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

Similar Messages

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.

Audio sample rates

I am getting the message after capturing "The audio sample rate of one or more of your captured media files does not match the sample rate on your source tape......etc". The only thing I notice is in the Aud Format window the captured audio is "16-bit integer" and the sequence is "32-bit Floating Point". Should I ignore this or is there a way to get them to match?
G5 dual 2.7 Mac OS X (10.4.5)

Hi mmona,
I did a little bit of googling and reading about 32 bit floating and 16 bit integer, and I left a bit more puzzled and without enough knowledge to be dangerous.
However, the warning FCP is giving you is for the sample rate, as opposed to the audio format.
To see what I mean, highlight the clip and then hit Command-9. Audio rate and audio format are the 3rd and 2nd to last items. The warning is refering to the audio rate, but the difference you are reporting is in the audio format (which I remain curious about.)
I'm going to go out on a limb and guess that you are capturing DV that originated in a camcorder that cost less than $10K. In many cases, these camcorders capture unlocked audio. Google unlocked DV audio and you'll get a lot of hits, and many explainations.
I've gotten the warning you describe, but only with DV originating in certain less expensive cameras. Not all cameras are equal when it comes to this. The XL1, despite being a higher end camera in this class, often shows this issue.
Sometimes in these cases, when I look at the audio rate on the clip (3rd from bottom after hitting Command-9) it will report a number like 48.009. The practical effect is that the audio drifts slowly out of sync as the clip progresses in FCP's 48 KHz world.
So I keep my clips short and the problem remains invisible.
Tom

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.

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

Change capture settings to 32k audio sample rate???

I have a DVCAM tape that my client recorded and the audio sample rate is 32k. I have noticed that the audio seems to get out of sync when I capture it using the 48k NTSC. I also get a warning saying that my source tape's audio sample rate doesn't match the capture settings and that audio could be out of sync. I looked at past posts and it does seem to be the 32k verses 48k.
My question is how do I change my capture settings to 32k? I see how you do it on the sequence settings, but I can't figure it out on the capture settings.
Thanks,
John Schenkel

John,
I recently read a post where there were four likely answers to the problem, yet I was not sure which one was the best solution.
Why? Because even though the question was marked as answered the person posting the original question didn't mark which individual's answer was the solution.
Please help future users to find answers more quickly, who knows, you may find yourself in the same situation someday.

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.

Mismatching Audio Sample Rates Upon Capture

I recently bought a cheap MiniDv JVC video camera, and I am having trouble capturing video and audio together in FCP 6.0. The capture seems fine, but when I finish an error pops up which reads: "The Audio Sample rate in your presets does not match the one on your source tape." As a result there is a delay between the video and audio tracks that creates huge amounts of extra work.
My video camera captures sound in 12-bit format, and from what I understand that usually corresponds to a 32MHz sample rate. I can't find any 32 MHz presets in the Easy Set Ups or other menus, the system seems geared entirely towards 48 MHz sound. I've messed around with some different combinations of settings, but so far to no avail. Any help someone can give me will be much appreciated.

You might want to duplicate the capture preset for DV/NTSC 48k and change the audio to 32k.
Then try that.
But I'd also change the JVC's audio settings to record in 48 while you're waiting. Should be able to do it... cheap or not.
CaptM

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...

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.

Error: "Locating Timecode Break"

I'm trying to capture some raw footage for a project. About 2 minutes into the capture, I get an error message saying "Locating Timecode Break". The capture stops - or almost stops - the camera keeps playing very slowly, one frame at a time. So when I finally hit the 'esc' key, I get another message saying that the 'audio sample rate of captured files does not match the sample rate of source files" (and it continues). I've restarted the computer, but the same thing keeps happening, at the same point on the (DV) tape.
I'm using the same camera, computer and software that I always do. Any ideas about what could be causing this?

Please give your full system specifications, what computer, what drives, what camera or deck, how everything's connected, what settings are on the media, what settings you're using in Final Cut, and finally what version of the software, of the OS, and of QuickTime are you using
There could be two separate things here: one, there might be timecode breaks on the tape, which the capture process is trying to navigate around; and two, there is a mismatch between the audio sample rate you shot and the settings you used to capture the material.

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!

Why is FME audio sample rate restrictive?

I am unable to select higher than 22kHz audio sample rate on
the FME Control Panel - this results in very poor audio quality
delivered! The input device delivers up to 48kHz and is succesfully
addressed by WME - the only reason why I'm still using WME.
In case you are interested, I have listed the input device
spec below:
DVBETA (LairdTelemedia)
Output Ports available (I use FireWire)
SDI Digital Video: SMPTE 259, PAL or NTSC, on BNC a
connector
AES/EBU Digital Stereo Audio: 16 Bit @ 48 kHz, Embedded on
SDI connector
* IEEE-1394 (FireWire) x2: maximum 400 Mbps, PAL or NTSC, on
a 4-Pin and 6-Pin connectorsMachine Control, RS-422, on a 9-Pin
connector
Time Code: DVITC or VITC, on a BNC connector

I have the same issue when using my video capture card in FME
on Vista.
I am having a lot of issues actually.
1. CPU while encoding video & audio is only reaching
around 15% but I am dropping frames when I try and use high quality
encoding settings.
2. Audio and Video get out of sync very easily.
3. Audio sounds VERY odd, especially during music. It seems
like the speed of the audio is constantly shifting. Makes things
sound like oooOOOoooOOoo etc..
I hope Adobe is addressing these issues.
Thanks and take care.

How do I determine that my audio sample rate matches my tape?

How do I determine that my audio sample rate of my capture preset matches the sample rate of my tape?

You've really got two issues.
DV records audio and video differently and if the camera is substandard, you will have drift over long captures. Solutions: new, better camera or capture in small (max 10 minutes) chunks.
The other issue is the 32k sample rate, which will exacerbate issue 1. You can duplicate the 48k preset and modify the copy for 32k and work in that until you are ready to deliver. Then copy it into a 48k sequence, render and output.
It's funny, sometimes I get that pop up but everything is fine.

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?

Capturing issues-audio sample rate & locating timecode break

Similar Messages

Maybe you are looking for