Audio Sample Rate for Video Clips?

How do you check the audio sample rate for a clip of video?
Thanks.

scroll over in the browser until you get to the column for audio. That's one way.

Similar Messages

  • Audio Sample Rate

    Can I change the audio sample rate of a clip (imported from camcorder) "after" it's been imported into Final Cut X?

    Thank you for the quick response.  My video and audio are out of sync (imported from mini-DV camcorder).  The audio sample rate is 32 kHZ, and I believe I should have used a "custom" sample rate setting of 48.  I really don't want to start over with this project and re-import all of my footage, but that may be the best thing to do.  I also learned from a previous post that I should capture no more than 10 mins. of footage at one time. 
    P.S.  I'm about done with this camera!

  • Problems with audio sample rate when capturing DV50

    Hello from Spain, I have a problem when capturing DV50 source footage: I setup my sequence settings and capturing settings like always, DV50 codec and 48k sample rate for audio, then when the capturing process finish the result is a file that contains audio to a non-48k audio sample rate, then video and audio don´t match. The sample rate is 47945 and other similars, not always the same, sometimes 47214 sometimes 48000, and there is no rules...My capturing device works ok in other computers and the footage has been capture ok in other Mac...I don´t know what happens.
    Sebas, kind regards
    G5 Quad   Mac OS X (10.4.6)  

    Hello!, first of all thanks for your reply. Yes, the camera is the same, the problem is on my computer, I have discovered a solution: first I capture the audio, only the audio, and then the video, only the video. After this I soncronize the two separate files on Final Cut timeline. It´s a slow process, but I am really desesperated. I don´t know what happens, I few minutes ago I have captured a 30 mins. video and the sample rate of the audio was 42.145k (???). It´s a crazy thing...

  • 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?

  • 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?

  • What is the best way to deal with different audio sample rates on the same timeline ?

    what is the best way to deal with different audio sample rates on the same timeline ?

    You don't have to do anything special. If possible, start your project with a clip that has the desired target frame rate and audio sample rate, and your project parameters will be set automatically. Other sample rates will be converted under the covers.
    For example, if your video is shot at 48khz, you can add music files at 44.1khz with no problem.
    If you are recording audio that you want to synch with video (multicam), you will get best results if everything is 48khz, but you can use 44.1 if that is all you have. Once I forgot to reset my Zoom to 48,000 and it still worked.

  • 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

  • Geting audio sample rate error, help

    Hey all, been doing a massive project where I ma bring in tons of old 8 mm tapes, hi8 and digital 8. This one tape I brought on though however is giving me grief. I keep getting 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 files to be out of sync. Make sure the audio sample rate of your capture preset matches the sample rate of your tape.
    If find this odd though cause the tape should be standard, NTSC dv 48 K. Any suggestions. Also how do I reset my final cut so that when I plug in a camera it always reads it as what it recognizes. I ask this because I had to used the setting uncontrollable device because on the original old 8mm tapes there is no time code so I had to capture that way.
    Anyhow, any suggestions on this would be great for if I recapture. Cause I could line it up by eye but want to find the problem so I know for the future. Thank you.
    Nathan

    This is a recent problem that seems related to a recent upgrade of QuickTime. Here's why.
    In the last month, a rash of these posts have begun to appear:
    "DV Capture Audio problem"
    http://discussions.apple.com/thread.jspa?messageID=6708693&#6708693
    "audio/video"
    http://discussions.apple.com/thread.jspa?messageID=6591262&#6591262
    Plus this thread, plus my own.
    In my case, nothing changed in my operating system or Final Cut Pro version. I upgraded to QuickTime 7.1.6, and the problem began. I have upgraded all the way to 7.4 to no avail. When I attempt to import a DV clip using the same Sony DVCAM deck that imported the same clip in December, I get the error every time. Nothing has changed in the tape, the deck, the Project or Final Cut. I am simply unable to import DV video. I can import other kinds (Panasonic P2, for example), but DV is a no go. I cannot get rid of this error.

  • Correcting audio sample rate

    I keep getting 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 your capture preset matches the sample rate of your tape."
    I used a Sony 3CCD cam set at 48K, I set FCP to Easy set-up, I have FCP Studio, easy set up is on all formats, all rates, NTSC.
    I replay the clips and the audio seems fine, but I don't trust it with this error.
    Anyone have any ideas????

    Hi(Bonjour)!
    Make sure that your sequence settings is 48 kHz 16 bit for audio.
    Michel Boissonneault

  • Determining DV tape audio sample rate

    I have DV tapes of "archived" final edits recorded on to mini DV tape using two "early" (late 90's) Sony DV cameras with some segments of the same tape recorded with 48k audio and other sections with 32k audio (on the same mini DV tape). I tried to capture the footage into FCP 5.1.2 using a Sony DSR-11 deck and a G4 1.42 GHz dual processor w/ 2 gigs RAM, Radeon 9800 Pro video card, Mac OS 10.4.8
    Everything seemed fine until I got a message after capturing a 3 minute clip using a stock FCP 5 NTSC preset with 48k audio; the message stated the "audio source rate and capture sample rates did not match & footage may be out of synch" - the captured clip played back perfect. The DSR-11 did not display the DV tape's audio sample rate; I put the DV tape into the camera it was originally recorded on and "16 bit" displayed on the camera's LCD screen.
    I tried capturing the same footage using FCP 4.5 installed on another drive in the same computer, and got no warning message re: audio sample rate. Just to "be sure", I created a custom capture preset that included 32 bit audio and captured the same clip using FCP 4.5 with a preset for 32k audio - no warning message, and everything was perfectly in synch. Everything was in synch for ALL the captured clips & the audio sample rate in FCP for captured footage was either 48k or 32K for each respective preset.
    The dilemma: how to be certain what the true/original audio sample rate is.
    G4 1.42 GHz   Mac OS X (10.4.8)  

