Changing the Sampling rate

Hi,
May I know how do I change the sampling rate on a Thermocouple node dynamically through a VI on the host ?
I notice there is a Config node in the "Start" case where I can set the "Sample Interval Mode" to "Host Driven" but I do not know how to continue after that...
Thank you very much !
Regards,
Xinke

A user message is created by simply right-clicking on the node and selecting "Create User-Defined Variable." Once you set it up, that variable will update over the network automatically.
Once you've got that setup, you need to figure out how you are going to use that variable. You should probably start here: The LabVIEW Wireless Sensor Network Module - Under the Hood
Note: User-defined variables are only available in LabVIEW 2011 for WSN.
Chris
Certified LabVIEW Architect
Certified TestStand Architect

Similar Messages

How can I change the sample rate (delta X)?

Hi, I create vi which can measure resistance, it will record the data and write to a file.
however, I want to change the sample rate which is delta x. is that anyway I can do?
I have try to put a clock inside the loop, it does slow down the speed but not the sampling rate.
Please help me!!
Thank you!

Hello Shan405,
Adjusting the Reading Rate of a DMM
As noted in the above DeveloperZone article, there are a lot of factors contributing to a reading rate of a DMM. 'Sampling rate' can only be configured in digitizer mode, which can only be achieved on the flex DMMs
Using the NI PXI-407x DMM as an Isolated Digitizer
Regards,
Glenn

Changing the sampling rate of the Digitizer 5142

Dears,
Can i change the sampling rate of the digitizer 5142 ( for the 100MS/s can i change it to be smaller), can i change it at the spectrum mode and the IQ mode?

Thank you for posting on the NI Discussion Forums. Since this question is related to your other forum post, let’s move the discussion over there to answer this question.
Chris_G
Sr Test Engineer
Medtronic, Inc.

IPhone: AudioQueue - is it possible to change the sample rate?

I've been playing around with the AudioQueue stuff for a few days and it's all working fine.
I was trying to build a low-latency playback system by making the streaming buffers the same size as the audio file and pre-loading the buffers (which works fine) but I've hit a snag.
I've been trying to get the streaming to work at different sample rates so that I can play back the same sample at different pitches. I managed to do it by modifying the sample rate in the AudioStreamBasicDescription structure but in order to actually make the stream playback at the new rate it seems you have to create a new output, reload the audio file into the buffers and re-enqueue the output queue before starting playback again, otherwise the sample rate change has no effect.
There is a method to set queue properties; AudoQueueSetProperty() but unfortunately the sample rate Property (kAudioQueueDeviceProperty_SampleRate) is read-only
Can anyone suggest a way to achieve this with AudioQueue or do I need to move over to OpenAL?
Thanks,
Neil

Dan,
there is one point in your understanding, which i am not sure what you think about when talking about it: I understand E-series devices do not support this property change while the VI is running.
infact, you cannot change the sample clock rate during acquisition. but
this does not mean that you cannot change it while the VI is running.
you have only to interrupt the acquisition. since you want to acquire
continuous, this would have the same effect as stopping the vi, i asume.
so the best way to accomplish this task is to use an external clock.
this is e.g. often used for acquistion on rotating shafts. the
acquistionrate is always e.g. 24 points per revolution regardless of
the rotational speed of the shaft, except for a maximum frequency of
course.
Norbert B.
NI - Germany
Message Edited by Norbert B on 09-14-2005 04:16 AM
CEO: What exactly is stopping us from doing this?
Expert: Geometry
Marketing Manager: Just ignore it.

Dynamically changing the sample rate

hello!
usually we can't dynamically change the sample clock when the program is running.So I want to ask how to dynamically changethe sample rate when program is running,Can you give me a example?
Thank you!

I think there is no such example.
But i think you could add a while loop structure outside your main program. In the front panel, add a button named change sampling rate, when you press the button, the program stops the present acquisition and into a new acquisition configuration and start acquisition.

