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  • Hey all, just changed over the backend after 15 years I figured time to give it a bit of an update, its probably gonna be a bit weird for most of you and i am sure there is a few bugs to work out but it should kinda work the same as before... hopefully :)

Audio Slipping at 24 FPS

Gabriel C.

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Hi Guys,

Sorry if this has been covered. I looked around and didn't see anything.

I shot a feature a few months ago, and we're in the final stages of post ... we shot and edited at 24FPS (not 23.98) and recorded sound at 48K. All the syncing worked for the initial edit in FCP, but once we exported an OMF for ProTools, mixed, and brought it back into FCP for our final compile, it had horrendus slipping problems (noticeable after a minute or so).

It's strange because when the audio guy sends me files at 48K AND 48.048K, they have different lengths in ProTools, but the exact same files have the exact same lengths in FCP - almost as if FCP is stretching and contracting them.

Does anyone know about this - and if so - is there a work around?

Thanks so much!!!
 
Re-sample

Re-sample

It's strange because when the audio guy sends me files at 48K AND 48.048K, they have different lengths in ProTools, but the exact same files have the exact same lengths in FCP - almost as if FCP is stretching and contracting them.

You need to look at the frame count for the images, the frame rate for the images, then resample the audio to match.

In my editing system I recomend using 48000samples/second true 24fps since that works with 2000 samples per image frame, and that way the audio comes out exact to the sample, "never" any sync problems.

If you are editing 23.976 you figure the sample rate by using the NTSC ratio of (1000/1001)*n, so you can need 2002 samples per image frame since the image frames run slower. (or you can need to resample on the inverse depending on if you are short or long).

You will need to find the running time of the images then resample the audio to match that running time.

Do not try to dub audio into a computer through the sound board, the crystals in the computer and audio recorder are not in sync so you can get running time errors that way, you then need to resample the WAV file to get the right running time.

If you can sync using SMPTE that may adjust the running times for you since some Audio programs can resample on the fly to match the SMPTE time code.
 
Thank you ... and I don't get it. But more over ... thank you. Is no one else having this issue?
 
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