    "audio source rate and capture sample rates did not match & footage may be out of synch"<<</div>
    William,
    The message itself seems to be a bug that started with FCP 5 - most users that receive this message have reported that there is NO actual problem.
    -DH

  • Audio Sample Rate Query

    Hi FCE users,
    I am about to start capturing 4 hrs of wedding video. I have a fair bit of experience on FCP but am new to FCE and its variations in functionality.
    When I ran a quick capture test, it warned me on completion that the audio sample rates were different. The video was shot in Hungary and I have no way of contacting the videographer, however I assume that it was recorded in 32khz as when I changed the capture preset to this, no warning message occurred. The DV cam I'm using most helpfully doesn't display a bit rate but given that no warning occurs after capture, I'm happy to go with the 32khz preset.
    Most of the articles posted on the net however state that the 'SEQUENCE PRESETS' as well as the 'Capture Presets' need to be changed. Almost all of it relates to FCP, as opposed to FC Express.
    The equivalent on FCE - 'Sequence Settings' doesn't seem to offer a 32khz option, and the Audio Rate shown in the Browser seems to be fixed to 48khz.
    My question is:
    Does the sequence setting variation matter? - Do I assume that given the scaled down functionality of FCE that it takes care of this automatically OR that 48khz is a fixed audio rate for sequences in FCE and all non 48khz audio clips needs to be rendered prior to output.
    No doubt I will be bringing music from CD's in to the project and sequence also, so your help would be greatly appreciated.
    I don't want to bring in 4 hours worth of footage and then have to recapture!
    Thanks for your help!
    Rich

    Thanks for your swift reply Tom.
    I've just re-reviewed the menus. I can display the 12/16 info on record, but it doesn't offer it as a playback display option. However... I've just recorded more footage into FCE and received no error message with the capture preset on 32 so can I be fairly confident that its 32?
    I've just figured out how to change the sequence format to 32 (it seems to be created by default with capture preset set to 32, wheras I was trying to alter an existing sequence).
    Rich

  • 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!

  • How do you set sequence audio sample rate?

    I tried posting this to another, but it was already answered, and noone will see it.
    I am getting the capture error "audio sample rate doesn't match" and yes, I can see in my browser that the clip is 48khz/16bit, but the sequence is 48khz/32 bit. Howver, wherever I look to change the sequence setting, it is 48/16 already. I've gone to FCP on the menu dropdown to audio/video settings - it's correct all through there. I've gone to the menu dropdown Sequence settings, and it's correct there. I've closed down, opened a new sequence, restarted, everything I can think of. Is there a secret to getting them to match? And, can I fix a project already edited with this discrepancy? Its export to QT is WAY out of synch.

    Annoying - I can't see your post when I am in reply mode.
    Yes, I get this error when i am capturing. From reading a previous post about the error, I thought the solution was to check the audio rate of the captured clip in the browser, and then make sure it matched the audio rate of the sequence. Like I said, everywhere you get to change the sequence setting, it SAYS it is 48/16, but yet, whe I scroll over in the browser, it says the audio rate is 48 KHz and the audio format is 32-bit floating point. Am I looking at the right places?
    I can't check the settings in the camera until this evening...don't ask.
    I'm not so sure this is not a QT issue instead of anything to do with capturing, etc. It plays back fine in the timeline.

  • Capturing-audio sample rate

    I have fce hd v 3.5. My camera is a sony dcr-trv11. I am trying to "capture now" but it looks like I am dropping frames. When i hit "esc" it stops capture of course, but I get a message that reads "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 ans 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." Okay, so how do I do that? When I originally captured my footage with imovie I had no problems. I had trouble importing them to FCE so I scrapped them all and decided to start from scratch with FCE. I have 199.5 GB of free space.

    Most miniDV cameras can be set to record audio as 12-bit or 16-bit; and most of them come set to 12-bit as the default.
    Your preset (easy setup) in FCE has to match both the video & audio setup of your camera. Normally, you would select the DV-NTSC easy setup in FCE, which would give you a sequence that expects DV-NTSC video and 16-bit (aka 48KHz) audio. If your video was recorded as 12-bit (aka 32KHz) audio, but you captured into a 16-bit (48KHz) sequence in FCE, that would give you the mismatch.
    Check your camera - if it was set to 12 bit audio for the tape you are trying to capture, then in FCE you should select the DV-NTSC 32KHz easy setup for your sequence before you capture your tape.
    There are many different reasons you might be getting dropped frames - can you tell us more about your exact setup, esp. if you have an external HD connected to your system. Oh, and by the way, 512MB is the bare minimum to run FCE, you will find things much better overall if you upgrade to at least 1GB.

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