Wave form chart for different sampling rate

Hi
All,
I have to use different sampling rate to get pressure data. Can I use the waveform chart to monitor the pressure data with time?
If not, what kind of graph should I use?

Different sampling rate than what? A chart or graph can be used. Depends mostly on how you want to update it.

Similar Messages

Different Sampling rates for different channels in Analog Input

Hi,
I would like to acquire data at different sampling rates on different channels say ACH 0, ACH 1 and so on. I have a PCI 6052E board and NI DAQ 6.9.2. Also is it possible to simultaneously perform Analog output on two different channels along with the Analog input? What will be the problems/consequences as far as the system resources are concerned. I am a beginner in this area and would greatly appreciate any help/pointers for my queries.
Thanking you in advance
Deepak

Search the eaxamples that ship with LV.
Theer is one called simultaneous input and output or something like that.
It will get you started.
re: multiple scan rates. This is acoomplished by sampling all channels at the highest rate and throw away the expttra samples you do not need.
Ben
Ben Rayner
I am currently active on.. MainStream Preppers
Rayner's Ridge is under construction

How can I add a curve with a different sample rate behind another curve to show it like one in the report

I saved two curves with different sample rates with signal express in waveform.
Now I want to add the curves behind and show them in a report.

Hello MReizner,
Both the DIAdem VIEW and REPORT panel use the time information from your Waveform channels (make sure they actually have the waveform symbol, not the numeric data channel symbol in the Data Portal) to plot the data in the same axis system.
In the example below I have two waveforms, one sampled at 5 Hz and one sampled at 1 Hz, both in the same axis with the same time channel. All I did was drag the data from the Data Portal onto the axis. DIAdem automatically takes care of creating the correct time channel and plotting the data with the correct points if the data is stored as a waveform.
I hope this answers your question, please let us know if further clarification is required ...
Otmar D. Foehner
Business Development Manager
DIAdem and Test Data Management
National Instruments
Austin, TX - USA
"For an optimist the glass is half full, for a pessimist it's half empty, and for an engineer is twice bigger than necessary."

Parallel acquire 2 signals with different sampling rate on 2 cards

Hi NI,
I have cDAQ-9178 and NI 9221, where sample rate is 10kHz and NI 9219, where I need sample rate about 10Hz, it's possible this confiruration for parallel acquire?
Thank you.
Neolker

Hi Neolker,
Do the tasks need to be synchronized or are totally independent tasks?
If the tasks are independent, than you basically have to create two different tasks, even in two different loops that will run with different sampling rate.
You can have them in the same loop the reading if you assure that the data are transfered in chunks to application memory according to tha sampling rate (Ex. for 10kS/s rate you can transfer data with 1kS chunks and for 10S/s rate with 1S chunk).
If you want to synchronize them, you will need a counter that will divde the sample clock from 9221 and route it to 9219.
Let me know if you need more help.
Best regards,
IR

On the fence for which sample rate to record at (44.1 vs 96)

Been reading tons of posts on the sampe rate debate. My friends (across the country) and I are about to start to collaborate on the great American rock album that we didn't quite get right back in the day in college. I'll be running the show sending them scratch tracks with clicks so they can lay down individual tracks and I'll import them.
I'm torn on which sample rate(s) to use -- and want the best quality possible, of course. I've boiled it down to the following pros per sample rate. Any advice/comments much appreciated. thanks
44.1 Pros
- Friends across the country can use GarageBand (44.1kHz is max) to lay down a single track and send to me to import into Logic Pro and mix (and will match up)
- GarageBand is free, Logic Pro is $199; Apogee JAM is $99, Jam96kHz is $129
- GarageBand requires fewer resources (1GB RAM vs 4GB for Logic Pro) so they don’t have to have beefy Macs
- Smaller file size to post/share (Dropbox cost per/storage size)
- Picture mixing a bunch of 44.1 tracks vs a bunch of 96 tracks even if on my beefy Mac running Logic Pro; fan running, gasping for air etc?
96 Pros
- 96 sounds noticeably better than 44.1?
- While 44.1 standard for CDs, that’s no longer how music is largely distributed
- Should always record to max resolution you can be “future proofed”?
- Current lossy formats support up to 48kHz so 96kHz good as will cut cleanly in half when bouncing
88.2 Pros
- Maybe choose this so friends using GarageBand can record at 44.1 so upscales more cleanly? But then bouncing down to 48kHz not as clean? (I don’t recall Logic Pro allowing me to choose 44.1 for bounce rate.. always says 48 for MP3/M4A)
thanks!

