Audio drift

Question

Level 1

23 points

Audio drift

I have a Tascam DR-40 audio recorder, which records 96k/24bit stereo WAV files. Using in conjunction with a Canon HF M40 camera (tapeless / prores 422 / 29.97).

I bump the audio down to 48k/16bit via Logic (or Soundtrack Pro or Bias Peak) before ingesting, and no matter what, it loses sync by the time it gets out to about ten minutes or so.

As noted, all the video is at 29.97. Not doing 24p here.

I exported the original video to standard NTSC-DV 16:9.

I took the 48k/16bit AIF file into FCP7, attached a matte to the video channel for the duration and saved it out as a DV clip with the exact same specs, verified as such in the QT Player.

Brought both of those new files back into FCP, and set 'em up in a standard DV sequence that matches their specs. It still doesn't sync.

Tried in FCP 7 and FCP X. Snow Leopard, 32GB RAM, Mac Pro.

Any help appreciated!

Posted on Jul 24, 2012 6:28 PM

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Answer 1

Jul 25, 2012 2:05 PM in response to FBC_21CA

Just for kicks, have you just tried using the 96k file for your main audio?

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Answer 2

Jim Cookman

Level 7

23,435 points

Jul 26, 2012 1:47 PM in response to FBC_21CA

What puzzles me is why you think this SHOULD work. If the audio is not driven by a SMPTE timecode clock sync is problematic.

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Answer 3

Jul 26, 2012 2:00 PM in response to Jim Cookman

Unless you're using really cheap equipment (which I can't imagine would record 96k) your audio should line up with maybe a frame or 2 over an hour (in my experience).

People have reported problems when they import audio and edit into a sequence when the sequence frame rate does not match the fcp easy setup.

How much of a drift are you seeing?

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