How To Lower Sample Rate

Is it possible to adjust the sample rate lower than the default when attempting to record an audio source?

44.1 is the lowest sample rate to record audio at in Logic. But put a Bit crusher plugin on the track and change the resolution there for more lo-fi sounds.
JG

Similar Messages

  • Low sample-rat​e measuremen​ts on the PCI-6115 DAQ card

    I need to measure an analog signal at a sampling rate of a few tens to hundreds of Hz in sync with the rising edge of an external clock. I have a PCI-6115 DAQ card w/ Labview 6.1 and NI-DAQ 6.9.2. The PCI-6115 is a high speed card and has a minimum sample rate of 10 KS/s. Is there any way of implementing a low sampling rate measurement using the PCI-6115 in sync with an external clock?
    Thanks in advance.

    Kuldeep,
    It is possible to do what you are describing above (in fact I don't think an external clock is required to do this), however, bear in mind that the reason for this minimum sampling frequency is due to the ADCs used on this high speed board. The ADCs used are pipelined ADCs, meaning that when a signal is digitized, it is digitized in distinct stages within the ADC (in the case of the 6115, I think there are 3 stages involved). Data is moved from one stage of the ADC to the next each time a sample clock pulse is recieved. If too much time elapses between these clock edges, the signal to be digitized can actually 'leak' off of the ADC. This can result in improper digitization, which can lead to less accurate measurements. So, while it is possible to mak
    e the device sample below it's minimum rate, it may be advisable to sample faster than the rates required by your application, and either average multiple data points per measurement, or throw away extra points taken.
    I hope this helps,
    Dan

  • Counter Problem at low sampling rate

    What I am trying to do is to count the input from the camera and generate the number of pulses depending on the delay and interval. At the rate of 33ms the program works perfectly but if the sampling rate is 150ms it generates 2 pulses at a time. Is there a better solution to my problem ? Am I doing anything wrong ? Is it a problem with the loop timing or acquisition timing? I have attached the program. What puzzles me is that it doesnt work if the rate is lower! Please HELP.
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    Attachments:
    CoolSnap.vi ‏142 KB

    Hello sha33,
    Thank you for getting back to me with more information. After looking through your code once more, I believe I have found the issue. Since this application is timed entirely in software, it is based on how quickly your operating system can run the program, and thus cycle through loop iterations. Since you have no Timing VIs in the program, this loop is running as fast as possible. Now, in the 33 ms case, you are getting a TTL edge a lot faster, so the DAQ assistant is updating its count faster as well. Lets say hypothetically that your loop is iterating at 60 ms. This means that every time you run through the loop your DAQ assistant will read the updated count value, and since you are counting every 33 ms, your count value will have updated.
    Now let's look at your 150 ms TTL signal. This is significantly longer than the 33 ms signal in the previous case. Lets say hypothetically your loop is running on average at 60 ms again. This time, your edge will come in and the counter will increment, generating a pulse with the second DAQ assistant accordingly. Now your loop will run through again 60 ms later but your next edge on the 150 ms signal is still around 80-90 ms away. This means that your DAQ assistant will return the same count value as the last iteration, and once again a pulse will be generated.
    Obviously the numbers I used for the loop iteration timing are hypothetical and entirely dependent on the operating system and its resources (i.e. it will run slower with an MP3 or Movie playing in the background). My suggestion would be to use a timing VI such as Wait Until Next ms Multiple to clock your loops more consistently. The goal would be to make sure that your loop is repeating faster than your counter is updating. Alternatively, you could account for the possibility of no counter input in your overall application to prevent a false positive.
    Let me know if I can clarify any of the above information, or if you have any additional questions related to this issue.
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  • Lossless to lower sampling rate automatic?

    I use Apple lossless for many of my imported songs. Will the iPod touch offer to lower the sampling rate on the fly? My shuffle used to do this.

    Okay, I tried to bite the bullet and include lossless ripped music files in a synch. I did a SmartPlaylist and limited it to 10GB taken from my general Music Playlist, not played in 90 days. It is now synching a little over 5.6GB in lossless and the rest of the 10GB 128/256. All told 1,235 songs.
    It is taking forever to sych. Pardon the pun, but this just about rips it I've got to downgrade the my lossless library which comes from CD's I own. I'm reducing the lossless files now. Hope this works without mucking up the sound too much.
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  • How to adjust sample rate in "capture preset"?

