Wrong sampling rate from Sound Input Read VI on Dell Latitude E6510

Hi,
When I
receive data from the sound card with the Sound Input Read VI, the actual
sampling rate is not equal to the sampling rate set in the Sound Input
Configure VI. I estimate the sampling ratio by dividing the number of samples received
by the difference between the first and the last t0 from Sound Input Read. For
example, if the sampling rate is set to 7000 S/s, the actual sampling rate is
around 6900 S/s. If the sampling rate is set to 7050 S/s, the actual sampling
rate is around 7200.
This problem appears when the
code is run on my Dell Latitude E6510 machine. When I run the
same code on my Lenovo ThinkPad, the sampling rate is correct.
The Labview application I
am developing is to be used to acquire data synchronously from several sources,
including the sound card input. It is therefore important that the sampling rate
is correct.
If anybody has any idea
about what may cause this problem, and a way to fix it, I would be grateful.
Ole

It has been a trend the last years that sound card only support some sampling rates, not a range. You should refer to your sound card manual for this kind of information. You should also be aware of that the NI sound interface is junk, and they do care about it either. Use the WaveIO interface instead See the link. But I am quite sure your problem is related to the sound card hardware not some NI driver issue
http://www.zeitnitz.de/Christian/waveio
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)

Similar Messages

PLEASE can a AE from NI take a look at my problem. Sound input read behave in strange manner then the buffer size is larger than 2X number of samples to read.

On my computer I have discovered some strange behavior then reading data from the sound card. Then the buffer size is 2x samples to read everything is as expected. But since I read the sound card 10 times pr second I feel a .2 second buffer is to small. I am using XP, and XP is not a RTOS so with a buffer set to 0.2 seconds I may lose data. Therefore I set the buffer size (number samples/ch on Sound Input Configure.vi) to be in range of 2 seconds. The result then is that then reading from Sound input.vi, a reading often take more than 0.1 second. On my computer it is often 500mSec. Then the next 5 read follows with almost zero interval. I do not loose data. But on my front panel the graphs looks like an very early silent movie. This error was introduced in Labview 8.x. To be honest I think the labview 7.x sound system was much better in many ways.
But before I point any finger NI. Other people has to verify the behavior I experience. I have made an example showing this error. It is a modified version of the "Continuous Sound Input.vi" example. Then the "buffer in seconds" control is set to 0.2 every thing works OK. Changer this to a larger number will produce the mentioned above hiccup. The larger number in this control the larger hiccup. Is it any way to fix this? My solution up to now has to use a free 3. part software(http://www.zeitnitz.de/Christian/index.php?sel=waveio) But I guess it soon will be outdated. It may not work with newer windows versions.
Any help at all will be appreciated
And yes I have the most updated version fo DirectX. Also I se this in Labview 2009 which I have trail version of. The VI I have made is in 8.6
Message Edited by Coq Rouge on 09-07-2009 10:54 AM
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)
Attachments:
Continuous Sound Input with timing.vi ‏23 KB

macaba wrote:
If you take a moving average of the 0.2s buffer vs. 3s buffer at an update rate of 10, then they are the same (just under 100ms), so the average refresh rate is the same. I agree that is odd behaviour that the time between sound reads go to zero quite a lot then take a long time once in a while (presumably to fill the buffer
I guess it goes to zero because it is reading data from the buffer it do not has to wait for data from the sound card. The mysterious thing is the periodic delay. You are also correct then saying that average timing is correct. And in my application I have no data loss.
If you search for sound in this forum you will find out that many people has reported trouble with the sound system.
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)

Sound Input Read VI hangs

I'm using the SI Read VI to sample from the Soundblaster card in my Dell PC. I've configured the device for 16-bit mono input, sample rate 44.1 kHz, and the buffer size is 1024 samples. The SI Read VI is in a Timed Loop whose period I've set to 23 ms (1024 / 44.1kHz). Following each acquisition, I do some filtering, frequency analysis, and plotting inside the loop.
The acquisition works well for the first 10 minutes or so that it is running. But sometime after about 10 or 15 minutes, the VI hangs. I've traced the hang back to the SI Read VI, but have not yet figured out how to avoid the hang or recover from it (I have to abort and restart the program).
Has anyone had similar problems with the SI Read VI? Could there be a problem with the DLL that the SI Read VI calls? Are there any upgrades or fixes for this? Can someone suggest a way to recover from the hang?
Thanks for any help,
Mark

