Similar Questions

• High sampling rate for 18 strings in a single task - DAQmx

  Hello

  My current experiences require the acquisition of 18 channels (6 HAVE custom voltage with excitement, voltage AI 12) data at a sampling frequency of 50 kHz. My question is if it is faesible using a single cDAQ-9172 chassis. Used modules are 2 x NI 9237 and 4 x NI 9215.

  The previous researcher on this project used 2 different Renault to do this, but I was hoping that I could reduce it to a single DAQ to simplify the synchronization of all channels.

  I wrote a *.vi (attached) to do this and write to a TDMS file using a structure of producer-consumer; However, when I run the * .vi, I can run for minutes, but the number of samples recorded for the PDM is rarely more than 100 k. Subsequently a buffer size error (attempted to read from the samples that have been overwritten) with less than 200 k samples ever record. I checked the max sampling rate (using a property timing node) with the configured task as in the * .vi and it shows a maximum of 235kHz.

  I can't say if I make just the structure of the producer consumer incorrectly or if I ask too much of the cDAQ-9172 unique?

  Any help would be much appreciated.

  See you soon

  Bart

  Hi bart.s,

  TiTou speaks sampling aggregate not simultaneous, so sampling rate 50kS/s is the same for all channels.

  I see the problem in the loop of the producer. If your sampling frequency is 50kS/s/ch and you read that a single sample/ch you will lose data because the producer loop cannot run so fast. You should read more than one sample. I recommend you also to move your tracing to the consumption loop code and work with larger amounts of data.

  The second problem may be with the error handling in your loop of consumer. Merge the mistakes of loop of consumer and producer and also add a few general for two loops off if the error occurs (for example, using local variable).

  Best regards

  CaravagGIO

• How to save data quickly to the high sampling frequency

  Hi all

  I use a structure of producers and customers to measure and record the data. The sampling rate must be as high as 10 kHz. Given that so much data, it takes a long time before the data is saved. At first, I saved the data in an excel sheet. Then I tried to save it in binary, but it still takes a while to complete save. How can I make the time savings a short circuit?

  Thank you

  The best

  First we will make some corrections to your DAQmx code.  Since you are using continuous sampling, do NOT connect the samples per channel.  Which is actually limiting your buffer.  And there is really no need to define your buffer size either.  It is default to very big, so this isn't a problem as long as you read your data quite often.

  Now your data connection... You simply create a very wide range while acquiring data.  Then, you save the data.  It's actually not through the advantage of using the producer and the consumer.  You should save your data in the loop of the consumer.  It will be elinate need a lot of memory and you save the data to the file while you are buying.

  But, in this case, I say that the producer and the consumer is not even necessary.  Use the DAQmx Configure registration VI.  With this VI, you can tell DAQmx to disseminate all data directly in a PDM file.  You don't have to do anything.  It is by far the best way to save on your DAQ data.

  

• Real-time display at the high frequency of data acquisition with continuous recording

  Hi all

  I encountered a problem and you need help.

  I collect tensions and corresponding currents via a card PCI-6221. While acquiriing data, I would like to see the values on a XY graph, so that I can also check current vs only voltage/current / time. In addition, data should be recorded on the acquisition.

  First, I create hannels to analog input with the Virutal DAQmx channel create, then I set the sampling frequency and the mode and begin the tasks. The DAQmx.Read is placed in a while loop. Because of the high noise to signal, I want to average for example every 200 points of the current and acquired for this draw versus the average acquisition time or average voltage. The recording of the data should also appear in the while loop.

  The first thing, I thought, was to run in continuous Mode data acquisition and utilization for example 10 k s/s sampling frequency. The DAQmx.Read is set to 1 D Wfm N Chan N Samp (there are 4 channels in total) and the number of samples per channel for example is 1000 to avoid the errors/subscribe for more of the buffer. Each of these packages of 1000 samples should be separatet (I use Index Array at the moment). After gaining separate waveforms out of table 1 d of waveforms, I extracted the value of Y to get items of waveform. The error that results must then be treated to get average values.