    I'm having to recapture footage shot some years ago on my old Sony VX1000 using my Sony Z5U. I'm working on FCE 4.0 (OS 10.4), Easy Setup at DV-NTSC and capturing by "capture now." I'm getting a warning saying that my sample rate of capture does not match sample rate of source tape and advises me to ensure that my audio sample rate of the "capture preset" matches the sample rate of [my] source tape. Browser tells me that my sequence and my clips audio rate are both at 48 KHz, but my sequence audio format is 32 bit floating point, and my clips are at 16 bit integer. How do I access capture preset and synch capture rate?
    Thanks

    48KHz = 16bit
    32KHz = 12bit
    Can't be both at the same time.
    Most cameras come set to 12bit as the factory default. On the Canon miniDV cams like your ZR500, go into the menus for *Camera Setup > Audio Setup > Audio Mode* and you should be able to toggle between 16 bit and 12 bit audio.
    You want to film your videos using 16bit audio if at all possible. If you have done so, then you should use the DV-NTSC easy setup in Final Cut Express (not the DV-NTSC 32KHz easy setup).
    If you inadvertently filmed your video using 12-bit audio then you will need to use the DV-NTSC 32KHz easy setup in FCE.

  • Low sample rate with NI USB-6008

    Hello,
    We are working on a project collecting data from a hybrid rocket motor for
    University research. We are using a NI USB-6008, which has a sample rate of
    10kHz. We are using 3 channels and getting a sample rate of ~60 samples per
    second. I have attached the VI. Is this slowed by software, or physical
    computer computation speed? This sample rate needs to be much higher. Any help
    is greatly appreciated.
    - Theo
    Attachments:
    Hybrid DAQ_10.vi ‏73 KB

    Hi Rocket 154,
    You’ve already had some
    great responses, but I just wanted to add a little bit. A great starting point
    to writing good code would be the Getting Started with NI-DAQmx: Main Page,
    as well as the shipping examples (In LabVIEW, Help » Find Examples) already
    mentioned.
    If you navigate a bit
    through the links (Basic Programming with NI-DAQmx » Getting Started with NI-DAQmx
    Programming in LabVIEW) you’ll end up with some great resources. I recommend Learn
    10 Functions in NI-DAQmx and Handle 80% of your Data Acquisition Applications.
    It will detail many of the functions you can use and give you a feel for proper
    code style in LabVIEW.
    I hope that helps.
    Mark E.
    Precision DC Product Support Engineer
    National Instruments
    Digital Multimeters (DMMs) and LCR Meters
    Programmable Power Supplies and Source Measure Units

  • Why LV 6.1 and Nidaq 6.9.3 can't acquire low sampling rate and high frame size

    My platform is LV 6.1, Nidaq 6.9 on win98
    Everytime i want to acquire data with sampling rate 6 Hz
    and frame size 7200, it acquire just 2 or more data (under 10 data), never 7200 data.

    Greetings,
    I assume by "frame size" you are referring to the numbers of samples to acquire. Please launch the "Find Examples" browser in LabVIEW 6.1. Open the example "Acquire N Scans.vi." This VI has only 4 inputs. After setting your device and channel you will set number of scans to acquire to 7200 and your scan rate will be 6. Since you're only acquiring at 6Hz and wanting 7200 samples, this VI will take 20 minutes to run. At that point the graph will be updated with your data.
    Regards,
    Justin Britten
    Applications Engineer
    National Instruments

  • How to improve sampling rates in mega samples spartan 3e board

     I am trying to acquire data in spartan 3e board in mega samples rate can you post me any examples for this purpose
    Solved!
    Go to Solution.

    If you're trying to get analog samples in that range, the issue that you're going to run into is that that sort of sampling rate is right at the bleeding edge of what you're asking for (it's a 2-channel @ 1.5Msps chip, the LTC1407A-1). If you actually want to capture at these rates, you're going to need to pull some interesting LV-foo using single-cycle timed loops in the FPGA diagram or drop some code down into an IPNode/CLIP Node, with outputs from those IP blocks hooked to the IO in a SCTL.
    Please let us know a bit more about your specific design.
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  • How to adjust sample rate of data?

    I have some data collected at 1683 Hz (yes, that was what I had!) and would like to reduce the sampling rate to some meaningful number, say 1024, 500, 400, or similar.
    What should I do?

    Well, the calculation is the approximation of your channel (variables with index 0) to a new one (index 1).
    The freq(0) and freq(1) are the sampling frequencies for the channels for the case you have waveform channels.
    The n(0) and n(1) are the numbers of the data points inside the channels. The new created channel should have the number n(1), calculated from n(0) with regard to different sampling ratios.
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    In short, here you copy the properties from the "old" channel and paste them to the new one.

  • NI 5761 with low sampling rate

    Hi,  is it possible to change the NI 5761 sampling rate without programming FPGA? Thank you very much!

