Sample rate is 48.009.2Hz...???

good day all...
wondering if anyone can help me...editing a doc...most clips have audio sample rate at 48k..but some...(a lot) have sample rate of 48009.2 ---48009.3...is there an easy way to convert without having to render all...? if not..how can i stop this from happening in the future...director of doc gave all footage on an external hard drive..he captured all footage...we are both first timers on this kind of project...any and all help would be wonderful
Raindog

Select each clip and go to File/Export as AIFF. Make sure settings are for 48Khz/16bit. (Might be able to do a Batch Export, but I've not tried and my system is busy, so I can't confirm.)
You will need to relink the new audio with original video.
This often happens when capturing looong portions of tape...best to log and capture smaller clips. I have seen this happen with short clips as well, but rarely.
I found that if you have say an event tape that runs long, it's best to at least capture in no more than 20 minute segments.
K

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Setting the sampling rate in SignalExpress

I am using a cDAQ-9172 with a strain gauge module and a thermocouple module and using SignalExpress. I want to acquire data at a pretty low frequency rate (2Hz), but I am unable to use 1 sample on demand in the acquisition setup. I get the following error.
Error -201087 occurred at DAQ Assistant
Possible Reason(s):
Measurements: Task contains physical channels on one or more devices that require you to specify the Sample Clock Rate.
Specify a Sample Clock Rate.
Device: cDAQ1
I am unable to setup a sample clock rate as it is 'blacked' out by the software in the 1 sample on demand mode.
When I try to use continuous or 'N' sample mode, the cDaq is sampling at a rate of around 1600 Hz, even though I am putting in a value of 1Hz on the seti[ screen. The large amount of data will prevent me from downloading the data to Excel for further reporting. I obviouisly don't need to sample my TC module at that rate either.
Is there anything I can 'easily' do to decrease the sampling rate down to a low level? Everything I have tried doesn't seem to work and I don't really want to go away from SignalExpress.
Jay

Hi Jay
I am assuming that you are using the NI-9237. This strain module has only a specific set of sampling rates. If you specify a rate that is not supported it will corers the rate to the next higher sampling rate. Also since you are using the cDAQ chassis, there is only sample clock so it must choose the highest sampling rate. So it is not possible to sample at a lower rate. Please see the following link.
If you have access to LabVIEW then I would suggest taking your readings at the fast rate and then average the samples before you write them to your file. The following link has some more information on this.
If you are limited to Signal Express, this task is tricky. If you select a N-Sample acquisition, set the samples to read to greater than two, and set the post acquisition delay under the execution control tab to 1000, you acquisition will acquire the two samples at the very fast rate, but it will wait 1000mS between each acquisition. This will slow the overall sample rate.
Chris_K
National Instruments
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Audio sample rates

I am getting the message after capturing "The audio sample rate of one or more of your captured media files does not match the sample rate on your source tape......etc". The only thing I notice is in the Aud Format window the captured audio is "16-bit integer" and the sequence is "32-bit Floating Point". Should I ignore this or is there a way to get them to match?
G5 dual 2.7 Mac OS X (10.4.5)

Hi mmona,
I did a little bit of googling and reading about 32 bit floating and 16 bit integer, and I left a bit more puzzled and without enough knowledge to be dangerous.
However, the warning FCP is giving you is for the sample rate, as opposed to the audio format.
To see what I mean, highlight the clip and then hit Command-9. Audio rate and audio format are the 3rd and 2nd to last items. The warning is refering to the audio rate, but the difference you are reporting is in the audio format (which I remain curious about.)
I'm going to go out on a limb and guess that you are capturing DV that originated in a camcorder that cost less than $10K. In many cases, these camcorders capture unlocked audio. Google unlocked DV audio and you'll get a lot of hits, and many explainations.
I've gotten the warning you describe, but only with DV originating in certain less expensive cameras. Not all cameras are equal when it comes to this. The XL1, despite being a higher end camera in this class, often shows this issue.
Sometimes in these cases, when I look at the audio rate on the clip (3rd from bottom after hitting Command-9) it will report a number like 48.009. The practical effect is that the audio drifts slowly out of sync as the clip progresses in FCP's 48 KHz world.
So I keep my clips short and the problem remains invisible.
Tom