This message is a reply to "Sound Input Read VI hangs" (National Instrument) NI Discussion Forums posting found at: http://forums.ni.com/ni/board/message?board.id=170&message.id=137258 and http://forums.ni.com/ni/board/message?board.id=170&message.id=110656&requireLogin=False
Hi everybody,
From the large count of how many of you guys read the above thread of messages, including myself, and who are dealing with the same problem of SI Read VI hanging found National Instrument (NI) LabVIEW (LV), I decided to share my (not so perfect) solution with you guys to save you time and frustration. Here is the story of 3 days (18 hrs/day) of my life, fighting a bug in SI Read VI (SI: Sound Input) (I am currently using LabVIEW Pro v 7.1.1):
For the reason that "A Message cannot exceed 10,000 characters" on this Forum I included my message in the attached Word doc. Hopefully, there is no such limitation on uploaded files!
Samir Berjawi
Research Assistant and Lab Instructor
American University of Beirut
[email protected]
[email protected]
Attachments:
LabVIEW SI Read freezes - Report.doc ‏39 KB
Sound Acquisition Test.zip ‏995 KB

Time mismatch with Sound Input Read VI

Hi Folk,
I am acquiring the signal of the PC
sound card with the "Sound Input Read VI" and I have
noticed that between subsequent waveform data packets there are time
mismatch, both overlap and gap.
To point out the observed behavior, I
have posted a modified example, the "Continuous Sound Input.vi". In the example, I have computed the
time difference between the t0 of the actual waveform packet and t0
expected on the basis of the previous waveform packet.
Consistently, in the indicator "Time
series" (Waveform Charts), each time there is a time overlap or
gap the Charts resets it self or presents a gap.
By reading 1 second of data the time
mismatch is about 0.015625 or 0.03125 sec. (both positive and
negative).
The repetition frequency of the time
mismatch decreases as the acquisition sample rate increases.
The amount of the time mismatch seems
to be sample rate invariant.
Do you have any idea from where this
problem is coming out and how to solve it?
Thanks for your help,
Asper
Attachments:
Time mismatch with Sound Input Read VI.png ‏51 KB

Pre made Labview functions are not some holly grails that is newer to be touched and modified. In fact many included functions in Labview has what I will name as "high flimflam factor" That will say a lot of functions you do not really need. Express VIs are grim examples of this.
Anyway I have made some modifications and removed some babyfat in the sound input VI. Take a look at it. The top level VI is the "time fixed sound.vi" It could be that you will get an error because Labview will not find a DLL. If a DLL is reported missing you will find it in C:\Program files\National Instruments\LabVIEW 2010\resource\lvsound2.dll
Remember to save the modified VIs in a separate folder, and not in the vi.lib folder at all. Be careful so you do not overwrite any Labview function
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)
Attachments:
Sound Input Read (DBL)_time_fixed.vi ‏30 KB
time fixed sound.vi ‏19 KB

How to have Sound Input Read VI read .wav in sections

Hi,
I'm trying to read audio data from my laptop's soundcard, and it's really slowing down my VI's processing power. I read on the NI website to read the .wav file in sections instead of as a whole, but I have no idea how to do that. Any suggestions? My VI is attached.
Thanks!
Attachments:
signalgen.vi ‏56 KB

You have set the sample rate for the sound card to 100000 samples pr second. That is a sample rate your sound card will not be able to handle. Hence a error will occur. And you will not get any data from the sound card. Some sound cards may support samplerates in 1 Hz step but most sound cards do only support some rates. The latter is most typical. All sound cards do support 44100 Hz sample rates but typical sample rates are 44100,22050,11025, and 8000. Some sound card do also support higher sample rates than 44100. Refer to your sound card documentation for this. Your way of using the Daq card by reading 1 sample on each iteration, using software timing is also a Daq NO-NO. Read data in chunks and do not do to many updates pr second, and always use HW timing. If you need the waveform to appear "live", or "real-time" 10 updates pr second is more than enough.
I have included a sample vi. I suggest that you use that as your workhorse from this point. At least if you want more help from me
Tip Labview is shiped with many very useful examples. Then you are stuck. Go to toolbar->help->fFind examples. As one example the Daq setup is from the "Cont Acq&Chart Samples-Int Clk.vi" vi example
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)
Attachments:
workhorse.vi ‏37 KB

I'm trying to record in Adobe Audition and it keeps saying my the sample rates for my input and output devices do not match. How do I correct this?

I'm trying to record in Audition and it keeps saying my sample rates for the input and output devices don't match. How do I fix this?

I finally have communication between the ISA One and Audition. I moved the optic cable to another SPDIF on the computer and from the SPDIF to ADAT on the ISA One and it finally agreed that the communication was at the same clock speed. The audio was not intelligible so I moved the optic cable back to the original configuration and I can record voice that is clean.

