LIST INTO LW_LIST.
      IF LW_LIST-ID > LV_MAX.
        LV_MAX = LW_LIST-ID.
      ENDIF.
    ENDLOOP.
*   u30B0u30EBu30FCu30D7ID
    LV_FNAME = CNS_ID.
    LOOP AT ER_DATA_CHANGED->MT_INSERTED_ROWS INTO LW_INS.
      LV_MAX = LV_MAX + 1.
      CALL METHOD ER_DATA_CHANGED->MODIFY_CELL
        EXPORTING
          I_ROW_ID    = LW_INS-ROW_ID
          I_FIELDNAME = LV_FNAME
          I_VALUE     = LV_MAX.
    ENDLOOP.
    CLEAR: LV_FNAME.
    LT_MOD[] = ER_DATA_CHANGED->MT_GOOD_CELLS[].
*   u30C7u30FCu30BFu30C1u30A7u30C3u30AF
    LOOP AT LT_MOD INTO LW_MOD.
      IF LW_MOD-FIELDNAME = CNS_ZZTTSN.
        LV_COLID-FIELDNAME = LW_MOD-FIELDNAME.
        LV_ROWID-ROW_ID    = LW_MOD-ROW_ID.
        CALL METHOD GO_ALV_GRID7->SET_CURRENT_CELL_VIA_ID
          EXPORTING
            IS_COLUMN_ID = LV_COLID
            IS_ROW_NO    = LV_ROWID.
        CALL METHOD GO_ALV_GRID7->GET_CURRENT_CELL
          IMPORTING
            E_VALUE = LV_RES.
        IF LV_RES IS INITIAL.
          CALL METHOD ER_DATA_CHANGED->ADD_PROTOCOL_ENTRY
            EXPORTING
              I_MSGID     = CNS_MSGID
              I_MSGTY     = CNS_MSGTY
              I_MSGNO     = CNS_MSGNO
              I_FIELDNAME = LW_MOD-FIELDNAME
              I_ROW_ID    = LW_MOD-ROW_ID.
        ENDIF.
        CALL METHOD ER_DATA_CHANGED->MODIFY_CELL
          EXPORTING
            I_ROW_ID    = LW_MOD-ROW_ID
            I_FIELDNAME = LW_MOD-FIELDNAME
            I_VALUE     = LV_RES.
       ENDIF.
      CLEAR: LV_RES,
              LV_COLID,
              LV_ROWID.
    ENDLOOP.
    CALL METHOD GO_ALV_GRID7->SET_CURRENT_CELL_VIA_ID
      EXPORTING
        IS_ROW_ID    = LV_ROW_ID
        IS_COLUMN_ID = LV_COL_ID.
ENDMETHOD.                    "handle_data_changed
Edited by: Changjiu Tan on Sep 7, 2009 1:31 PM
Edited by: Changjiu Tan on Sep 7, 2009 1:32 PM

Disallow non digit input in a text field?

Is there a class method in Java SDK to do this easily like in Visual Basic or has it to be coded completely manually?
If the user presses a non digit character then that character should not be printed into the text field.

K this is easy!
I have a class:
import java.awt.event.*;
public class AlphaNumConsumer extends KeyAdapter
//this class is used to consume any alpha letters
     public void keyTyped(KeyEvent ent)
          char ch = ent.getKeyChar();
          if('0' < ch && ch > '9')
               ent.consume();
and all you do when you declare the textfield anywhere else you do the following:
JTextField field = new JTextField();
AlphaNumConsumer kCon = new AlphaNumConsumer();
field.addKeyListener(kCon);
this will consume and Alpha numeric characters!
Gluck!!