OLED display & signal sampling rate

Dear Ni technicians
I am new to LabVIEW Embedded module. I use LabVIEW 2009 and Luminary LM3S8962 evaluation board. Recently my project need to use this board for data acquisition and display the input the waveform on the OLED display. However I find that the sampling rate is as low as around 10Hz, hence I can only get the waveform below 2Hz without aliasing. While the datasheet claim that the sampling frequency is about 16MHz. Is it because the display rate "drag" the "sampling rate"? if so, how could I improve it?
Here I attach the code and the actual result in the ppt file.
Many thanks!
Best regards,
Haiwen
Attachments:
testing.ppt ‏4333 KB

Hi, I'm working on it, I wonder if at this time if you manage to increase the sampling rate?
Atom
Certified LabVIEW Associate Developer

Audio sample rate does not match (HDcam to dvcam)

I'm trying to import clips and keep getting this message:
"The audio sample rate of one or more of your captured media files does not match the sample rate on your source tape. This may cause the video and audio of these media files to be out of sync. Make sure the audio sample rate of your capture preset matches the sample rate of your tape."
Footage was originally shot on HDCAM and transferred to DVCAM elsewhere. Using FCP 5, am importing via firewire from a Sony DSR-11 deck. Using DV NTSC 48kHz Anamorphic as capture settings (though I've tried everything that I thought might possibly work with no success). The audio does not seem to drift over the course of several 5 minute or so clips. Clip settings show audio at 48 kHz (don't know if that's from capture settings or from actual data). Seems to me all audio should be 48 kHz 16 bits, so can't figure out what's going on. I have to export an EDL for the project to be finished in HD. Read some similar threads that ended in December, seemingly without much resolution. My broader concern is why this is happening; my immediate concern is do I need to worry about this right now since the media files will need to be recaptured in HD anyway. Any thoughts?
Thanks

A little more info. I'm having this problem on 4 tapes (from different cameras) that were transferred to DVCAM in a squished format to appear full screen on a 4x3 monitor. Video that was letterboxed and I can bring in with the standard DV NTSC capture settings does not have this problem. Still have the problem if I try to import the clips from the squished video with standard settings. Any thoughts?

Sample rate off the audio input and out devices do not match - what to do?

This is fundamental, I know, but nevertheless I can't find my way around it. I get this error message when trying to recor:
Sample rate off the audio input and out devices do not match
and am asked to do this:
Use the appropriate operating system or audio device control panel to adjust the sample rates of the input and output devices to use the same settingt.
I have defined the sample rate to 44.1/16 bit in accordance with my inbuild soundcardt.
I am trying to record from LineIn.
When running on a M-Audio sound card I don't face any problems.
HP 8560W, sound card IDT/High definition audio Codec
Any suggestions?
Knud
Copenhagen

You're sure you have set BOTH the input and output settings to 44.1 16 bit?
Which version of Windows are you running? There are a number of posts on this forum about how to fully access both the Windows Mixer and the Mixer for your soundcard. Especially, you need to ensure that all "Windows Sounds" are turned OFF.

Separate sampling rate for two different channels for a USB-6009 daq

Hi,
I am using a USB-6009 and incorporating the 'daq assistant' to change the sample rate. I am trying to find a way to set the sampling rate to two unique values for two separate channels. I've tried setting up two daq assistants and adjusting the sample rate different for each channel, though this does not work. Is there any way to set the sample rate high for all channels then reduce the rate for a different channel - or an alternative? I would appreciate any input on this, thank you!
- Anthony
Solved!
Go to Solution.

All tasks on a DAQ board that use a sampling clock must use the same clock. Therefore, you cannot have two tasks on the same DAQ board sample at different rates.
Alternatives would be:
1. to combine all of the channels into a single task and just accept the extra data
2. get an extra DAQ board to use in parallel
There are only two ways to tell somebody thanks: Kudos and Marked Solutions
Unofficial Forum Rules and Guidelines

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

Multiple sample rate question

I have one project with edits of different files all over the place recorded at 44.1 kHz. I have another project with edits all over the place recorded at 88.2 kHz. I wish to combine all tracks from these two projects. I realize I will need to convert the sample rate of all the 88.2 kHz files, but I don't want to mix down any track. I want all of the edits to be cut at the same places in time, just with the referenced files being at a different sample rate.
I've tried everything. It seems that it is impossible. If I convert the sample rate of the referenced files, then open the project and change it's sample rate, the edits are now cut at differnt locations in time. It seems I have to mix down each 88.2 track, then sample rate convert these to 44.1.
But I need to have all the edited segments as they appear in both projects in the same project. Any ideas?

