Inconsistent Audio Sample Rate - Export issue

I am attempting to export my project into a Windows Media file and I'm getting an error that says "inconsistent audio sample rate" Does anyone know how to fix this problem?

That error was introduced with the release of Quicktime 7.2. Flip4Mac released a beta with the fix for that particular audio error thing HERE. I don't think the final is ready yet, just the beta, but it's working for me. Don't be "put off" by it, but, before you can download the beta, they ask you to fill out some registration info so you can report back any problems you might encounter.

Similar Messages

  • Cannot export as WMV - Inconsistent Audio Sample Rate error

    Hello,
    I am getting an error message when I try to export a sequence using quicktime conversion to a WMV. "Inconsistent Audio Sample Rate - The media you are exporting contains audio with multiple sample rates".
    I used Soundtrack Pro for the first time- to analysis and fix an audio file with pops. It worked wonderfully. I don't have much experience with this program and the audio is set to 32-bit stereo with a file ext stp. Not sure how to change this or if I need to.
    I have done some tests and I have no problem exporting as a quicktime. I am using the latest version of flip4mac software. Your help is appreciated.
    Cheers,
    Chris

    Hi Susan,
    Thanks for your quick reply. The link helped alot.
    For others facing this problem, I found this solution.
    1. Export sequence as a Quicktime movie (not conversion) using current settings and make movie self-contained.
    2. Create a new sequence and change the sequence audio setting (under sequence-general) to 44.1 kHz - Dept 16-bit config default
    3. Drag your Quicktime movie into this new sequence.
    4. Export using Quicktime conversion to WMV with these settings for audio.
    5. WMA9 Standard - One pass, CBR at 64kbps, 44.1 khz, Stereo.
    Voila.
    I have struggled with flip4mac from the start. Never had any luck using the 2 pass method which 1'm told is better quality. Tech support at the flip4mac discussion group generated much discussion but no solution. I have settled with single pass quality in order to get the job done.
    Does any remember the days of linear editing when you delivered a master, protection copy and some VHS dubs and you were done? Oh, I'm showing my age. Ha.Ha.
    Cheers,
    Chris

  • Message: inconsistent audio sample rate

    when trying to Share my movie to create a .wmv file, I get the following message: "inconsistent audio sample rate -- the media you are exporting contains audio with multiple sample rates."
    The audio to this clip is not the native audio. I replaced it ( on the second track - the first track audio I muted by turning off the check mark next to that track) with audio imported from Itunes. That audio clip is an aac audio file, 128 kbps, 44.1 khz. In expert settings I matched this, and indicated coding method: One pass constant bit rate (CBR).
    Anyone know what it is I am doing wrong? Many thanks

    No. That is not how it works.
    44100Hz is the frequency and the audio sample rate could be 8, 12, 16 or even 24. Same with 48, 96 and higher frequencies. Open the Audio MIDI Setup app to see what your machine and sound card can offer.
    QuickTime files can combine up to 99 tracks so a mix of sample rates wouldn't be much of an issue. WMP exports (I've never made one) seem to require only one audio source or constant sample sizes.
    iMovie uses 48KHz because that is how it comes off DV tape. AIFF files are usually the same as audio CD (44.1KHz and 16 bit sample size) and I don't know if iMovie upsamples or otherwise changes them at import but I doubt it.
    But 44100 does not equal 12 bit sample size.
    Hope this helps clear that issue up.

  • Error message stops .wmv creation: "Inconsistent Audio Sample Rate..."