Stft from sound input VI dat to plot spectrogram, whats wrong?

Hi everybody,
I listen continously
via a microphone and sound card to 25ms long pulses in the range of
9kHz to 12kHz. I get 10000 samples from my Sound Input VI and use the
STFT Spectogram to plot a spectrogram(frequency versus time). Using the
default settings or any other window lengths results into the error
message:"...possible reasons: invalid windows length".
What's wrong?
Thank you very much,
Martin

Hey Brian,
I would like to listen to 25ms long pulses in the range of
9kHz up to 14kHz
+by using a microphone and a standard PC sound card
+continuously up to 1.5 hours.
The VI should display a real-time spectrogram(frequency versus time) like in the example VI "Moving STFT PtByPt.vi". It is important that the spectrogram always shows the last 5 minutes and the current frequencies. It is not necessary that the spectrogram is very real-timed(up to 600ms delay is ok), BUT it has to be 100% CONTINUOUSLY to not miss any of the short 25ms long pulses.
In addition I also would like to determine the frequencies of the received pulses, which are somewhere between 9kHz and 14kHz and I need to count the amount of received pulses.
My issues are:
1. missing some 25ms pulses due to problems with continuous sampling of SI Read VI in a while loop(I guess there is a small break between the 1000 samplesets)
2. How connect SI Read VI output to STFT VI or STFT PtByPt VI to perfom moving spectrogram?
3. Why doesn't show my spectogram no frequenciues at all?
I hope this helps to understand what I try to achieve and in what problems I have run.
Here is a corrected and more use-friendlier version of my VI.
Thank you very much,
Martin
Attachments:
spectrogram_NI.vi ‏142 KB

Can't change sample rate for digital input on Mac Pro

Hello all,
on my Mac Pro with 10.4.10 I can't change the sample rate for the digital input. Whenever I choose 48000 or 96000 Hz it returns to 44100 Hz after a few seconds. Feeding a 24 bit/96000 Hz signal from an external ADC into the optical input doesn't help. No input signal is available for digital recording software such as Sound Studio or Cubase. After changing the sample rate in audio midi configuration to 96000 Hz the sound can be heard for a few seconds but when the setup returns to 44100 automatically the signal is, of course, lost.
I've deleted all relevant preferences and restarted with resetting paramter RAM. Still the same. With external hardware such as M-Audio Firewire equipment setting the sample rates works properly.
Thanks to all for helpful clues.

Hi,
when E&M signaling is configured on digital interface like the VWIC is, 2 or 4 wires operation is not applicable because there are no wires at all, and reported only for compatibility with the analog E&M card.
Consequently, you cannot configure that and it will not make any difference to effect of the connection.
Please rate post if it helps!

Samples rates for sounds

I am thinking of purchasing some sounds from sounddogs.com, they offer their sounds in sample rates of 44.1 kHz and 48 kHz. Obviously 48 kHz is better, but would there be much of difference between 48 kHz and 44.1 kHz?

As David says, 48kHz is better but you probably wouldn't notice the difference. Almost everything I have ever used has been 44.1kHz and I've never had a problem.
If you are buying them it makes sense to opt for 48.

Sound input read multiple soundcards

Heey everybody,
I have a problem. I want to read two different signals at the same time from 2 different soundcards. 1 internal soundcard from my laptop, and 1 usb soundcard.
So i used the Acquire Sound VI.
Reading both signals at the same time is not a problem until i set the measuring time to about more that 30 seconds. It gets a buffer overflow.
So i tried to do the following:
I first made this with 1 signal first, and it worked perfectly, every second it got updated.
But, when i added the second signal it only wanted to read the first second of the first signal and then it got stuck, and i had to stop the software by closing it. And then killing labview with the taskmanagement thingey from windows.
Anyone have any idea what the problem is?
Greetings,
Jory
Solved!
Go to Solution.

Ok, never mind, i've solved it. I made the buffers larger (at the sound input configure block) and now it works like a charm

Sound Input Read Spectrum Analyzer

I am working on a project with Arduino to do a spectrum analyzer. The LVIFA part of it is working fine.
The part that I am struggling with is the audio input.
I am using the Sound Input Pallet.
My troubles are two-fold.
1. A single read takes quite some time and I need to have a pretty quick response. I decreased the samples per read and came up with problem #2.
2. I can get one read out of the Sound Read In and then the VI locks up. I have my timeout set to 1 Sec and it never times out. I have to restart LabVIEW.
While it would be nice to have this VI working, I'm not even sure that I am using the correct tool. If I am going to be running an audio spectrum analyzer display off of the data input, I need to response to be within a few mS (ie. <200) so that it looks visually correct.
If this library will not achieve this, then I need to look into a different tool.

