Adjusting Sample Rates

I just got Audition CC 2014 and when I try to record it says the sample rates of the input and output devices don't match. How do I adjust those rates in Windows 8.1?

Okay.  You need to go to the Windows Sound control panel and select the Playback tab.  Highlight your output device (if you have no external card there may well only be the one) and select Properties.  Then click the Advanced tab and there'll be a drop down menu for bit depth and sample rate.  I'd suggest 16 bit/44100Hz (CD Quality) for pure audio that will end up on CD or MP3 or 16 bit/48000Hz if it's going to end up on video.
Next, go back to the Windows Sound control panel, click the Recording tab, and follow the same procedure (highlight, Properties, Advanced) and select the same settings in the drop down menu there,
This should sort you out.  In the longer term, consider an external USB audio interface--it'll be hugely better quality and, if it has ASIO drivers, it'll stop Windows applications from changing audio settings without telling you.

Similar Messages

  • 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.
    The real code is the line Call ChnSplineXYCalc(..... The properties swapping can be commented out, but then the new channel would have a "system name", something like "Approximated XY", and the same for description and units... Probably one can avoid it by changing of settings, but I use to do it by code.
    In short, here you copy the properties from the "old" channel and paste them to the new one.

  • Adjusting Sampling Rate to write continuous data to excel?

    Dear colleagues,
     I am reading data from a DAQ from 4 channels and am wanting to write the data continuously to an excel sheet.  Problem is, I cannot get the sampling rate and # of samples to appear "constant".  instead, labview is only sampling x number of samples in a row and storing those to excel.  What I want is this:
    Beginning at time t=0, and every 0.2 seconds after that (5 Hz), read the values from all 4 channels and append them to an excel file.  What I am using is a 1D to 2D array converter and the Write-to-Spreadsheet function outside my while loop.  If I set it to only do 1 sample every 0.2 seconds, then it just stays at t=0, just refreshing the value and thus overwriting my excel data.  How can I make it sample continuously from t=0 to t=when STOP button is pressed, and then log all samples into excel?  I have attached my sample vi below.  Thank you.
    Attachments:
    working_final.vi ‏547 KB

    Hi Mike, Thanks for replying.
    The latter of the two methods you mentioned, " you write your values every iteration to the file, therefore you have to place the "Write vi" inside the while loop.", this would be ideal.  What sort of conversions need to be done in the while loop before the data is written? In my file, you can see that I am trying to use a converter to go from Dynamic Data to a 1D array.  Would this be the appropriate method for doing this?  What I am trying to accomplish can't possibly be this complex, I'm sure lots of people require the use of a similar write-to-spreadsheet function.

  • 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.

  • Drop the sample rate

    I'm trying to drop the hardware sample rate. It's at 44800Hz. When I want to capture audio through my Audigy 2 soundcard at 44100Hz I get the message "We do not support recording when your file does noyt match your hardware sample rate. Your current hardware sample rate is: 48000Hz."
    Edit -> Adjust Sample Rate does nothing.
    Edit -> Hardware Audio Setup shows a 48000Hz sample rate but I can't change it.
    Oh. Is there a "global" reset on settings?
    Thanks
    Gis

    >I'm trying to drop the hardware sample rate. It's at 44800Hz. When I want to capture audio through my
    Audigy 2 soundcard at 44100Hz...
    My very strong advice to you is not even to try. Audigy soundcards work internally at 48k, regardless of what you tell them - the internal architecture is fixed. If you tell them to work at 44.1k, then all they do is 'spoof' the output by doing an internal sample rate conversion on the fly. The problem with this is that it's not a very good conversion - and Audition can do it rather better. So the way to go is to record at 48k, and then do the conversion rather more transparently in Audition. Doesn't take long, and won't degrade the sound even more than the card already has. So...
    ...a better way to go would be to ditch that pile of Creative rubbish and get a proper soundcard!

  • ADjusting the sampling rate on a FIR filter?

    How do I adjust the sampling rate on a digital FIR filter? Thanks in advance.
    -David

    You should really start a new thread instead of posting to one that is 5 years old.
    To answer your question, it depends on your data. I don't use the DFD but with the filter functions in LabVIEW, if you pass a waveform data type to the function, then the waveform data type contains a dt value. So, set the DAQmx Read to return waveform data. If you are using low level filter functions where the input is a 1D DBL array, then the filter has to be configured. With the low level functions in LabVIEW, you use the various coefficients functions that have a sampling frequency input.