rcook349 wrote:
44.1 Pros
- Friends across the country can use GarageBand (44.1kHz is max) to lay down a single track and send to me to import into Logic Pro and mix (and will match up)
- GarageBand is free, Logic Pro is $199; Apogee JAM is $99, Jam96kHz is $129
- GarageBand requires fewer resources (1GB RAM vs 4GB for Logic Pro) so they don’t have to have beefy Macs
- Smaller file size to post/share (Dropbox cost per/storage size)
- Picture mixing a bunch of 44.1 tracks vs a bunch of 96 tracks even if on my beefy Mac running Logic Pro; fan running, gasping for air etc?
96 Pros
- 96 sounds noticeably better than 44.1?
- While 44.1 standard for CDs, that’s no longer how music is largely distributed
- Should always record to max resolution you can be “future proofed”?
- Current lossy formats support up to 48kHz so 96kHz good as will cut cleanly in half when bouncing
88.2 Pros
- Maybe choose this so friends using GarageBand can record at 44.1 so upscales more cleanly? But then bouncing down to 48kHz not as clean? (I don’t recall Logic Pro allowing me to choose 44.1 for bounce rate.. always says 48 for MP3/M4A)
thanks!
44.1 kHz still is pretty much standard for MP3's.
Your friends/collaborators can pretty much use any application that can record PCM (or even MP3) audio; even if they're not playing to a steady tempo, you can line everything up in Logic, with flex.
Using Garageband and one set tempo should also work. Just remember that you cannot open Logic files in Garageband, only Garageband files in Logic. The Audio Files recorded by either, can be used (imported) by either.
Higher sampling rates will not "future proof" anything. In fact, that whole concept is flawed. Your best bet for now is simply 44.1 kHz 24 bit uncompressed PCM files in their most widely used form: AIFF or WAV.
96 does not noticeably sound better than 44.1, unless you have a top end interface and a very delicate and very complicated mix, and admirably acute hearing. In some interfaces 96 or 88.2 have been found to sound worse than 44.1, because of clocking inaccuracies getting progressively worse at higher sampling frequencies. I would stick to 44.1, it has lots of practical advantages (as you pointed out), and the sonic difference with 96 kHz is marginal at best, and certainly not worth the price: "double" rates need double the CPU power for any plugin processing. That's the biggest loss. Half a Mac.
Bitdepth on the other hand does make a significant difference. There is no reason not to record everything at 24 bits. Shorter: always record at 24 bits.
O, also just spotted your remark about Logic not "letting you" bounce MP3/M4a to 44.1 kHz. You must remember incorrectly, because I never bounce MP3 or AAC to any other frequency than 44.1 kHz. However, it may be that this rate is tied to the projects' sampling frequency as set in the project settings, and the last time I used 48 kHz was in LP 8. I'll check that now.

How to acquire data from 2 chs of the same DAQ card at different sampling rate

I am using single DAQ card (either 6013 or 6014) in my system i want to acquire data from 2 (or more) channels with following requirements
1. sampling rate of each channel should be independant of each other (say one is 20 Hz and other is 15 kHz)
2. data from all the channels should be acquired simultaneously.
3. coding must be done using DAQmx VIs
I have tried out following things
1. I created separate task for each channel: i found out that two tasks can not run simultaneously even though the channels are different
2. I tried out single task with two channels included in it. and i used 'channels to Read' property to determine from which ch. i want to acquire data: this method works fine if the sampling rates are same. but if i change the sampling rate of one channel it gets reflected in other channels as well.
can somebody help me out to solve this problem.
i will appreciate if somebody can post the sample code as my deadline is approaching
Tushar Jambhekar
[email protected]
Jambhekar Automation Solutions
LabVIEW Consultancy, LabVIEW Training
Rent a LabVIEW Developer, My Blog