If your Lightroom 6 is crashing in the Slideshow module, check the SAMPLE RATE of your music track

After spending over 2 hours with an Adobe technician who had control over my Windows 7 machine, I've learned something that I don't think Adobe knows yet... I upgraded from LR 5.7 to LR 6 (not CC) and began having issues when I was working in the Slideshow module. When trying to add a music track, Lightroom would crash ("unexpectedly quit"). I couldn't for the life of me figure out why. I turned off the GPU since a fair amount of chatter on the forum related to that, but it didn't change anything.
So I finally waited patiently for the Adobe Chat line to finally get a tech to respond...and he was exceptionally detailed and methodical in his approach to diagnosing my issues. I could follow on my screen as he did a variety of things that indicated he understood what he was doing. Unfortunately he got to a point where he thought (as did I) that he had found the problem/solution...which he thought was related to the bit rate. He concluded that music with a Variable Bit Rate (VBR) as opposed to a Constant Bit Rate (CBR) was causing the problem. Also unfortunate was that this "discovery" was near the 2-1/2 hour mark and we did not apply the scientific method to property determine if we had narrowed down to cause/effect.
Based on my further analysis, he was close, but not right on. It isn't the BIT rate mode that is causing the crash, but rather the SAMPLE rate. Turns out that the vast majority of songs in my iTunes library have a sample rate of 44.100 kHz--and those work just fine. However, I have about 15 songs, mostly downloaded as free background music from the YouTube creation library, that have a sample rate of 48.000 kHz--every single one of those cause my LR6 to crash.
If you're having a problem with Lightroom crashing in the Slideshow module, check the SAMPLE RATE of your music track.

If you're on Windows, check out a free program called Audacity--it's fantastic. With Audacity the process of converting a 48.000 kHz to 44.100 kHz is as simply as opening, changing the sample rate, and then exporting again, as an MP3.

The sampling rate of FP-AI-110

How can I change the sampling rate of cFP-AI-110 in Labview 8.5?
And the filter settings of cFP-AI-110 (50,60 and 500Hz) is equal to the sampling rate?

Hi!
   I've found this on cFP-AI-110 Operating Instructions, pag. 8:
   "The filter setting determines the rate at which the [c]FP-AI-110
   samples the inputs. The [c]FP-AI-110 resamples all of the channels
   at the same rate. If you set all of the channels to the 50 or 60 Hz
   filter, the [c]FP-AI-110 samples each channel every 1.470 s or
   every 1.230 s, respectively. If you set all of the channels to 500 Hz
   filters, the module samples each channel every 0.173 s. When you
   select different filter settings for different channels, use the
   following formula to determine the sampling rate.
   (number of channels with 50 Hz filter) ×184 ms +
   (number of channels with 60 Hz filter) ×154 ms +
   (number of channels with 500 Hz filter) × 21.6 ms =
   Update Rate"
graziano

Changing audio sample rate in FCP

Hi all
A quick question, can someone explain how to change the sample rate of audio in FCP, if at all possible. Or do I need to do it in Soundtrack?
Many Thanks
Adrian

You can't change the sample rate in FCP; you can only change it by exporting it. You can do that from FCP, STP, Compressor, QuickTime, iTunes, and a bunch of other applications. Making a Compressor droplet to do this is perhaps the easiest way.

Why cant I control the sample rate?

I am sure there is a simple programming answer to this but I am drawing a blank. I would like to slow down my sample rate in the following block diagram, however changing the sample rate at my Sample Clock seems to have no effect. Any ideas? Thanks in advance for your help
Solved!
Go to Solution.

NIquist wrote:
How about a Timed Loop? It's one of the main structures. You can also just drop the Wait.vi (Timing Palette) in your current loop.
I recommend the Wait (ms) function over a timed loop unless you specifically need the timed loop behavior. I just learned today (looking up something related to a different thread): "Each timed structure on the block diagram creates and runs in its own execution system that contains a single thread, so no parallel tasks can occur." In a regular while loop, tasks can run in parallel.

Manipulate the sample rate through a layout

hola
i'm new user of dasylab and i would like to manipulate the sample rate through a layout ;i don't know how
also other small thing when i use coded swich and i want modify the text in the Switch Window i don't know how
Hope that someone would be able to help.!Lots of thanks!