    Just recently when attempting to export a video into the .wmv format, QuickTime Pro 7.2 gives me this alert message, "Inconsistent Audio Sample Rate. The media you are exporting contains audio with multiple sample rates." QuickTime will not compress (export) to .wmv. It will export to .mov. and .m4v.
    This problem is new. For more than a year my weekly video exported to the .wmv format without fail.
    I am working on a Mac G4 Dual Processor and iMovieHD.
    Any help you can offer would be greatly appreciated.
    Mac G4 Dual Processor   Mac OS X (10.4.10)  

    Just in from flip4mac support:
    The recent release of QuickTime 7.2 created an audio compression error for Flip4Mac WMV Studio, Studio Pro and Studio Pro HD customers. Specifically, when encoding WMV files with Flip4Mac WMV v.2.1.2 using uncompressed audio sources on QuickTime 7.2 systems, customers were receiving "Inconsistent Audio Sample Rate" error and the encode will fail. This problem has been resolved with the beta release of Flip4Mac WMV v2.1.33. If you are using QuickTime 7.2, please upgrade to this release at http://www.flip4mac.com/wmv_beta.htm

  • Error Msg.: "Inconsistent Audio Sample Rate" with Flip4Mac

    If you're using Flip4Mac to produce WMV files, the latest QT updates will create a conflict with Flip4Mac, which produces the error msg. referenced in the subj. heading above.
    If you're using Flip4Mac, you may want to wait until Telestream resolves the issue before upgrading QT.
    More info is here: http://www.flip4mac.com/knowledge/kb_0070.htm
    and here: http://forum.flip4mac.com/forum/messageview.aspx?catid=10&threadid=270&enterthre ad=y
    MacPro, MBP 17in (2 GB RAM), PB G4 15in, PB G4 12in, iMac G3   Mac OS X (10.4.9)   Loyal to Apple since the Apple ][+

    Just in from flip4mac support:
    The recent release of QuickTime 7.2 created an audio compression error for Flip4Mac WMV Studio, Studio Pro and Studio Pro HD customers. Specifically, when encoding WMV files with Flip4Mac WMV v.2.1.2 using uncompressed audio sources on QuickTime 7.2 systems, customers were receiving "Inconsistent Audio Sample Rate" error and the encode will fail. This problem has been resolved with the beta release of Flip4Mac WMV v2.1.33. If you are using QuickTime 7.2, please upgrade to this release at http://www.flip4mac.com/wmv_beta.htm

  • Fixed but still says Inconsistent Audio Sample Rates

    I imported some purchased music .aif files; completed my piece then tried to export a .wmv file through Quicktime. It told me I had "Inconsistent Audio Sample Rates" so I went back and re-exported the file through Quicktime with a sample rate matching the rest of my project.
    I'm still getting the "Inconsistent Audio Sample Rates".
    What do I need to do to get the project exported?
    Cheers,
    D

    I have also been getting this error - just starting today. I tried exporting a mixdown in aiff, then converting that to MP3, bringing it back into a new sequence -- all with no luck.
    However, I then exported this mixed-down version as a self-contained movie. Then, I did the wmv conversion with Cleaner 6 and it works. Slight glitch in frame 1 for some reason (motion file within the FCP sequence) -- but it seems like it's on the right track.
    I also get the feeling this is due to the audio update. These are 8-track sequences I've been working with on and off for two months -- and the first time I try to render out after the audio, osx, and QT update, and it's crapping out on me. Could also be the changes to the h.264?
    I've been spending all day on this, and I'm at my wits end!

  • Export/share wmv "inconsistent audio sample rate"

    I've never used iMovie before today. I successfully imported a mini DV clip and need to "share" it as an wmv. I keep getting an error stating "inconsistent audion sample rate". What does this mean and how do I fix it so I can save my clip as a wmv file?
    Thanks much.

    That error was introduced with the release of Quicktime 7.2. Flip4Mac released a beta with the fix for that particular audio error thing HERE. I don't think the final is ready yet, just the beta, but it's working for me. Don't be "put off" by it, but, before you can download the beta, they ask you to fill out some registration info so you can report back any problems you might encounter.

  • Inconsistent audio sample rates....? pt. 2

    I'm running Compressor3. The last three projects I've done have all been put together on fcp6, and the audio has been sampled at 48khz. When I go to compress, I get an 'inconsistent/multiple audio sample rates' warning. However, all my audio is set to the same rates. I've tried cutting down to only one audio source, and tried also resampling through Soundtrack to 44.1, adjusting my settings in Compressor as well. Any suggestions?
    I should add that I can compress from .mov to .mp4, etc.....but the one format I need is .wmv, and that's the one that is giving me trouble.