Don't record at 88.2! Why would you do that?!
I'd back up and archive your current session first as you are probably going to have to do some very destructive messing.
I haven't put this to the test but maybe you could try converting the route files in a third party software like ProTools. You'll get a bunch of prompts when you reopen Logic but this may work. But for me the simplest (albeit time consuming route) would be to convert all of your regions to new original sound files (you can do this from the audio pop down on the arrange page). Then in your audio pool select all unused audio, delete and go through each new file and convert the sample rate. It's a bit of a car crash situation needless to say next time preparation preparation preparation!
Finally when you have been through this kind of process make sure that you are operating at the correct new clock rate before continuing otherwise this stuff still aint gonna work.
Hope this is of help.
Good luck.

IIR Filtering and response .vi: Butterworth filter magnitude response depends on sampling rate -why?

Hi folks,
I am not expert in filter design, only someone applying them, so please can someone help me with an explanation?
I need to filter very low-frequent signals using a buttherwoth filter 2. or 3. order as bandpass 0.1 to 10 Hz .
Very relevant amplitudes are BELOW 1 Hz, often below 0.5 Hz but there will be as well relevant amplitudes above 5 Hz to be observed.
This is fixed and prescribed for the application.
However, the sampling rate of the measurement system is not prescribed. It may be between say between 30 and 2000 Hz. This will depend on whether the same data set is used for analysing higher frequencies up to 1000 Hz of the same measurement or this is not done by the user and he chooses a lower sampling rate to reduce the file sizes, especially when measuring for longer periods of several weeks.
To compare the 2nd and 3rd order's magnitude response of the filter I used the example IIR Filtering and response .vi:
I was very astonished when I the found that the magnitude response is significantly influenced by the SAMPLING RATE I tell the signal generator in this example vi.
Can you please tell me why - and especially why the 3rd order filter will be worse for the low frequency parts below 1 Hz of the signal. I was told by people experienced with filters that the 3rd oder will distort less the amplitudes which is not at all true for my relevant frequencies below 1 Hz.
In the attached png you see 4 screenshots for 2 or 3 order and sampling rate 300 or 1000 Hz to show you the varying magnitude responses without opening labview.
THANK YOU for your ANSWERS!!!
chris
Solved!
Go to Solution.
Attachments:
butterworth-filter-differences.png ‏285 KB

Hello Lynn,
thanks for the answer. You are right that there are few points "behind" the curve in the graph, see png.
However, this is the filter response which Labview (2009) provides to me directly out of the "IIR Filter for 1 Channel. vi" in the "filter information" output cluster. Where up to now I do not know how to influence it - apart from adjusting the input parameters "IIR filter specifications". OK, I assume I have to gain more knowledge of this. The curve of the magnitude resonse dies not change when I change the number of samples of the input signal of the signal generator, only wehn I change the sampling rate.
I used directly the example vi from Labview with the name indicated in my first post "IIR Filtering and Response.vi".
So I assumed that everybody has it in his/her examples shipped with LV and it is not necessary to post it.
I just adjusted the size of the diagram of magnitude response to see the curves better as you see in the attached vi.
So I did no changes to the vital parts of signal generation and filter of the example. The screenshots are like they come from the example when using the option "one waveform" where I as user assume that this which is behind is quality-controlled by NI.
I was also astonished that the filter magnitude response is different to the one I copied out of graphs 1 year ago - but I unfortunately cannot reconstruct which example I used there...
Thanks for any further comments
chris
Attachments:
IIR Filtering and Response_CH.vi ‏55 KB
butterworth2nd_order_bandpass_0p1to10Hz_mag_response.PNG ‏18 KB

Sampling Rate- How do I check and adjust the sampling rate on my Laptop?

For the purposes of matching the sampling rates on my laptop to my usb microphone, (recording in garageband) how do I find out what the sampling rates are on my 2009 macbook? Many thanks!

GarageBand '11: Set the audio resolution: http://support.apple.com/kb/PH1873

How can I programmatically determine the capabilities of a card under NI-DAQmx (e.g. max sample rate, number of AI/AO/CTR channels, etc.