  But how to get these averages without delaying my code?

  My idea/concern is this: I've read 1000 samples after about 0.1 s. These then are divded into single waveforms, time information are subtracted, a sort of loop to sprawl is used (I don't know how this exactly), the data are transferred to a XY Chart and saved to a .dat file. After all that's happened (I hope I understood correctly the flow of data within a while loop), the code in the while loop again then 1000 samples read and are processed.

  But if the treatment was too long the DAQmx.Read runs too late and cycle to cycle, reading buffer behind the generation of data on the card PCI-6221.

  This concern is reasonable? And how can I get around this? Does anyone know a way to average and save the data?

  I mean, the first thing that I would consider increasing the number of samples per channel, but this also increases the duration of the data processing.

  The other question is on the calendar. If I understand correctly, the timestamp is generated once when the task starts (with the DAQmxStartTask) and the time difference betweeen the datapoints is then computed by 1 divded by the sampling frequency. However, if the treatment takes considerable time, how can I make sure, that this error does not accumulate?

  I'm sorry for the long plain text!

  You can find my attached example-vi(only to show roughly what I was thinking, I know there are two averaging-functions and the rate are not correctly set now).

  Best wishes and thank you in advance,

  MR. KSE

  PS: I should add: imagine the acquisition of data running on a really old and slow PC, for example a Pentium III.

  PPS: I do not know why, but I can't reach my vi...

  
  

• High sampling rate is not correct

  Dear alls,

  Anyone know why the time loop is not accurate for the period defined in the Trace Viewer in real time? In figure VI, I put the period for the time loop is 250 US, but when I analyze in real time Trace Viewer, it the true period time about 676 shows us. This means that my data is sampled in the incorrect sampling frequency.

  Could you give me some explane it please? And how can I solve this problem? I tested this code with MyRIO Council.

  Best regards

  Kien

  Why do you have a waiting in your timed loop?  The point of the timed loop is the percentage rate of the loop.

  You can post your real VI?  I suspect something is happening here that we cannot see (configurations, channels, etc.).

• Audition 3.0 how to disable ASIO and the default Sample Rate recording?

  Hi people,

  New here, but I hope someone can help me with a few questions, I'll have with Audition 3.

  Firstly, some background questions.

  I use hearing parallel to a broadcast audio broadcast program called SpotOn,

  This software requires that I run the sound card, a RME Madiface XT in 48 k mode, and that all the outputs that it uses are defined as WDM Windows so that the windows kernel mixer can control them.

  This means that when I use the hearing at the same time, I have to configure it to use the "Audition 3.0 Windows Audio" driver to stop him from taking control of the sound card directly and change setting which prevent SpotOn to see its output.

  The problems I encounter are that hearing itself seems to randomly change mode in the edit window ASIO driver, I suspect that this happens when I import audio data from a key which is from 44.1 to modify for use in SpotOn. This often seems to not only make the outputs of the card its invisible to the windows kernel mixer but also change the sampling frequency of 44.1 sound card and stops work SpotOn.

  The second question I have is that the sampling frequency of default record when I record in edit mode is always 44.1 and if used it again change the map sound 44.1 and causes the same problems, I'd be very keen to know how to change this default to be 48 k if possible.

  Then.

  What I ultimatly looking is...

  1. a way to disable the ASIO drivers in hearing so that it is not only this option is available, and cannot use the Audition 3.0 Windows Sound Drivers.

  2. a way to make the sampling rate 48 k to stop people choosing 44.1 mistakenly when saving default record.

  Any help or advice that anyone can give would be much appreciated.

  Thanks in advance for your comments

  What other pilots ASIO sees your installation of 3 AA? If it's the Madiface one and you use only WDM drivers you can just uninstall the ASIO RME driver?

• Lose the analog input samples to the high recovery rate?

  
  

  I got it.  I changed to a life-long.

  I then changed code so that I was counting the samples collected in the software and stops when I had them all.  Previously I was following the task of tension HAVE with the Done.vi of task completion is DAQmx.