The answer to the sample rate question depends on the device - is it using the DRIVER time base or a different device timebase.
To change the sample rate, you must replace the sample rate value with a Global Variable (e.g. ${Var_1}. But, there is a limitation - you must Stop the measurement and restart for it to use the new rate. This is because most drivers set up the scan parameters before the measurement starts, and it cannnot change without stopping the measurement scan.
For the Coded Switch,
You can select Value and unit or Name to be displayed. Select the radio button next to name (not Module name or channel name, just Name) and enter the alphanumeric string that you want to display. You can also change the Value for each switch setting, so that it is possible to have a unique value and a unique name for each switch position.
- cj
Measurement Computing (MCC) has free technical support. Visit www.mccdaq.com and click on the "Support" tab for all support options, including DASYLab.

Adjusting the sample rate

How do I adjust the default sample rate setting to 44100?

Blastbot wrote:
If you are not going to answer this because it is on an older product, that is lame. I tried to get on a forum for said older product, but that is grayed out from the list of products you have a forum on, which is also lame.
Well, I certainly don't wish to be described as "lame," Blastbot, as it is a personal fear of mine. The forum for previous versions of Audition is available at http://forums.adobe.com/community/audition/audition_previous but since we're here and my earlier obligations today have wrapped freeing up a few minutes of my time, I'll be happy to walk you through the process for Audition 3.0.
In Audition, click Edit > Audio Hardware Setup... In the dialog that pops up, you'll see the current hardware sample rate listed and a Control Panel button that will display the device configuration panel for the selected driver. You should be able to change your sample rate by clicking that button, depending on which audio driver is selected. The default "Audition Windows Sound" driver, which is generic and best suited for audio devices without their own drivers, defaults to the Windows sample rate. If you have a good audio device with its own, native ASIO driver, you can usually modify the sample rate within their control panel.
Since you're apparently running Windows, although I'm not certain which version you have, you may need to change the sample rate of your audio device at the Operating System level. You can usually reach this menu by opening the Windows Control Panel and launching "Sounds and Audio Devices" (on Windows XP) or "Sound" (on Windows Vista and Windows 7.) Here, you'll need to navigate to your playback device selection, open the device properties, and adjust the default sample rate that you desire.
More detailed information can be found in the Adobe Audition Help documents either from the Help menu item, or the Help button located in the bottom-right corner of the Audio Hardware Setup window. Additional information about Sample Rates can be found on Wikipedia at http://en.wikipedia.org/wiki/Sample_rate
I sincerely hope this helps answer your question and relieves me the burden of lameness, at least as far as this issue extends.

Gain changes on sample rate change

I wrote an application with 2 channel analog input with different gains. If I change the sample frequency the gain setting seems to mix up. But the sample frequencies work if used as first. So, only after changing the sampling frequency the gains are mixed and could not be 'reset' by calling Scan_Setup() from NIDAQ library again.
Any ideas?

Hello,
So it seems that changing the sampling frequency is affecting the gain of a channel?
I am not that sure what is causing this behavior, if you could answer the following questions, we could begin to figure this one out.
1.) How do we know that the gains are mixed up? Is one signal significantly larger then before? Is the code width decreased?
2.) Are you trying to change the sampling rate during acquisition? Are you running the acquisition to the end, then calling Scan_Setup again?
3.) Do you get this error if you call DAQ_Clear before reconfiguring the sampling rate?
Also, if you could post some pseudo code that explains which DAQ calls you are making, and their order, we should be able to get this problem figured out.
Best regards,
Jus
tin T.
National Instruments

How can I change the frame rate mode?

Adobe Premiere Pro exports my videos with a variable frame rate. Because of this long videos are asynchronous. How can i change the frame rate mode? So Adobe Premie Pro exports my videos with a CONSTANT frame rate? The original video has a constant frame rate...

