Sample Rate 48.001 not 48.000 kHz

Hi,
Sometimes when I capture clips the resulting sample rate of the audio is 48.001. Then, in FCE HD, when playing the clip from the sequence there is a beeping over the normal audio, indicating I need to render the audio.
Does anyone have any advice to get my audio to import at 48.000 and not 48.001 kHz sample rate?
I'm using a Sony TRV310 D8 camera. Computer is G5 Dual 2.3, QT7.0.2, FCE HD 3.0, Mac OS X 10.4.2. Capturing to stock 250GB hard drive.
Thanks,
Chad

Yes, easy setup is set to 48 kHz and my camera is set to 16 bit.
I have the feeling it has something to do with the start/stop points of the footage screwing things up somehow. The footage in question is of a soccer game, about 45 minutes in length and with about 20 start/stops.
FCE HD was set to "stop capture with dropped frames" and the capture (using "clip mode") was done with no perceivable problems.
For now I ended up doing the following workaround on the sound track portion (sorry, I can't remember who originally posted this):
Open your iMovie .dv file in Quicktime Pro.
Extract the audio track
Export the audio track as a Quicktime Movie, Linear PCM format, Channels Stereo (L R), 48kHz, 16bit sample size
Delete the audio track from the .dv file
Open the exported audio file, select all and copy.
Select All in the .dv file
Add the exported Audio file using Add to Movie
Export/Save As the .dv file as a self contained Quicktime Movie
Use this movie in FCE
Trouble is, the sound is slightly out of sync now (I suppose the sound file shrunk by a wee bit going from 48.001 to 48.000 kHz). For this footage the sync is not critical.
I'm still curious as to what exactly is going on with my captures.
Thanks,
Mace

Similar Messages

  • Sample Rate And Bitrate Not Allowed ... ?

    I have almost 70GB of my music in my iTunes, ALL in AAC 320kbps 48Khz. Now that I just updated to iTunes 9, I just tried to add more of my new CD's to my library and they are being imported at 44Khz...?? I went to change it back to 320/48 and I get ...
    "The selected combination of bitrate and sample rate is not allowed"
    Bug? Anyone else have this problem? HELP!
    btw ..I tired to use my laptop and my work PC to see if it's just this machine that is causing the problem....and those too (now updated to iTunes 9) are doing the same thing.

    I don't know why a previous version might have allowed a 48khz sample rate, but audio CDs are 44.1kHz (that sample rate is locked into the Red Book CD standard) so there would be no good reason I can think of to attempt to import at a 48KHz sample rate. It won't increase quality and could cause artifacting.
    Regards.

  • Sample Rate 44.1 kHz creates Hum/Buzz in speakers

    I noticed a quite hum in my speakers, Mackie MR8s, and cold not figure out what it was. So going through the menus in Logic Express 8 I found the "audio" under settings. Where it says sample rate I changed it to 48 kHz and the hum disappeared. I went to switch it back and I get a buzz, but only when the 44 kHz sample rate is chosen. Does any know why it would create a hum? Thanks.

    It's more of a a real quite hiss than a hum. My battery backup was making them buzz so I've moved that across the room. Could it because of their close proximity to my imac or other electrical items on the other side of my wall in the other room. The hum doesn't come through on headphones or in recordings. Thanks.

  • Sample rate for digital sampling (cDAQ-9172 & NI 9401)

    Hi!
    I have a cDAQ-9172 with a NI 9401 C-series module (digital). I would like to sample the digital inputs with a sample rate of e.g. 400 kHz or 200 kHz. My problem is that I can only select a the 100kHzTimebase clock, and therefore only get a 100 kHz sample rate. The 20MHzTimebase clock is too fast, since it gives me a sample rate of 20 MHz). Is it possible to get a user defined sample rate of e.g. 200 kHz, by e.g. dividing down the 20MHzTimebase clock?
    Solved!
    Go to Solution.