Is there a DAQ_Get_Device_Info() equivalent for NIDAQmx? I need to iterate thru all the devices on my system, and build up a list of device capabilities. The system may include M-series and E-series cards.

Attached is a program I've used in the past to determine number of AI channels. It could be easily modified to check for AO or digital or counter. Also, there is a ton of properties that you have access to (i.e. max sample rate, max/min voltage inputs, etc.) that are accessed as properties of the type of channel, or timing properties, as opposed to properties of the board. Check out the DAQmx C Reference Help (Usually at Start>>All Programs>>National Instruments>>NI-DAQ). Expand the NI-DAQmx C Properties, and look at the List of Channel Properties, and Timing Properties, etc.
-Alan A.
Attachments:
Device_Info.vi ‏25 KB

Creative Audigy 2 NX Bit Depth / Sample Rate Prob

This is my first post to this form
Down to business: I recently purchased a Creative Audigy 2 NX sound card. I am using it on my laptop (an HP Pavilion zd 7000, which has plenty of power to support the card.) I installed it according to the instructions on the manual, but I have been having some problems with it. I can't seem to set the bit depth and sample rate settings to their proper values.
The maximum bit depth available from the drop down menu in "Device Control" -> "PCI/USB" tab is 6 bits and the maximum sample rate is 48kHz. I have tried repairing and reinstalling the drivers several times, but it still wont work. The card is connected to my laptop via USB 2.0.
I looked around in the forms and found out that at least one other person has had the same problem but no solution was posted. If anyone knows of a way to resolve this issue I would appreciate the input!
Here are my system specs:
HP Pavilion zd 7000
Intel Pentium 4 3.06 GHz
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Windows XP Prof. SP 2
Thnx.
-cmsleimanMessage Edited by cmsleiman on -27-2004 09:38 PM

Well, I am new to high-end sound cards, and I may be misinterpreting the terminology, but the sound card is supposed to be a 24bit/96kHz card.
I am under the impression that one should be able to set the output quality of the card to 24bits of depth and a 96kHz sample rate, despite the speaker setting that one may be using, to decode good quality audio streams (say an audio cd or the dolby digital audio of a dvd movie.) I can currently achieve this only on 2. speaker systems (or when i set the speaker setting of the card to 2.) Otherwise the maximum bit depth/sample rate I can set the card output to is a sample rate of 48kHz and a bit depth of 6bits.
Am I mistaken in thinking that if I am playing a good quality audio stream I should be able to raise the output quality of the card to that which it is advertised and claims to have?
Thnx

High sample rate data acquisition using DAQ and saving data continuously. Also I would like to chunck data into a new file in every 32M

Hi:
I am very new to LabView, so I need some help to come up with an idea that can help me save data continuously in real time. Also I don't want the file to be too big, so I would like to crete a new file in every 32 mega bytes, and clear the previous buffer. Now I have this code can save voltage data to TDMS file, and the sample rate is 2m Hz, so the volume of data increase very fast, and my computer only have 2G ram, so the computer will freeze after 10 seconds I start to collect data. I need some advise from you briliant people.
Thanks very much I really appreciate that.
Solved!
Go to Solution.
Attachments:
hispeedisplayandstorage.vi ‏33 KB

I am a huge proponent of the Producer/Consumer architecture. But this is the place I advise against it. The DAQmx Configure Logging does all of it for you!
Note: You will want to use a Chart instead of a graph here.
There are only two ways to tell somebody thanks: Kudos and Marked Solutions
Unofficial Forum Rules and Guidelines
Attachments:
hispeedisplayandstorage_BD.png ‏36 KB

Error message trying to synchronise audio and midi sample rate 42463

I have a Rode Podcaster mic that is terrific. It has been working fine in GB. suddenly I am getting this when I try to record my podcast:
'Error message trying to synchronise audio and midi sample rate 42463 recognised. Check conflict between Garageband and external device'
Has anyone experienced this before? How do I resolve it? I'm still using my Version 6.0.5.
Apple, is this a way of forcing me to upgrade to V10?

Just click the icon on each track that stops it acting as a live input. It looks like a sound wave. Only have them clicked orange when using the track to record! I don't know if that will solve your problem!

Sample rate is 48.009.2Hz...???

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