    It is a well-known issue but it's not highly publicized. Flip4Mac has a problem with the last round of Quciktime. You willneed to download a new rev of Flip from Telestream's site. I think it's still being called a beta version, even after several months.
    bogiesan

  • Inconsistent audio sample rates....?

    I'm running Compressor3. The last three projects I've done have all been put together on fcp6, and the audio has been sampled at 48khz. When I go to compress, I get an 'inconsistent/multiple audio sample rates' warning. However, all my audio is set to the same rates. I've tried cutting down to only one audio source, and tried also resampling through Soundtrack to 44.1, adjusting my settings in Compressor as well. Any suggestions?

    Which codec are you using in Compressor? If you are using Flip4Mac, have a look at this knowledge base article: http://www.flip4mac.com/knowledge/kb_0070.htm

  • Problem with audio sample rate when exporting

    I am having an issue with my audio sample rate. For some reason it is set at 8 khz by defualt when I am exporting. This was not an issue until the latest update (2 weeks ago). I'll try and give all the info I can up front. Also I tried customer support...that was a nightmare.
    Ok the issue I am addressing is that the audio sample rate when exporting is at 8khz by default. Which to my knowledge has NO relevant use what so ever. Even when I set the file format to h.264 it insists on resetting to 8khz. Even if I manually set everything look at the render queue and click on output module again it changes the sample rate back to 8 khz automatically which I must adjust again. Why can it not just be set at 48000 like every other adobe product is and like it used to be two weeks ago. Does anybody have an answer?

    This problem is addressed by the After Effects CC (12.2.1) bug-fix update, which is now available:
    http://adobe.ly/AE_CC_1221
    Note the part at the end of that page about a crucial update for the Creative Cloud desktop application, which addresses some severe problems with AME, Premiere Pro, and After Effects.

  • The whole audio sync/weird sample rate/drift issue (FCP6)

    I can tell this question is doomed to fail on account of viewing the other unanswered questions on the subject in the forums, but I'll try nonetheless!
    So yeah, importing longer DV clips....audio rate winds up being 'weird' (as in 48003.9kHz instead of 48). Any way of straightening that out that doesn't require a buncha hoops? I've already tried re-exporting it from QT at 48kHz and a few other things. Do I just have to capture in separate chunks instead of one big go?
    I found a doc that says, "In previous versions of Final Cut Pro, the "Sync Adjust Movies Over [n] Minutes" option could be enabled in the preferences, triggering an adjustment in the audio sample rate based on the amount of audio recorded to disk during capture. In Final Cut Pro 4 [and up, I'm guessing] this preference has been replaced by automatic audio rate analysis, specific to DV media..."
    Apparently this rate analysis ain't working as well as it should...or something...
    Anyone?
    Thanks,
    E

    If your footage was shot with the Panasonic dvx100 camera, there is a known audio sync issue. You can install a patch from disc 1 of the FCP install discs inside of the "extras" folder. I've had this problem from time to time and I capture DVCPRO 50, so it's not just a mini DV issue. I usually get an audio rate slightly below 48, like 47.89999 or so. Are you using capture now? I used to get this problem a lot more when using capture now as opposed to log and capture. My solution has always been to always use log and capture and keep clips 15 minutes or shorter. I've tried trashing prefs and all that jazz, swapping out hard drives, but log and capture and shorter clip lengths always seem to clear it up.

  • DV 16:9 but Only Exports 4:3 WHY? Also Audio Sample Rate Problem

    I'm quite new to Final Cut and have FC 6.01. I use PAL 25fps and a 16:9 SD DV Camcorder with FireWire SONY DV VTR Deck. I have two problems:
    1. When i capture my 16:9 DV footage the Logging Window shows only 4:3 with a distorted image in it. Though I have chosen the easy set up and told Final Cut that im editing Anamorphic 16:9 I also found that FC will only export 4:3. However, during the editing process i see 16:9 in the Browser Preview and Canvas Window.
    2.(Not sure if its related) I get the following message everytime after I capture individual clips or if i press the Escape key during a capture:
    "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 captured preset matches the sample rate of your tape"
    Does any one else have this problem?
    Apple, as yet have not given me any answers.