• With the help of time interleaved sampling (TIS) to get the highest sampling rate of data acquisition card

  I tried enabling to TIS using nodes of property for the appropriate channel and by disabling the TIS for strings, I don't want to use. When I do that, I get the following error:

  niScope read Cluster.vi:1internal software error has occurred in the extended software. Please contact the support of National Instruments.

  Name of the component: nihsdru
  File name: p:\Measurements\highSpeedDigitizers\hsd\driver\trunk\1.10\source\redirection\tHardware.cpp
  Line number: 1038

  State code:-223425

  Any ideas what to do?

  Thank you, but I managed to fix it with the help of a few colleagues.

• Save the completion up to data

  Hai, I had 9 VI. Each one is bound by exec system. To begin to put an end to each data should save to a file. Each display of vi in a different window. How to add each of the data in a single file

  Chains of multiple screw export to the same text file?

• How to save the program table AFTER data is complete? (a single solution sad time)

  Hi all

  I have a Labview program that gathers data from the oscilloscope as an experience runs.  Some hardware problem made me have to stop the program at the beginning and some silly programming on my part was no infallible control to backup my data if a situation of demolition program was born.  (I have it autosave the table when occurs a software error or the program ends or you press a manual off button).

  As soon as now, I have a table of data filled on my screen before I need program with all of the experimental data, so I need a way to save this current data table. This is a great set of data which is not easy to repeat.  Is there a way to do this? (in addition to writing a better program for the next time

  Thanks for your help!

  Yes, you should be able to click with the right button on the table and copy the data or export them to Excel.

  You can also try to create a constant in the table (do not delete the original until you have your data if!) and then you can copy + paste into a simple VI which records data in the constant to disk.

• Two issues that are frustrating me: save the photos and mobile data.

  For about a week now that I have had a pop alert on my phone indicating that my "Storage Space Is Running" Out.  Today is the first time that I was able to connect my phone to the PC.  I am trying to export my photos from the phone to my PC, so I can clear some of the memory, but try to understand this is DOING MY HEAD IN!  It is so much easier with the iPhone. (I could am SD memory card but the * beep * things are not cheap here in Australia).

  Can someone please explain in simple terms how to go on this subject?

  The second question is that, with the plan of my phone, I have a 1 GB data limit a month and I have a widget on the home screen telling me how much I used etc.  The other day I got a "Use of data" alert on my phone that I put 1 GB and let me know when I'm near my limit (took me to alert me when I have 100 MG remaining).

  I don't want to disable mobile data.  I have my phone will automatically connect to the wifi at home.

  Please help me before I break this phone.

  OK, so whenever you connect your phone to a PC, find the Xperia Z2 on 'My Computer' then click on internal storage then look for a folder called DCIM and click it. Then click on ANDRO100 inside this folder are photos and videos.

• Blu - ray error: "file already exists", Code: "6", "Audio buffer overruns. Total too high sampling rate

  I hope someon can help out me here - I'm stuck in my tracks.    I read through many discussions on this topic, but do not understand what is happening here.    CS6.  Note: I used even before I put this material on a disk.  I currently have a separate sequence paired down, with only about 1/5 of the entire production.   1920x1080i, 29.97, H.264 Blu - ray, VBR 2-pass, PCM inside PP to produce .m4v/.wav.  This is what confuses me - I tried a target of 10 and a maximum of 20, and I still get this error.   I tried different bits max rate 'default' settings for transcoding in again.    Note the timecode listed "0.000000" - maybe someone said something.    I don't really understand yet again, so maybe it's something that I've set wrong in still vs my exports?

  

  Good news!  Suggestion of Stan update seems to have solved the problem.  I think it was on the side of the PP, not yet.   I have updated both, but still using to try to make a comic about made export with the PP before update failed.   But once I upgraded to PP, a new rendering, etc. I am able to build a BD.

  Disappointment, before I had installed all updates patches, but later everything uninstalled and ran the vacuum cleaner, try to debug another problem.

• How to save the old and new data

  Hello...
  
  I want to create a table that works even the concept of the Audit table.
  I want my table to be updated with the two old values n new values when a change is made.
  