    Hi, as mentioned above you can play with the sample rate by using an external Sample clock. Take a look at the picture attach for the general idea. Simply replace the Boolean buttons for digital input ports. I hope this helps
    Alejandro | Academic Program Engineer | National Instruments
    Attachments:
    Capture.JPG ‏256 KB

  • Lock-in amplifier, low sample rate options

    Hello!
    I would like to implement a lock-in amplifier in labview (version 8.5).  I understand that there is a developer kit for the LIA available, and have checked it out; however, it is incompatible with my equipment.
    I have a usb 1608fs card that I am using for for digital to analog conversion, which has a max sample rate of 16000 hz for two channels (signal and reference).  The signal I am trying to lock in on is between 5 and 10 hz.  The main contribution to the noise is 60Hz noise from the air, which shouldn't be a problem.
    However, the lock-in amplifier by labview refuses to work at under 50KHz sample rate, so I am at an impasse.  I could put one together myself, but I wanted to see if there were any options available to me; no point in reinventing the wheel.
    Regards,
    MB

    Here it says the minimum rate is 50kHz. Sorry, but ther is no option to change that.
    Also look at Lock-In Amplifier Startup Toolkit Compatibility with National Instruments Hardware.
    Btw, there is no USB 1608fs card from NI.

  • Resample DASYLab Data to Lower Sampling Rate

    Hi.
    My hardware has built-in antialiasing filters at 20 kHz so I'll be scanning at 40 kHz minimum. 
    My needed data range is low pass at 1 kHz. 
    Can DASYLab resample the 40 kHz input to "decimate" the signal to the needed 1 kHz input? 
    40x less data would be very helpful.  And if the data can be reduced, is this process CPU intensive (would it impose a large computational load on my computer)?
    I prefer to use DASYLab but does LabView have this interpolation/decimation capability? 
    Thanks,
    Bob

    signal1,
    In LabVIEW there are several VIs (functions) that can be used to resample a signal.  Among these are the Decimate.vi and the Resample Waveforms.vi.  The Decimate.vi averages values to obtain the resampled waveform while the Resample Waveforms.vi employs interpolation.  You can view the help for these VIs at http://zone.ni.com/reference/en-XX/help/371361A-01/lvwave/resample_waveforms_cont/ and  http://zone.ni.com/reference/en-XX/help/371361A-01/lvanls/decimate/.  You can get the specific details about DASYLab from DASYLab Online support found at http://www.dasylab.com/content/mydasylab.php?action=start.
    Regards,
    Neil
    Applications Engineer
    National Instruments

  • How to know the sampling rate for NI6624?

    Dears,
    I am trying to measure a transient signal that is a time-vary counter train.  The target frequency is increased from 0 Hz to 50 Hz when the measurement time rises to 1 s from 0 s.  The NI 6624 card and the LabVIEW DAQmx have been adopted.  In the block diagram, the terminal of measurement method “Low Frequency with 1 Counter” is set in “DAQmx Create Channel (CI-Frequency).vi”, and the “Finite Samples” mode is chosen in “DAQmx Timing (Implicit).vi”.  Then the transient signal points (increasing-frequency points) will be got successfully within 1 s.  Now I have a question: how do I estimate the time step “dt” between these data points?  Knowing the default sampling rate of the card seems a better way to help me to define the "dt", and calculates the time stamp at each data point.  If the foregoing concept is true, how the internal sampling rate in NI6624 obtains?  Beside, for the transient counter signal, any way to get the time stamp of data points is also welcome.
    Thanks for anyone comment,
    Adan

    Adan,
    When selecting "Implicit" as the DAQmx Timing type, you are indicating
    that a data point will be taken for every measurement the counter
    performs. When you create a task of type "Low Frequency with 1
    Counter," the counter simply uses the card's internal timebase to
    measure the period between edges of your signal. It then takes this
    period measurement and converts it to a frequency. Therefore, the
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    Ryan Verret
    Product Marketing Engineer
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    National Instruments

  • 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.

  • How to change audio capture devices sample rate

    I am going to develop a online voice chat applet. Now everything works fine. But I didn't find anyway to capture audio in a lower sample rate. The default sample rate is 441000Hz. I am using MediaLocator to capture audio and pass it into Processor.
    MediaLocator ml = new MediaLocator("dsound://");
    DataSource ds;
    DataSource clone;
    try {          
    ds = javax.media.Manager.createDataSource(locator);
    } catch (Exception e) {
    return "Couldn't create DataSource"+e.getMessage();

    Use GSM format and do it with setting of BitRateControl. It is better than convert the sample rate of your capturing device. I suppose that you try to reduce load of network traffic with changing sample rate, right?
    Best regards from Germany,
    r.v.

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