Sample delay from software to hardware

Hi,
I am interested to find out the time it takes for a sample generated in LabView until it hits "the air" (channel output) of one of your DAC cards? I couldn't find any data on this in the different product specifications nor in the forums.
Thank you very much,
Thomas Schmmid

Hello tschmid. The time that a value takes to appear at the analog output of a DAQ board can be divided into 2 distinct parts. The first part is the time it takes to get from the software to the buffer on the DAQ board. The second part is the time it takes to progress through the buffer on the DAQ board to the output. The first part is nondeterministic since there is no way to benchmark this process, but should be very quick since all the software does is send the output value to the board. The second part can be found in the user's manual of a specific DAQ device under "Maximum update rate" for analog output channels. For example, assuming one channel is being used, the PCI-6251 has a maximum update rate of 2.8 MS/second. Thus, if you have a clear buffer, it will take this second part 1/2.8M seconds to execute. If the buffer contains 99 values before your desired output, the second part will take 1*100/2.8M seconds since 100 values have to be written to the output in order to see that value that was just written to the board from software. There is a way to ensure that your onboard memory doesn't slow down this process too much. There is a property node in DAQmx called AO.DataXferReqCond which allows you to chose one of the three following options: "On Board Memory Empty," "On Board Memory Half Full or Less," or "On Board Memory Less than Full." Choosing the "On Board Memory Empty" option will allow a more responsive output assuring that the current value written from software will be seen on the board's output quickly, but it will limit the achievable update rate. However, if you are concerned with seeing one value appear on the board in the shortest amount of time, "On Board Memory Empty" is your best option. I hope this helps and if you have any additional questions, I would be happy to address them. Have a great day!
Brian F
Applications Engineer
National Instruments

Variable Length Digit Input CVP

So basically I want to make something that says.
"if you have your account number please enter it now or if you don't have it press the * to continue".
Same thing on some of my menu's
"Welcome to blah blah, for 1 press 1, for 2 press 2, or if you know your parties 6 digit extension please Enter it now". What is the best way to accomplish this with CVP?
Chad

Chad, here's my 2 cents.
Resist the tempatation to do it that way. You don't get the automatic length checking and all the other good things that come from entering fixed length strings; and you can make CVP handoff to Unity handlers that do functions like dial by extension much better.
A lot of that crap came from a variety of proprietary PBX and voice mail systems. "Star" on one system means "continue" and means something else on another system.
Try to standardize along the lines of VXML.
Provide clear and unambiguous menus that have less than 5 options. Avoid the "... or stay on the line for customer service" option.
Something like ...
"If you know your 8 digit account number, press 1.
Otherwise, press 2".
If they choose 1, you can have a fixed length "get digits". Self checking and all that good stuff.
And then in another menu of yours ...
"For the auto-attendant, press 1.
For xx, press 2.
For yy, press 3."
Then in auto-attendant ...
"If you know your party's extension, press 1 (leads to a Unity dial by extension handler)
To dial by name, press 2.
To check your voice mail, press 3.
(2 leads to to the Unity dial by name handler. 3 leads to the Unity sign-in handler).
With type-ahead (a natural feature of the VXML standard) regular users will type 1-3 to check their voice mail.
I'd like to hear from other CVP designers. It's not that they can't be coded.
Regards,
Geoff

How to use counter output pulses to trigger analog input?

Hello all,
I hope the kind people using this forum can help me, a lowly beginner LV programmer! I have been attempting to create a VI that produces a user defined number of TTL pulses, separated by every n seconds. Each TTL would be outputted to a stimulator, which in turn generates its own TTL. Using the stimulator-generated TTL, I would like to trigger finite analog data acquisition (e.g. for every TTL, trigger the collection of a data sweep that contains 4000 samples (collected at 4000 Hz), with 1000 samples collected pre-trigger. I would like to also be able to see each data sweep as it is triggered on a chart. As I understand things (lots of online/book/forum reading), I should be using the counter output to generate my TTL pulses, and syncing each counter produced TTL with analog input, as well as using a reference trigger. Also, the AI part should be started first, so that I don' t miss any counter outputs. If it matters, I also need to use one of the AI channels to acquire the TTL, so I can see my stimulator-induced responses to the stimulator in time.
I am able to generate the TTL pulses from the counter output, but I am having a problem with the AI part. I am unsure how to sync the counter output with AI. Also, since I need to acquire pre-trigger samples, I would be needing to acquire samples continuously, but when I set 'continuous samples' on daqmx timing, the VI doesn't work (hence why's its set to 'finite samples').
I hope someone out there can help, as I have been at this for what seems ages, with limited success. I am using a USB-6259 and LabView v8.2. Thanks!
Attachments:
RC001 v_1.vi ‏49 KB