  • 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.

  • 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

  • CS6 will not accept my condensed mic stating; 'the sampole rates of the audio input and output do not match.  Audio cannot be recorded until this is corrected.  Use the appropriate operating system or audio device control panel to adjust the sample rates

    !

    The first things we need to know in order to be able to help are what are your operating system and audio device?
    Also the FAQ Setting the Sample Rate in Windows Vista and 7 may help.

  • 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

  • IIR Filtering and response .vi: Butterwort​h 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

  • Can I mix down to 32 bit at a higher sample rate than 44.1 kHz ?

    When I use Ableton Live, it lets me choose 16, 24, or 32 bit, and then I can choose a sample rate all the way up to 192000.  Is this possible in Audition ?  I have been going through all the preferences and all the tabs and I can't find this option.  All I find is a convert option, or the adjust option.  But that's not what I want.  I want to mix down this way.
    The closest thing I found is when I go to "Export Audio Mix Down", I can select 32 bit.  Then there is a box for sample rate, with all the different values.  But it won't allow me to change it from 44100.

    JimMcMahon85 wrote:
    Can someone explain this process in laymens terms:
    http://www.izotope.com/products/audio/ozone/OzoneDitheringGuide.pdf ---> specifically Section: VIII "Don't believe the hype"
    I don't read graphs well, can someone put in laymens terms how to do this test, step by step, and where do i get a pure sinewave to import into audition in the first place??
    UNbelievable: So I have to first run visual tests using a sinewave to make sure dither is working properly, then do listening tests with different types of dither to hear which I like best on my source material, and then for different source material it's best to use different types of dither techniques???... Am I getting this right???...
    Hmm... you only need to run tests and do all that crap if you are completely paranoid. Visual tests prove nothing in terms of what you want to put on a CD - unless it's test tones, of course. For the vast majority of use, any form of dither at all is so much better than no dither that it simply doesn't matter. At the extreme risk of upsetting the vast majority of users, I'd say that dither is more critical if you are reproducing wide dynamic range acoustic material than anything produced synthetically in a studio - simply because the extremely compressed nature of most commercial music means that even the reverb tails drop off into noise before you get to the dither level. And that's one of the main points really - if the noise floor of your recording is at, say, -80dB then you simply won't be hearing the effects of dither, whatever form it takes - because that noise is doing the dithering for you. So you'd only ever hear the effect of LSB dither (what MBIT+, etc. does) when you do a fade to the 16-bit absolute zero at the end of your track.
    Second point: you cannot dither a 32-bit Floating Point recording, under any circumstances at all. You can only dither a recording if it's stored in an integer-based format - like the 16-bit files that go on a CD. Technically then, you can dither a 24-bit recording - although there wouldn't be any point, because the dither would be at a level which was impossible to reproduce on real-world electronics - which would promptly swamp the entire effect you were listening for with its own noise anyway. Bottom line - the only signals you need to dither are the 16-bit ones on the file you use for creating CD copies. And you should only dither once - hence the seemingly strange instructions in the Ozone guide about turning off the Audition dither when you save the converted copy. The basic idea is that you apply the dither to the 32-bit file during the truncation process - and that's dithered the file (albeit 'virtually') just the once. Now if you do the final file conversion in Audition, you need to make sure the dithering is turned off during the process, otherwise all the good work that Ozone did is undone. What you need to do is to transfer the Ozone dithering at the 16-bit level directly to a 16-bit file proper, without anything else interfering with it at all. So what you do with Ozone is to do the dithering to your master file, and save that as something else - don't leave the master file like that at all. After saving it, undo the changes to the original, in fact - otherwise it's effectively not a file you could use to generate a master with a greater-than-16-bit depth from any more, because it will all have been truncated. Small point, but easy to overlook!
    just what's the easiest way to test if a simple dithering setting is
    working for 32-bit down to 16-bit in Audition?...  Why is there no info
    about dithering from 32 bit to 16 bit (which is better then dithering
    from 24-bit isn't it)?
    I hope that the answers to at least some of this are clearer now, but just to reiterate: The easiest way to test if its working is to burn a CD with your material on it, and at the end of a track, turn the volume right up. If it fades away smoothly to absolute zero on a system with lower noise than the CD produces then the dither has worked. If you hear a strange sort-of 'crunchy' noise at the final point, then it hasn't. There is info about the 32 to 16-bit dithering process in the Ozone manual, but you probably didn't understand it, and the reason that there's nothing worth talking about in the Audition manual is because it's pretty useless. Earlier versions of it were better, but Adobe didn't seem to like that too much, so it's been systematically denuded of useful information over the releases. Don't ask me why; I don't know what the official answer to the manual situation is at all, except that manuals are expensive to print, and have also to be compatible with the file format for the help files - which are essentially identical to it.
    Part of the answer will undoubtedly be that Audition is a 'professional' product, and that 'professionals' should know all this stuff already, therefore the manual only really has to be a list of available functions, and not how to use them. I don't like that approach very much - there's no baseline definition of what a 'professional' should know (or even how they should behave...), and it's an unrealistic view of the people that use Audition anyway. Many of them would regard themselves as professional journalists, or whatever, but they still have to use the software, despite knowing very little about it technically. For these people, and probably a lot of others, the manual sucks big time.
    It's all about educating people in the end - and as you are in the process of discovering, all education causes brain damage - otherwise it hasn't worked.