    Danny Boy.. Thanks for your reply and I'll be happy if it is my fault and not FCP's. Actually i made an error in my post. It does indeed display correctly in DVDSP, it's iDVD it does not even when told to display it in 16:9. However, when i used the PAL DVD Anamorphic file preset, iDVD still couldn't display it. To get it working I had to tell Compressor specifically (in the additional settings) to encode 16:9 despite what the presets stated! No matter how one looks at this, this is confusing to say the least! If a preset says 16:9 then one should expect 16:9! Remember, Im using the display window of iDVD to show me the output.
    To detail my steps as hanumang has said, im doing the following:
    1. in FCP I encode to QuickTime Move
    2. Open iDVD & Create a Project
    3. 'Drag' The QT file into the menu
    4. Using iDVD Preview function, Preview the QT file
    5. ITS STILL 4:3
    now, the above was done with QT Conversion which as also set to encode 16:9 and still had the same result.
    Thnaks to you both.

  • Capturing issues-audio sample rate & locating timecode break

    I am a first time FCP user.
    I am working on an educational video with hours of DV source tapes.
    The source tapes are from a 2R50MC Cannon mini DV camcorder also used as the deck I am capturing from via firewire. My scratch disk is Maxtor III 500GB external drive hooked up via a firewire (I'm realizing the firewire is years old could that be a problem?)
    In my first capture test I captured 1 minute of tape by setting in and out points and after completeing the capture I received an error 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."
    For the past day I have been scouring the user manual, this discussion group and training tapes from Lynda.com and can not resolve this issue. I have determined that the source tape is 12 bit or 32 kHz and yet can not find a way to set up capture preferences to 32 kHz. Is this the problem/resolution? Page 320 of the user manual shows the QuickTime Audio Settings dialog box but I can't find it in FCP or Quicktime. Is this where I make the change to 32kHz?
    After doing the first test, I tried again and ran into a new problem—it stopped capturing with the message: Locating timecode break [press esc to abort]. It never seemed to resume capturing and I pressed escape to abort. If I'm reading FCP time code window correctly I received this time code break message within a section I had already captured previously (with the audio sync warning).
    I have done several capture tests at different points in the tape and on different tapes all with the same results. I've used the capture clip and capture now buttons. I've tested with drop frame turned on and off. Confirmed my setting of: At timecode break "Make New Clip." Confirmed my Easy Setup as DV-NTSC. My capture preset is DV NTSC 48 kHz. I've turned off and on FCP. Restarted my computer. Restarted my computer with the shift key down and ran permissions.
    Any ideas are GREATLY appreciated, T.

    I am concerned about available memory when capturing to my hard drive and the babysitting and extra steps involved considering the amount of tape I want to capture...but it does work that's GREAT!
    This is always a concern, but in your case I think having the camera and external on the same bus was causing your problem. You may have to capture a little and then transfer, rinse and repeat. Just don't try to do too much at once and let your System Drive get too full, you'll run into other problems there. Slow and steady is the pace!
    I shouldn't have stacked my questions since you answered one and Chris answered the other. Don't know how to apply the answered question and who gets the points.
    No problem, just mark it answered and divide up the solved and helpful points as you wish. All in all we really don't care too much about the points, but they do make us feel good! Thanks for your desire to use the forums properly.
    K

  • MiniDV audio sample rate issue

    I cannot get the audio sample rate to match. I have captured miniDV footage (32mhz/12bit). In Sequence settings the audio options for bit are 8, 16, 24. My audio is out of sync!!!! What can I do? I have 2 hours of raw footage and can't proceed! Any help is VERY appreciated. thanks in advance!