Hi,
I have a similar project, actually I am using a Audio frequency range as input to the Arduino and do an FFT to receive the spectrum out of it.
See the YOUTUBE video I did and the code in *.jpg. I took actually the Continous Input example and modified it.
Hope that helps.
Keep me posted, since we might work on a same project.
http://youtu.be/aqzyofQHXDM
Y3G
Attachments:
Uno Blockpanel.JPG ‏104 KB
Uno FRONTpanel.JPG ‏99 KB

How do I change the audio sample rate from 48kHz to 44.1kHz for Mpeg2

Hey all,
I've been searching for a while but haven't found any direct answers in the forums or the user manual so here goes.
I have to dispatch a file to Bloomberg TV and the file specs they have given me are as follows:
Video Standard: MPEG-2, MP:ML, 4:2:0
Frame Rate: PAL 25fps
Video Size 720 x 405
Aspect Ratio 16:9
Audio Standard MPEG-1 Layer 2
MPEG-2 Program Stream Mux rate 6mbs per second
Bit Rate Type: CBR
Video Bit Rate 5.7mbs
Audio Bit Rate 192kbs
Audio Sample Rate 44.1Khz
Interlacing: Upper Field first (why they want interlaced for web streaming is beyond me)
GOP Structure: IBBP
I-Frame distance 12
Now everything above is fine except the audio encoding because even though I have set up a new setting from scratch I cant find anywhere to adjust the audio sample rate. The Inspector tells me the Audio encoder is set to:
Format: MPEG
Sample Rate: 48.000kHz
Channels: 2
Bits Per Sample: 16
Anyone Know how or even if I can change these audio settings? The only adjustments I can find are the filters or the transport/program stream option. I have it set to program as specified by Bloomberg.
Thanks in advance
J

The only setting that I could find in compressor that lets your change the bitrate to 44.1 is when you create a new dolby digital setting and then under the inspectors audio tab/Target System button, change the button to Generic AC-3. When done, you can change the Sample Rate to 44.1.
Hope this helps?

Help please? Logic playing back wrong sample rate

So I created a new session and recorded a new song that I have already done some work to this song and have had no problems whatsoever. Last night I went to open this song and work on it some more and when I pressed play the song started playing back very slowly. This session was recorded at 88.2k and I am getting a Logic error message that says sample rate 44.1k is recognized however my audio interface (a Metric Halo 2882 +DSP 2d unit) is set correctly to 88.2k. I look at my bin and all the files are correct, my preferences are all correctly set at 88.2k but Logic is playing the song at about half speed!!
What happened?? I have a couple projects going and I need to get this fixed.. Anyone??
Thank you in advance!!
DDD
Mac Book Pro Dual Core, OSX 10.6.8, Logic 9.1.5 (do not want to upgrade Logic to 9.1.6 I tried that and it was a BAD experience so Im sticking with 9.1.5), Metric Halo MIO Console 5.4

Sounds to me that your best bet is to use a utility like SoundHack (http://music.ucsd.edu/~tre/soft//SH893.hqx) to change the audio file headers. What you want is to flag the files as 48k, not do any sample rate conversion--they should play back natively at 48k. The only SRC you'll need to do is if you actually do want the files to end up as 44.1k at some point. (In your case, I don't think Logic is converting the sample rate in real time, by the way--what it's doing is playing back the files with a clock setting of 48k, which only sounds right because they've been mis-labeled.)
Either way, you should fix the file headers now, I think. And even if your final product needs to be 44.1k, since your files are quite clearly at 48k now, I'd do all the necessary editing and processing at 48k, then convert to 44.1k as the final step. And until the SRC in Logic is improved (Apple ?), maybe use some other app to convert to 44.1k in the end. Peak gets high marks--not sure what else out there sounds good and doesn't cost a mint.
Best luck!
James
[email protected]

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!

G5 sound input reads drums but no sound coming out

i'm not sure what the deal is, the input is reading my drums fine but i hear no sound not through headphones or surround sound or computers speakers. All my sound works for anyhting else. please help me

My present computer has no sound input capabilities but there may be a setting in sound control panels or whatever software you are using for selecting sound output, including passthrough. There's also a midi application in your applications that controls sound features.

Wrong sampling rate from Sound Input Read VI on Dell Latitude E6510

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