  • How to use on board counter to change sample rate dynamically on pci-6134

    Hi,
    I am relatively new in LabView.
    I am making power quality measurement system and I need to vary the sampling rate of my pci-6134 dynamically (all channels simultaneously). What I need is to have a constant amount of samples in each period of measured signal (grid voltage), which changes slightly all the time. Therefore I will have to measure the voltage, find its exact frequency and then adjust the sampling rate of daq accordingly. I know that there will always be some delay, but I would rather like not to go into any predictive algorithms...
    I have found an information in the Forum that one of possible solutions is to use an onboard counter to change the sampling frequency but I have no idea how to make that. Can someone help me or possibly show an example? 
    Is there a simple way to solve that problem?
    Thanks in advance
    Andrzej

    At least at a glance, the code generally looks like it ought to work.  Two thoughts:
    1. Instead of getting into PFI3 vs PFI8 routing stuff, can't you just specify "Dev1/Ctr0InternalOutput" as the AI Sample Clock source?  (You may need to right-click the terminal to get at the menu that exposes the so-called "advanced terminals").
    2. Try writing both the freq AND duty cycle  properties when you want to update the freq.  Or try using the DAQmx Write vi instead of a property node.  My past experience suggests that writing only the freq property *should* still work, but writing both isn't hard to try and may turn out to help if the behavior of your version of DAQmx differs somehow.
    -Kevin P.
    P.S. Bonus 3rd thought.  I just went back to reread the thread more carefully, including your first screenshot.  I'm now thinking that maybe the hardware actually WAS behaving properly, and that you just weren't aware of it.   When you query the AI task for it's sampling rate, all the task can know is whatever rate you told it when you configured it outside the loop.  So even as you change the counter freq to change the actual hardware sampling rate on the fly, the AI task will continue to report its orig freq.  After all, how is *it* supposed to know?
           Try an experiment:   Set your original freq very low so that the AI task produces a timeout error without getting all the requested samples within the 10 sec timeout window.  Run and verify the timeout.  Then run again, but after 3-5 seconds set a new frequency that will produce all those samples in another 1 sec or less.  Verify that you get the samples rather than timing out.  That should demonstrate taht the counter freq change really *does* produce a change to the hardware sample rate, even though the task property node remains unaware.

  • How can I ensure that the audio sample rate of my capture preset matches?

    Hi
    1. I have found that none of my ten projects has sound, although audio meters & settings moves up & down.
    Simultaniously, canvas displays that In: Not Set- & Out Not Set. Below those reads: Video V1: 00:03:42;13 Audio A1: 00:03:42;13 A2: 00:42;13. These happen on the images of all my ten projects when the playhead stops in timeline
    2. In Viewer the In & Out sets show a thin red line connecting them.
    I never had this problem before. A/V of all projects were doing just fine till last night.
    I wonder if these issues are interrelated, or if I may have clicked something wrongly that has triggered this, which I don't remember now.
    I would appreciate it if you could kindly address this problem and help me to resolve the isssue. Thank you. Faruk.

    The red line indicates the audio must be rendered. This may be caused by several things but your playback settings may be set too high for your system to handle. You may have too many audio tracks for your computer to play in realtime.
    Select one of the audio files in the timeline, get properties, not the audio format and sampling rate.
    Now got to Sequence>Settings and determine if the settings match your audio settings.
    Your online help system can guide you through adjusting your system settings, user settings and sequence settings.
    bogiesan

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