    Hi again!
    Hum ... that´s not as simple as it could be!
    That way i´ll have to capture twice?!
    There must be another way of doing it , i don´t think Avid can do it and FCP don´t .
    As i told you i use both systems and i know they´re limitions (or i think i know!), but it´s strange because Avid Xpress DV 3.5 (from the age of stone ) do it in a blink ... ok ... found an FCP limitation!
    Thank you!
    If there´s another way ... please feel free to post it!
    (and is was supposed to "reply" a "miniDV audio sample rate issue.
    HBars

  • Highest audio sampling rate in CS4?

    Hello,
    I apologize if this has already been asked, but I have been searching everywhere and I simply cannot find the answer to this.
    What is the highest audio sampling rate that can be utilized in Premiere Pro CS4? Can it import and export 192kHz 24-bit audio?
    Thanks in advance

    Hey Hacienda,
    I might not have the experience in audio work you have since I've only been doing this for the past 6 years or so.  But I've been a musician for far longer than that, and I've learned A LOT mostly from really smart people in the industry.  So, I'm not gonna lie to you and say that I've done extensive testing in this area because I simply do not have the equipment, nor the money to buy it (WAY too expensive).  But we do share the neophyte status when it comes to video editing :-P
    Anyways, the Nyquist Theorem is not a theory, which is what people are led to believe.  It is a theorem, meaning it's already mathematically proven.  It is proven that, as long as you follow the premise of capturing twice the highest frequency of the sound source, you'll get a perfect reproduction of it.  To capture more than that is a waste of bandwith specially because most people won't even hear above 18KHz, nor do they have the equipment to reproduce such frequencies.  Most consumer systems and audio gear, including those found in professional studios, go up to about 22KHz.  You need to spend BIG dolars for anything that goes beyond that.  So, who are we really making music for here?  The super rich?  Dolphins?
    Now, I know you're not just talking about higher frequencies, but the amount of samples needed to recronstruct a perfect copy of the original waveform.  OK, well, this is the kind of snake oil marketing BS I was talking about.  The biggest one being that 1bit DSD crap that Sony/Phillips is pushing.  Adding more samples to the recording will not make any difference on how faithfully you can reproduce a sound.  It'll just make the files bigger for no reason.  Again, the Nyquist Theorem already proves this.  This is FACT!  Here's a link I found interesting regarding these audio industry lies, maybe you will too: http://theaudiocritic.com/back_issues/The_Audio_Critic_26_r.pdf It starts on page 5, but the one pertaining this discussion is lie #3 on page 6. :-D
    Don't forget that modern converters already sample at much higher frequencies than the target sampling rate.  I believe my RME Fireface 400 samples at 5.6MHz, which is twice the amount of samples compared to DSD technology, before going back down to the target rate.  But, like I said, it does so for other reasons and NOT because it needs that many samples in order to faithfully reproduce a waveform.  Of more importance are the quality of the FIR (Finite Impulse Response) filter and the clock inside the converters.  These components are what make a converter high grade, among others.  The converter chips themselves are very inexpensive (in the tens of dolars) which why you hear some companies advertizing having the same converter chip as a ProTools HD rig (not the best example I know).
    By the way, I didn't say humans can only hear up to 20KHz.  I'm sure there are people who can hear above that.  My point was that the 20Hz - 20KHz range is what's generally accepted as an average for humans (which implies that there are people who can hear avobe/below that).  Also, the reason why modern-day pop records causes headaches and sound horrible is because of a totally different issue known as "The Loudness War" (I'm sure you know about it so I won't go into details).  However, I do agree with you as far as compressed audio goes.  Unfortunately, there's a reason for that and there's nothing we can do about it until the day Internet bandwith becomes more accessible and cheaper.  Eventually it'll get to the point where uncompressed audio can be streamed reliably through the net.  But, until then, we're stuck with MP3, AAC, DTS and other audio compression formats.  As far as digital media distribution goes, it's the future and companies are seeing that.  More and more people download music rather than buying CDs, so I do believe those numbers are accurate.  Just look at sales from iTunes and even games like Guitar Hero and Rock Band.  It's just a matter of time.
    Take care!

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