Sorry my mistakes. I am from Germany.. so...I mean with asynchronous that the video file and the sound file are displaced. In the beginning of the Video it's normal, but if the videos is very long it goes VERY "asynchronous"
I analyzed the video with "mediainfo":
General
Format                                   : MPEG-4
Format profile                           : Base Media / Version 2
Codec ID                                 : mp42
File size                                : 2.00 GiB
Duration                                 : 20mn 13s
Overall bit rate mode                    : Variable
Overall bit rate                         : 14.2 Mbps
Encoded date                             : UTC 2013-12-11 13:56:13
Tagged date                              : UTC 2013-12-11 13:58:30
©TIM                                     : 00:00:00:00
©TSC                                     : 30
©TSZ                                     : 1
Video
ID                                       : 1
Format                                   : AVC
Format/Info                              : Advanced Video Codec
Format profile                           : [email protected]
Format settings, CABAC                   : Yes
Format settings, ReFrames                : 3 frames
Format settings, GOP                     : M=3, N=30
Codec ID                                 : avc1
Codec ID/Info                            : Advanced Video Coding
Duration                                 : 20mn 13s
Source duration                          : 20mn 13s
Bit rate                                 : 14.0 Mbps
Width                                    : 2 560 pixels
Height                                   : 1 440 pixels
Display aspect ratio                     : 16:9
Frame rate mode                          : Variable
Frame rate                               : 30.000 fps
Minimum frame rate                       : 30.000 fps
Maximum frame rate                       : 30.030 fps
Standard                                 : PAL
Color space                              : YUV
Chroma subsampling                       : 4:2:0
Bit depth                                : 8 bits
Scan type                                : Progressive
Bits/(Pixel*Frame)                       : 0.127
Stream size                              : 1.98 GiB (99%)
Source stream size                       : 1.98 GiB (99%)
Language                                 : English
Encoded date                             : UTC 2013-12-11 13:56:13
Tagged date                              : UTC 2013-12-11 13:56:13
Color primaries                          : BT.709
Transfer characteristics                 : BT.709
Matrix coefficients                      : BT.709
mdhd_Duration                            : 1213733
Audio
ID                                       : 2
Format                                   : AAC
Format/Info                              : Advanced Audio Codec
Format profile                           : LC
Codec ID                                 : 40
Duration                                 : 20mn 13s
Source duration                          : 20mn 13s
Bit rate mode                            : Variable
Bit rate                                 : 158 Kbps
Maximum bit rate                         : 254 Kbps
Channel(s)                               : 2 channels
Channel positions                        : Front: L R
Sampling rate                            : 44.1 KHz
Compression mode                         : Lossy
Stream size                              : 22.8 MiB (1%)
Source stream size                       : 22.8 MiB (1%)
Language                                 : English
Encoded date                             : UTC 2013-12-11 13:56:13
Tagged date                              : UTC 2013-12-11 13:56:13
mdhd_Duration                            : 1213777
And there i can read that the "frame rate mode" is variable... and that's the bad thing... I want to change that point to a constant frame rate mode. I use the H.264 codec

FMLE won't let me set the sample rate with Blackmagic Decklink Studio 2 (Windows)

Hi,
I am trying to set the audio device on FMLE to the Decklink Studio 2's audio inputs. However,
the sample rate sticks to 44.1k and I can't change it to the 48kHz sample rate that
the Decklink defaults to (standard for broadcast is 48k).
Try as I may, it isn't changeable. I seem to remember a similar problem where the
sample rate defaults to the system audio card (on the motherboard) which is at
44.1.
Any ideas on a work-around? The audio from the Decklink is *much* better than
the built in card.....
thanks
jeff

One thing you might try doing is trashing the preference file in your user directory:
~/Library/Preferences/com.apple.compressor.Compressor.plist
Like repairing permissions, this is a bit of a "catch-all" fix, not something specific to this particular problem.
Also, have you tried different file types like: TIFF or JPEG, there could be an issue with output from Fireworks. You could use QT Pro to transcode the file. Sometimes that can work to fix funky problems.
Does the Text Overlay filter work?
BG

How do i change Audio sample rate to suit?

Hi,
I Just captured a small clip from my DV camera on final express 4. At the end of the capture i got a message saying that the sample rate on my source may be different to the rate on my capture and it may cause the video and audio to go out of sync. This did happen so i checked the camcorder and it is set to 12bit. How do i set the captute settings to suit??
Thanks,
Dave.

Can I ask a stupid question?
What do you mean "if you overrun" I just encountered all the above issues. First tape imported perfectly once I changed it to 32. The next one appeared to but then like jerryws1431 said, FCE quit. I wasn't here at the point it quit, but if was either after or during the "Validating Audio". Last time I looked it was validating.
So, if I have a 60 minute tape and I tape 51 minutes, is that too much to have on one tape, or am I completely missing something......
Tom, thank you so much for all this information! I would have been stuck with out of sync audio had it not been for you!

Changing the Sampling rate

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