    The cDAQ-9172 chassis does not have an internal timing engine for digital input however you can use one of the onboard counters to generate your clock.  Set your pulse train generation counter to be one of the internal counters, such as "cDAQ1/_ctr0" and your digital input sample clock source to be /cDAQ1/Ctr0InternalOutput". 

  • FCE Capture-sample rate error

    After capturing Digital8 footage off my Sony DCR-TRV310, I get this error.....
    "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 capture preset matches the sample rate of your tape."
    So far, in spot checking the media it seems to be OK, but I'm anal about errors! I couldn't find the audio sample rate for the camera/tape in the sony manual, any ideas? Also were do I select the audio sample rate of capture preset, even if I knew the tapes sample rate? Or is this nothing to worry about
    Thanks, Schooner

    Hi(Bonjour)!
    Usually, camcorders allows to set the audio sample rate to 12 bit (32 kHz) or 16bit (48 kHz).
    The audio sample rate for DCR-TRV 310 can be set to 12 or 16 bit. See page 57 in this user manual:
    http://www.mediacollege.com/equipment/manual/sony/camera/dcrtrv315.pdf
    FCE easy setup allows to select DV-NTSC 32 or 48 kHz for sequence. This settings apply to capture session too. The message you've got means that the audio sample rate on tape doesn't match the sequence audio format.
    This problem may give a drift synch between audio and video.
    First check the item format for video clip in browser. The audio sample rate should be 32 or 48 kHz. Note it and select the right setup in easy setup dialog. Create a new sequence and insert your video clip.
    Michel Boissonneault

  • Core Audio - Sample rate

    I've just got myself a copy of Logic Pro 8 as a complete newbie and have hit a bit of a hurdle within 24 hours of opening the software.
    I was using the software without a problem just tinkering around using a Carillion MIDI interface to play some software instruments. I don't have a mic yet so thought I'd try my bluetooth headset just so I could play with recorded vocals. When I started speaking while Logoc was recording it came up with an error message
    "Core Audio
    Sample Rate 8000 not allowed."
    I then turned off my bluetooth headset and turned bluetooth off on my Mac so I could go back to using the MIDI keyboard but got no sound. I closed Logic and re-opened, selected "Empty Project" withing the "Explore" tab and was again presented with a series of the messages, all reading as per the above error messages but I get 3 with "8000" and one with "73536".
    When I'm playing the keyboard I can see the note I'm playing or the chord in the dialog box at the bottom of the Logic interface screen but again, no sound.
    Any help would be hugely appreciated. Again, I am a COMPLETE Logic novice so please be gentle with me.
    Regards,
    Mark

    Hi Jounik,
    I went into Audio MIDI Setup and the headset didn’t appear. The keyboard appears as “Midilink” but I couldn’t see where to check the sample rate. I did however use the Test Setup facility and the keyboard was only producing a sound when the key was released.
    I also had a look at: Logic Pro > Preferences > Audio (Devices) but wasn’t really sure what to look for. This is what I was confronted with http://adoseof.co.uk/resources/Picture1.png
    The project sample rate was set to 44.100 KHz
    I like the idea of using the internal mic. I’d tried that before but was getting feedback. Plugging the headphones in seems like a good solution.
    To try and get the keyboard working again I opened a new Empty Project and left the dialog with the default settings as indicated here: http://adoseof.co.uk/resources/Picture2.png
    I’m not sure what you mean by “In the I/O slot of the channel strip open a synth, e.g. ES1”
    I then used the on-screen keyboard and still no sound was coming out.
    Any other ideas?
    I tried the keyboard in garage band and it worked fine so I can only assume it must be something to do with Logic.
    Mark
    Message was edited by: hotsawz

  • 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.

  • How to coerce the sampling rate??

    I think I found my problem with sampling rate.
    I'm using a PCI-5122 scope card, and in many of my aquisitions, I'm setting the sample rate to 40MS/s. Apparently, this is not a valid number and the scope reverts to 50 MS/s
    Later when I try to calculate cycles per second based on cycles per sample, I need the actual sample rate, and 40 MS/s ain't it.
    I'm trying to coerce sampling rate.
    please,.