Hi Dennis Knutson
Thanks for your suggestion.
Tushar Jambhekar
[email protected]
Jambhekar Automation Solutions
LabVIEW Consultancy, LabVIEW Training
Rent a LabVIEW Developer, My Blog

How to create a wav file from 24bit 96Khz sampling rate data

Hi
I am trying to make an VI which will play sound while acquiring data from PXI 4472 DAQ card.
My sampling rate is 96Khz and PXI 4472 card is a 24bit card.
Wave files are in 8 or 16 bit and the sampling rate is 8000, 11025, 22050 and 44100. How will I be able to play the data which I am acquiring.
How would i normalize the data into the required format needed for most of the sound cards to play.
Or are there any codec available in Windows XP which i call to play a 96KHz 24 bit sample
Does anybody ever encountered this type of problem.
Thanks in advance
Nitin

Whilst the 'standard' RIFF format specification usually accomodates 16 bit data, there is of course no reason that you can not create your own extension. It just won't be playable by Media player using the 'standard' installed drivers or codec. This may not be a problem....
WAV files can and do support other formats, you just need to know how to handle them......
There is howerver a 4GB limit (related to the pointer size in the WAV specification) which with higher bit depths on the sampling does start to become a bit of a problem.
To give you a few samples of other types of wav files check out the following site here
http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
The following definitions for WAV audio formats may also be of interest here
http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/WAVE.html
Good luck with 24 bits.

Multi plot wave form chart

i am acquiring 16 channels simultaniously scan rate of 1000samples/second. using chart with single channel (array of data format) i can see 10 sec duration. with multiplot view of least 4 channel i used waveform property but this will not getting 10 sec duration of x. this is showing just like a graph.
how to show my multiplot at least channel in one chart with continous updation....?
Regards,
Balaji DP

please find an attachment.
chart 1.jpg
i am trying to show the 4 channels of with 1000samples/second. even i am using the chart. i can see only 1 secon of data.
if i try with single channel array of data with out using wave form property with chart i can see 5 minutes of updattion.
how to see multiple plot atleast 4 channel with 5 minutes of duration?
Regards,
Balaji DP
Attachments:
chart1.JPG ‏39 KB
chart2.JPG ‏9 KB

Acquire 2 signals with different sampling rate?

Hello,
i am using first time labview signal express together with a NI pci 6251 & NI Scc 68 device.
There are 2 signals I wish to acquire over a 2 week time period. They should be saved in several tdms files. One is a dc Voltage.(sampling rate 100Hz) The second is voltage given by an acceleration Sensor. (sampling rate 4kHz)
When I try to acquire them I couldn't figure out how to set the sampling rate for each signal, only for all signals.
When I created two DAQmx assistant acquire tabs. An error occurs always: "the specified resource is reserved".
So my question is is it possible to do this?

hello markus_umd,
no, it is not possible to create different samplingrates for individual channels on one daq device.
if you want to record one channel slower, you must reduce the samples for that channel in your program.
if the channels have a common divisor, you could implement your task. you find more information here:
Sampling Different Channels at Different Rates with NI-DAQmx
kind regards,
robert h
NI germany

Converting wav files to a different bit rate

my friend has sent me some wav files of a very cool song for me to put vocals on, in 48khz. i use a korg stand alone work station to record which will only accept 44.1khz wav files. is there a way for me to convert these 48khz files to 44.1 files with itunes? i have tried a few things with no success. thanks for any input, ric seaberg

Assuming the file is already in your iTunes library, just go to iTunes > Preferences > Advanced > Importing, select WAV from the Import Using drop-down list and Custom from the Setting drop-down list. In the resulting WAV encoder window you will need to set the Sample Rate to 44.100 kHz.
Now, select the WAV track in your iTunes library and use the Advanced > Convert Selection to WAV menu to produce the new WAV file. You can distinguish between the files by selecting each and typing Command (Apple)-i to view the summary information, where the encoder parameters are displayed.