  It is possible to do with it, using the features of audit table.
  
  Suppose that a user changes more than paid my table must capture the areas where changes are
  
  fact...
  
  I would also like to record the IP address of the specific system where is made the change/remove. How to get the ip address of the client system.
  
  My table structure
  
  Number of Sr_no
  DateTime varchar2 (15)
  VARCHAR2 (20) user_name
  TableName varchar2 (20)
  operation varchar2 (6)
  ip_address varchar2 (15)
  old_data varchar2 (20)
  you see varchar2 (20)
  
  
  Thanks in advance...
  
  
  with cheers
  Sprity...

  spirty...,
  As your username connects using their identifiers, capture this information in a set of global variables of forms (for example. OVERALL. UserID, OVERALL. Username, Global.PW, etc.), then change user to PAY. Since you switched the PAYROLL user at this point I guess that all transactions would be recorded against the user PAYS. Now, if you create an API to register the transaction audit, you can pass the pre and post values as well as the user information stored in your global variables from forms to the API to record the transaction on the real user account instead of the payroll account. I am considering this API as a fairly simple procedure of stored database that other takes a lot of individual settings for the new/old & user information or the stored procedure would preferably take a Ref Cursor (RC) or a Table PL/SQL of Records (PTR) with all the news, old and user information entered in the RC or PTR. This gives more flexibility and allows you to overload the procedure to handle any table need checked simply passing the name of the table concerned as one of the parameters to the procedure.

  Once again, my description is pretty generic because it is supposed to inspire your solution. The only real concern I have with the demands of your customers, it's you will need to enter their password. Why would they want the written in the verification of password table?

  I hope this helps.
  Craig...

• Sample rate Max USB-4065

  Currently, I am trying to log readings of DC voltage with an AA battery in an ASCII using LabVIEW 2009 of SingalExpress file and the USB-4065 digital multimeter (DMM). I have two stages:

  (1) acquisition of Signal > voltage DC using DMM

  -resolution 4.5 with 3.333333E the value-5 sampling period

  2) save in ASCII

  -The value to add to the file, delete the file after each race

  Faster reading, I can get is a data point written in the ASCII file every eighth of a second.

  Furthermore, I am new and software OR LabVIEW, the LabVIEW SignalExpress software I have is only for evaluation as it was included in the CD of the driver for the USB 4065 DMM.

  1. Max (30 000 samples per second) sampling rate is only achievable by a LabVIEW VI?
  
  
  2. Don't I have the wrong settings for DMM step?
  
  
  3. Is it because I haven't activated SignalExpress and am only using the evaluation version?

  Thanks in advance for any help!

  Hello Lukos,

  You are assuming that you need access to lower level functions in order to obtain the higher sampling rates. In order to get these speeds, we need to disable some settings that are not accessible via Signal Express. You can create a VI and then use a step VI call in Signal Express to stay in the same environment.

  Kind regards

  -Travis E

• The sample rate conversion

  Hello

  If I scan documents to Audition 2.0 48 kHz and then convert to 44.1 using software for recording on CD, would there be loss of quality?

  Thank you.

  If it has voice recordings and you need an original CDs can just master at 44.1 k and have done with it. Since there is no content in your files anywhere near the limit of Nyquist (this is half of the sampling frequency and represents the highest frequency at this rate, you can save), then all you need to do is an extra oversampling when the format of distribution requires.

  The speech contained in frequencies up to 12 kHz (hopefully) and even if you add music the answer will not exceed 20 kHz - and in any case, this is where the human ear cup (in children - rather less than that of adults). Therefore recording at 44.1 k means that you will capture all this, and you will be able to register without creating large files by scanning of a plu top noise load audio - which is all that recording at a higher sample rate will reach.

  If you need pace to a different distribution format, then you have lost nothing - just do it when you need to.

  I record a lot of original acoustic music and unless specifically requested to (which is never arrived, I might add...) I always master to 44.1 k. It's been proven by research academics caution that no one can tell the difference, then what is the point of a high sampling rate, besides wasting disk space unnecessarily?