    The digitizer coerces the sample rate because of how the sample clock is derrived from the Reference Clock.  The following information is on page 13 of the specifications:
    http://digital.ni.com/manuals.nsf/websearch/C6B059C1BDD70101862574C8005567F1
    The sample clock is created by dividing down the Reference clock (internal reference clock is 100MS/s) by decimation, and it divides it by N, which is an integer between 2 and 65530.  
    Thus 50MS/s uses a decimation factor of 2, and 33.3MS/s is the next valid sample rate with a decimation factor of 3.  So when you specify a sample rate that is not possible, the driver automatically coerces the requested sample rate up to the next valid rate.  You can obtain the actual sample rate used in an acquisition using the NI-SCOPE property "Actual Sample Rate", or the LabVIEW VI "niScope Sample Rate.vi".  Using this property, you can get the values you need for your calculations.
    I hope this helps!
    Nathan
    Product Support Engineer
    National Instruments

  • Can I set an arbritrary sampling rate in Biobench (need 60/sec to sync to video)?

    Is there a way to set a sampling rate that is not in the drop-down menu in Biobench? Thanks!

    Hi George,
    BioBench comes with a set of predefined sampling rates. I encourage you to try using LabVIEW to trigger/sync your DAQ with video. The sample freq. in BB are std. for low freq. biomedical signals.
    Please let me know if you have any further questions.
    Regards,
    Morten Jensen
    National Instruments

  • Sample rate of cio-das08j​r

    hello;
    after an internet search. and reading the help, I still post on this forum, hoping that I find an answer;
    At first, I realized the aquisiion using cio-das08 Jr. with the help of someone on this forum (cj), I thought I solved the problem by that when I want to change the sample rate I can not find my device inTime base, and the acquisition is made with a sample rate that Ido not know, but when I check hardware Clocked; of hardwaresetup, I find my device in time bases, but when execution, I found a message that says invalid option
    my question is how can I change the sample rate of my devicecio-das08jr
    Attachments:
    hardware Setup.GIF ‏42 KB

    hello
    I have not found the solution to control the sample rate of cio-das08jr, please help me it urgent
    Attachments:
    sample rate.GIF ‏40 KB

  • Which DAQ card can give accurate specified sampling rate

    Dear friends:
    I have an application to sample the data from an analogue device with user specified sampling rate, e.g.171KHz. May I know which NI DAQ card can specify any sampling rate below 200KHz within tolerance  of inaccuracy of 1KHz? Appreciate for your help
    Andy   

    Do you need both accuracy *and* precision?  
    Precision will largely be determined by quantization effects as the sample clock must be an integer divisor of the board's master timebase.  For example, the 6259 M-series board that I commonly use has an 80 MHz master timebase which can be used to generate the sample clock.  In fact, I'm pretty sure it *is* used by default.   The nearest integer divisor when requesting 171 kHz will be 468, producing an actual nominal sample rate of 170.9402... kHz.  (A divisor of 467 would produce a nominal sample rate of 171.3062...)     The quantization doesn't really scale linearly over wide ranges of target sample rate, but for rates in the vicinity of 171 kHz, quantization steps are a bit less than 0.4 kHz, so you can expect to hit your target to within about +/- 0.2 kHz.
    Next you need to consider clock accuracy.  The 6259 board is rated at 50 parts per million accuracy at nominal temperature of 25 C.  I don't know the temperature effect or whether you need to concern yourself with it in your app.  But 50 parts per million on a target of 171 kHz amounts to less than 0.01 kHz accuracy error.
    So it would appear that the 6259 would be a candidate.  There are likely other cheaper M-series boards that could also work.  The older E-series had only a 20 MHz timebase, which would lead to quantization error of about 1.5 kHz.  Don't recall their accuracy specs, but quantization already puts you over your error budget.
    -Kevin P.

  • Why can't I record in 96.000 khz sample rate ?