OMF - Mixing Different Sample Rates

Hello -
So, I've got an OMF file from a FCP project that I opened in Logic and have been mixing without any issues. Interestingly, all of the audio files associated with this project are 44.1 kHz, but I am mixing in 48k. But everything is right (sounds right, looks right, syncs with video correctly).
But... if I open the same files in an external editor or quicktime, they play back incorrectly. And if I save a file in a different program, even if it is still at 44.1, and bring it back into logic, it plays back incorrectly. This is problematic if I need to edit an audio file somewhere other than within logic (say I want to do some noise reduction in soundtrack pro).
Anyone run into this issue or have any ideas about how this happened?

There are a few ways to look at this.
1) Regions in the arrange all play back at the session sample rate. Example: 44k session, 96k audio file in arrange=slower playback
2) Logic automatically converts output sample rate so you can record independent of CA devices. Example: You have a session recorded at 96k, your interface is not connected, Logic will load Built in Audio, Logic runs tyhe sessions at 96k and converts the SRate of the session to match the supported sample rate. So you can run sessions at unsupported sample rates, this rarely makes sense if you cannot capture your audio at session sample rate (if needed).
3) There are a few other options for handling this, such as EXS24, which automatically handles SRC.
4) When Importing Audio Files there is a song preference which you can en/disable to automatically convert SR upon import.
I think your friend may have referred to point #2 and it was interpreted as point #1...perhaps. Hope this clears things up. J

Why is it that I can't do a continuous streaming to disk with a 5102 scope card (PCI) when I can do it with a DAQ Card of much lower specs (my requirement is for small sampling rates only)?

I am told that the 5102 Card (PCI) does not support continuous streaming of data to the hard disk. My application requires only very low sampling rates. If I can do it with a low spec DAQ Card using LabView why can't I do it with this card?

Hello,
The PCI-5102 is a high-speed digitizer card that has a slightly different architecture than the DAQ cards and was not built with the ability to stream data to the PC. However if you are sampling at low rates you can still acquire up to 16 million samples, which is done by using dma to tranfer data from the onboard memory on the 5102 to the PC memory. However, you will not be able to save the data to disk until the acquisition is complete.
Another option would be to purchase either a DAQ card or a PCI-5112. Both boards can continuously stream data to the host PC and you should not run into any PCI bus limitations if you are stream to disk at relativiely slower rates.

Audition 3 seeing a different sample rate setting than what the device shows

Hi,
I have just installed Adobe Audition 3, along with the 3.01 patch, on a brand new system running Windows 7 64 bit. The mother board is an Asus Sabertooth X58 using Realtek High Definition Audio. The device drivers show that the audio sampling rate for line input is set to 24 bit 192K. I wanted to set it to the maximum that the sound card would allow to test performance and audio quality.
The problem is when I bring up Audition 3 and hit record, I get the message "We do not support recording when your file does not match your hardware sample rate. Your current hardware sample rate is 44100Hz". Clearly this is not the case since the Line In Properties - Advanced tab is displaying "2 channel, 24 bit, 192000 Hz (Studio Quality).
Under Audition's Audio Hardware Setup it shows only one choice for Audio Driver: Audition 3.0 Windows Sound. It also displays Sample Rate: 44100Hz, Clock Source: Internal, Buffer Size: 2048 samples with no way to change these values.
If I click on the Control Panel button I get:
DirectSound Input Ports:
Device Name: Line In (High Definition Audio Device
Audio Channels: 2
Bits per Sample: 16
Anyone know of how I can change these settings to get Audition to agree with the device settings?
Thanks
Dale

