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Sync problems during analog capture?
Anyone ever had problems with sync drift when separately capturing analog audio and video?

Lately I've been backing up some VHS tapes using my Sony Blu-ray recorder. It records in h264 instead of MPEG-2, which is great for someone like me who has little experience with video encoding. But it only records audio in 256kbps AC3, so I've been using my M-audio 2496 sound card to record the audio to PCM at the same time. Only thing is, when I try to sync up the video from the recorder with the audio from the PC, even after I sync the start times, the PC recorded audio drifts forward out of sync at about a rate of 40 milliseconds every 10 minutes, which becomes pretty noticable after a while. Now that I think about it, Jonno had similar problems with a capture of an analog-only LD that I made for him a couple years ago using the same equipment.

Not sure why the sync drift occurs, or how I could prevent it... I also investigated an LD I recently recorded with digital audio, and there's also a slight sync drift, but it's only about 30ms for every hour of video.

Guess I'll just have to cut bits of silence here and there whenever I want to make a backup with lossless audio. I probably won't notice the difference, so I guess I'll only bother with titles I really care about.
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It's because analogue audio doesn't come with a fixed sample rate, instead it is sampled by the internal clock of the recorder. Your Blu Ray recorder has its own clock for that, as does your M-Audio sound card (or your PC, not 100% sure). Those two internal clocks are unlikely to be perfectly in sync with each other, hence the drift.

For what it's worth, the drift should at least be measurable (like you said, 40 milliseconds every 10 minutes) and constant. So you can just figure out the exact amount of drift, and then reinterpret the sample rate to have the correct timing and then resample the result to a standardized sample rate (like 48kHz).

You could prevent it if you had professional equipment that can sync clocks, like broadcast video cameras typically have it. With that, one device would be generating the timecode/clock and the other one would be receiving it. Thus both would be in perfect sync. More or less anyway, I guess.
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Thanks for the response, I figured it might have something to do with the internal clocks. Not sure if I'd want to resample the audio though, I'd be too paranoid about affecting the quality, even if it was imperceptible.
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Well you either have to live with a weird sample rate then or change the framerate of the video to something weird instead.
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I would at least try resampling to see if it affects the quality at all.
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What you gotta take into consideration here is that even if you don't resample, it will very likely be resampled upon playback, as most audio hardware just defaults to standard sample rates. So if your sound card operates at 48kHz and you give it something else, it will do resampling on its own, and likely not very well. So arguably you could get better quality by doing the resampling yourself with good software (for example Izotope 64-bit SRC)
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Sorry I forgot to respond to this, but I ended up keeping the original 24/96 audio and just making cuts every 10 minutes or so in order to keep everything in sync. So there's some slight drift at each cut, but it's only a max of like 20ms so it's not noticable, and no need to resample or use out of BD spec sampling rates.
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