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Concert – Audio Sync Headache
We recorded an 11 camera concert and took sound feeds from audio desk into cameras. Cameras are all sync’d up on timeline (camera flashes at end of concert all match perfectly) – I am now putting the final mix on the timeline and things are drifting out noticably over about 4mins. Sound provided is 48k 24bit WAV file. Should perhaps ask for 16bit, I know, but I am not sure that this is causing the drift.
Cameras were DSR-570 DVCAM (PAL) recording 48k,
Should I ask for the sound in a different format – aiff or something?
I have always managed this no problem before, using desk sound as ref, then matching up later. But this is causing a headache.
Any ideas? Thanks
Martin
G5 Dual 2Ghz
FCP 5.1.latest -
6 Replies
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Thanks Dave – I tried that first of all, but without success.
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Sample rate conversions should not at all effect timing.
Sounds like you should of synced up the cams and had the audio recorder/workstation ‘slaving’. Or for ease of edit, live switch it with the recorder/workstation slaving to your master record deck and recording a couple of iso cams to cover your mistakes. Keeps you on your toes and lots of fun!Only thing for the time variance is resync after 4 mins and overlay shots over the cut. you could try ‘fit to fill’ for each song – but may be hard depending what in/out points (drum hits etc) to fit too.
Good luck!
Jon -
Bit depth (24 vs 16) doesn’t affect sync. It sounds like you were relying on the sync accuracy of the DVCams internal crystal and that should be fine for long runs providing the audio recording system was also accurately clocked.
What was the audio recorded to and is the drift constantly in one direction. If so then it is a matter varispeeding the original mix files by the required amount.
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You’re right that bit depth _shouldn’t_ affect sync. However I noticed some weirdness on a recent project when dropping 24bit 48k WAV audio files into a 90 minute timeline. The audio clip fell slightly short – by a few seconds – of the last video clip. So I rechecked everything – The audio files showed up as 90 minutes in Quicktime and Audacity, everything started at zero on the timeline… then I tried converting the 24bit WAV to a 16bit 48K AIFF. At first I relinked the audio clip to the new file, and this didn’t help, ut when I imported the new 16bit AIFF and overwrote the audio clip in the timeline, everything synced up.
And yes, I realise that doesn’t really make sense, but I was happy to have found a solution, and I still have doubts about audio sync with 24bit WAVs in FCP, but I admit I haven’t had the time to experiment further…
Matthew
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Thanks for all the advice. Used Audacity in the end to slow it down by 0.041% and export to 48K 16bit AIFF. Has done the trick perfectly without altering pitch or anything.
Thank again.
Martin.
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I’m going through the same thing but opposite. I shot a five camera concert and the band recorded it with 24 tracks. They mixed the show down and sent me the audio file as a 44.1k wav file. I noticed everything not lining up exactly but i cut their audio a few frames a song to line it up as best as they could tell. They would now like me to mix in a bit of the audio from my shotgun mics because they never mic’d the audience and want more of that live feel back in the dvd. Damn.
So I’ve tried the fit to fill option, but either it’s just not working or they songs are too long for it too work by the end of the song and it still has a noticeable drift.
Can I take their audio into something like Audacity or use Final Cut’s speed tool too make them match up? Is there an equation to go from 44.1 to 48k? Am I asking the impossible? Should I just admit to them I tweaked their stuff and as for a 48k mix out of the board?
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