Hello,
Due to the fact that analog tasks themselves are not retriggerable, a
pulse train produced by a counter is always used as the sample clock
for the analog input task in order to recreate a retriggerable effect
for analog input. This can be done by creating a finite pulse train set
to retriggerable using the DAQmx Trigger Property Node, or the pulse
train could be continuous and just be gated by another signal. Neither
of these methods can be properly applied in hardware to create a
retriggerable reference trigger. You can however implement something
similar in software by just stopping and restarting your reference
triggered analog input task within a loop. There will be some delay
between when the task is stopped and restarted, as these events require
software intervention, but if there is enough time between when each
trigger signal is generated, there should not be any noticeable delay
or missed samples.
I have attached an example of this!
Mark B
===If this fixes your problem, mark as solution!===
Attachments:
RC001 v_1mod.vi ‏25 KB

How do you find the frequency of a small sample size and a square wave.

I am collecting data from ten 12v pulses that are seperated pretty uniformly. The pulses make a square wave. I need to know how to input this small sample size and find it frequency. I need this frequency to be output in a readable file. Im not very knowledgeable in labview so if you try and help me please be very descriptive.
I know i need some sort of trigger because of uknown wait times before the sensor will be triggored but other than that im pretty lost. Any help would be appreciated.

What have you tried so far? What parts are giving you trouble? Please be as specific as possible. Post the code you have written.
You ask about frequency measurement and then you are talking about triggering.
If you always have 10 pulses, just measure the time from the beginning of the first pulse to the beginning of the last pulse. The frequency is 10/time.
Break down the problem to smaller parts. Acquire the data. Measure the frequency. Save the result to a file. Get each part working separately, then put them together.
Lynn
Sorry about duplicate post. My internet connection is very slow today.
Message Edited by johnsold on 05-31-2010 12:05 PM

PXI-6682 + multiple Serie S & M cards cause timestamp delay

I'll try to give as much info as possible and hopefuly someone will have an answer to this problem.
Language: C++
Introduction:
1- I have (1) PXI-6682 (Timing card), (6) PXI-6143 (Serie S (8 analog inputs each)), and (2) PXI-6224 (serie M (8 analog inputs each))
2- Every card is a task (so that's 8 tasks
3- Every card is in sync (they give data equal...more or less... to one another
PROBLEM:
1- Timestamp is off by up to 200 ms (not drifting...simply off by a set number of ms)
2- Every call to niSync_ReadMultipleTriggerTimeStamp gives me a timestamp off by that much
I don't have any info on how the tester managed to know it was 200 ms off, my theory is he tested it with some older equipment to validate the new one

Hello there,
Kindly help me with DAQmx Time stamp related problem as described below.
Setup : PXI 6133 Qty 4 (Slot 3, 4, 5, 6)
PXI 6255 Qty 1 (Slot 2)
PXI 6255 doing a Continuous Acquisition at 1 mSec Sampling and 1000 samples/iteration.
PXI 6133 doing a Ref Triggered Acquisition at 1 uSec Sampling interval and Pre and Post Trigger samples of 100000 Samples each.
Test : PXI 6133 - Reference Trigger Configured on PFI 1 line of First PXI 6133 card.
input Signal given to AI 0 OF First PXI 6133 card.
All PXI 6133 cards are included in a single task.
PXI 6255 - Input given to Channel 0 AI
External Function generator - One Digital Trigger generated of width 50 mSec High Interval.
Another signal generated for input to above cards. (Square wave of 10 mSec Period - 5 periods generated on the rising edge of trigger)
So It means When I acquire square wave in my waveform, I can consider rising edge of square wave as a rising edge of trigger (as both signals are generated aligned from external function generator)
I have attached Screenshots and programs I have used for that.
Problem: As both singal are generated at a same event, Both 6133 and 6255 should acquire that singnal at a same timestamp.
I have also read related documents about timestamp but I could not succeeded to find any link in this.
I am getting square wave 100 msec earlier in 6255 then of 6133.
If I used only single card of 6133 then that inaccuracy of 100 msec is shifted to 25 msec.
Please help me as soon as possible.
Attachments:
Timestamp mismatch between 6133 and 6255.docx ‏377 KB
Timestamp mismatch in 6133 & 6255.zip ‏70 KB

Triggered dig. input, 1.st sample delay?

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