DaleChamberlain wrote: Anyone know of how I can change these settings to get Audition to agree with the device settings?
I'm afraid that life is nowhere near that simple. The main issue here is that Audition, in common with most audio software, uses a driver system called ASIO to talk to the sound device - this cuts out a lot of the OS and reduces the latency of the system considerably. There are several problems with ASIO though - the first being that it only supports a single device per system (or sometimes multiple identical devices if the manufacturer can make them look like a single device), and with software designed to use this driver, then to use any other driver (like a native Windows one) you have to use a converter stage like ASIO4ALL. This will convert the ASIO streams to WDM, and let you use multiple sound devices - but with increased latency.
It's the second problem that's really going to stuff you though - and that is that quite reasonably, ASIO is limited by its inventors to run only at three sample rates; 44.1k, 48k and 96k. So there's no way you can run at what you think might be a higher quality setting. All settings above even 48k are making your sound device work much harder, and for what? All that happens is that you increase the potential frequency response to way beyond the human hearing range - to no purpose at all. You don't have sources that can produce useful output at these frequencies, and you certainly don't have the means to reproduce them. This has all been well documented and explained before, so I'm not going over all that again. In a nutshell, Nyquist points out that any digital sampling device has a frequency response limited to a maximum of half of the sample rate, so for 48k that gives us a frequency response up to 24kHz - comfortably higher than any adult can hear by quite a long way. Anything you sample and record beyond this by using even 96k is nothing but noise as far as humans are concerned, and unpercievable noise at that.
So what the line input properties tab is saying is, if you have a non-ASIO driver designed to support all potential rates, possible. You don't have an ASIO driver available, because it's a built-in sound device, and anyway you've already pointed out that it's using the Audition Windows driver (a cut-down version of ASIO4ALL, effectively), so a conversion is already taking place. What Realtek refer to as 'High Definition Audio' is no such thing - all on-board sound devices of this nature are of universally low quality, and to improve this you'd need an external device - of which there are many available, usually with dedicated ASIO drivers. But none of them will work with ASIO beyond 96k, simply because the standard doesn't support any higher rates.
If you download and use ASIO4ALL (it's free), then you will get an additional control panel which will show you exactly what your sound device is capable of doing as far as Audition or any other ASIO software is concerned, and this is a useful diagnostic tool anyway, so it's worth doing. You just select this option when installed, instead of the Audition Windows Driver.
I'm sorry to be the bearer of what seems like bad news, but actually, it isn't. You will percieve no quality difference at all running at anything beyond 48k sample rates; all you will be doing is wasting your computer's resources unnecessarily. You waste both processing resources and hard drive space by processing at ridiculously high sample rates, and there are zero returns.

Effects of using differing sample rates?

I recently had a great deal of trouble rendering a 10 minute sequence. I thought it was because I had some effects (pan and scan stills layered on top of each other).
After much time and hair pulling, I finally was able to render the sequence and export it to DVDSP.
However, it turned out that the DVD had skipping issues.
I went back to look at my sequence and noticed that while my video was all in 48K, the narration track was at 44.1. Could this difference be the reason I'm having so much trouble with this sequence? Should I convert the narration to 48K? I haven't gotten any error messages regarding the audio or sample rates and it seems the audio has been rendering ok.
I'm using FCP3.
Thanks for any advice.

I think it's likely that the 44.1 is throwing it off. You can take the narration clips from your timeline directly into a bin, export and reimport as 48kHz, and replace them on the timeline. Be aware of whether they still fit in their slots or not, and see what happens.
If that was it then you get to give yourself points!

Different sample rates from different cameras

I took some video with my Panasonic video camera which samples at 48kHz. At the same time I was shooting with my Digital camera which samples at 8kHz. I am trying to pan between the two of them but the timing does not match up through the video. The clips are approximately 3 min and 20 seconds. Any idea how I can get these two things to synch up?
thanks

Video camera is Panasonic. PV-GS250
Digital camera is Lumix DMC-LX2
The Browser in FCE shows me that the video camera's audio format is 16-bit integer and 48kHz, the audio from the digital camera is 8-bit integer and 8kHz. I assumed that was the bit depth and sample rate.
Good question on the video format of the digital camera. I just assumed it was DV-NTSC, and so thats how I imported it. I will check into that as it seems to be the most likely solution.
Thanks!

Wave form chart for different sampling rate

Similar Messages

Maybe you are looking for