    I am running Logic Pro 9.1.7 (1700.57) (32-bit). My computer is Mac OS 10.6.8. I am using a Duet Apogee
    I have always recorded in 44.100 khz, but now someone has asked me to record in 96.000 khz. Everything is fine until I change the sample rate in Logic. Then I get a tinny, distorted sound, and this is for any audio I play through my computer, not just the track that's in Logoc. Is there a way out of this ?
    Thanks

    You own the hardware and don't know how to use it?
    Didn't you install some software for the Duet? Somewhere on your system will be a software applet that let's you change settings on the Duet.
    I just took a quick look online.. the "Maestro software Mixer Applet". Try looking in Applications or Utilities.
    Need to read up about what you're using. Do not change Duet settings with Logic running.
    I don't own a Duet but all of this stuff works basically the same.

  • The selected combination of bit rate and sample rate is not allowed.

    I've always burned my purchased CDs into iTunes using a AAC 320 kbps and 48.000kHz. Now when I select that option in iTunes 9 I'm getting a "The selected combination of bit rate and sample rate is not allowed." pop up screen. So is it impossible to burn my albums in a 320kbps/48.000 AAC format?. I had no problems in previous versions of iTunes using this setting. I guess I might have to switch over using a MP3 import setting instead.

    Audio CDs are encoded with a sample rate of 44.1 kHz. Ripping them at 48 kHz requires resampling and does not improve the quality. Unless you have a special need for 48, use 44.1.

  • 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?

  • Could you confirm that ALAC doesn't play sample rate of 96 kHz?

    Hi,
    Could you confirm for me that ALAC doesn't play sample rate of 96 kHz? Because my codecs converter, regardless of what sampling I set up gives me 44.1 KHz in ALAC.
    Thanks

    Newme wrote:
    Ed 2345 wrote:
    "Unfortunately, if you are working with hi-res audio files, Apple Lossless is not lossless."
    Really? Can you back up that. It should be easy to prove. if it really is so. What kind of files don't fulfil that?
    Newme,
    Simple enough to demonstrate.  If you take a WAV 96/24 and use iTunes to convert it to Apple Lossless and then back to WAV (using "Automatic" setting in both cases), you end up with a significantly smaller file.  Something gets lost.
    Asking this because DSD files can't actually be tuned into other formats without losing data. Whether there is actually anyone that can hear the difference is another matter of course :-)
    Wasn't referring to DSD, but it is an interesting question.  The usual lossless formats, ALAC and FLAC, are designed to improve on the inefficiencies of PCM encoding as is used in WAV, AIFF, and audio CDs.  DSD is a horse of a different color, as it is not PCM based at all.

Maybe you are looking for

  • Error occurred in using interactive form

    We are using ABAP WebDynpro to build application based on adobe form. We can run it without problem for the first time. However it produced an error when it runs for the second time. To clarify the problem see description below. The WebDynpro Applica

  • Urgent Inspection type

    Hi friends, Can anybody tell me How I can post material from QI in material to material transfer posting 309 mvt.Their is any inspection type that I can assign to material or I can create new inspetion type for this requirement.using customer enhance

  • Mouse Integration Incompatible w/ SMS 2003 Advanced Client

    Ref Setup: http://wikis.sun.com/display/VDI3/Getting+Started+-VDIDemo Ref Bugs: http://www.virtualbox.org/ticket/414 http://www.virtualbox.org/ticket/1324 Ref Docs: http://support.microsoft.com/kb/933986 http://technet.microsoft.com/en-us/sms/bb67678

  • Attaching URL to text in smartforms

    hi,   i need to  attach an url to a text .For example if my text is go  if a  click on go it should take to a url that i have attached.please teell me how to aatach url to text in smartforms. regards, sreelakshmi.

  • Connect to DB2 from oracle 11g on windows 2008 server

    Hi Folks, I want to conenct to DB2 from oracle 11g using DB link. DB2 : user : db2user pwd : db2pwd database : db2database (OSBLDEV) able to connect to db2 server (installed on machine M1) using db2 client with above details from machine M2 (where or