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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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2020-09-29, 10:17 AM
(This post was last modified: 2020-09-29, 10:26 AM by pipefan413.)
Sorry to necropost, but maybe worth saying for anybody else worried about this (as I was): it would *appear* that AmarecTV might be able to handle this by locking into the clock of a video capture card. As in, if you record the audio, even from a completely different capture device from the one capturing your video, it seems to be able to align it to the video without losing sync.
That said, I also can't seem to capture 24-bit audio with AmarecTV; if I manually overtype "24" on top of the "16" in the bit depth argument, it actually captures at 44.1 kHz / 16-bit even though I asked it to capture 96 kHz / 24-bit. I wonder if this is a speed thing and it's trying to avoid overloading itself, but that shouldn't be necessary with the speed of my hardware...
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Never tried AmarecTV, but is it possible that it's just resampling the audio rather than directly controlling the capture card?
I've wondered if there's a way to alter the clock in Windows somehow, but I wouldn't know where to start.
Finally got my M-audio 2496 installed in my old XP HTPC, so I can finally do audio captures again after I get all the software set up. Wonder if it will have the same clock problems.
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(2020-10-01, 01:12 PM)BusterD Wrote: Never tried AmarecTV, but is it possible that it's just resampling the audio rather than directly controlling the capture card?
I've wondered if there's a way to alter the clock in Windows somehow, but I wouldn't know where to start.
Finally got my M-audio 2496 installed in my old XP HTPC, so I can finally do audio captures again after I get all the software set up. Wonder if it will have the same clock problems.
Hmm, no idea. I'm a bit worried about running into similar issues when I attempt to capture through a dedicated audio interface, but I haven't attempted it yet because I'm waiting on some different cables arriving first... aaaaaaand they literally arrived *as I was typing that*, hahaha!
I think I'll try recording audio directly through the capture card first and then try a separate run capturing it through a USB audio interface, and see if the starts align or not. However, I do not really trust either of the USB audio interfaces I currently have, so I'm probably going to have to go shopping again... this is a bloody expensive hobby, isn't it?
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