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Old April 27th, 2007, 07:38 AM   #1
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CD audio versus camcorder audio

I am trying to sync audio provided by the client to video I filmed. The client recorded 8 channels to a DAT machine, then transferred that to computer and created his master for CD authoring. I requested that he give me the audio file "in one piece" (before cutting out idle times, long clapping) so I can easily sync it to the video. Well, it's not happening... and I'm wondering why... My audio is 48 KHz/16 bit, his audio is 44.1 KHz CD quality - is that the reason? I ended up cutting up the CD audio in short pieces and sync the pieces one by one at the middle of the piece, this way the deviation is still within acceptable limits, but if I leave it in one piece, it' either longer or shorter (I forgot).

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Old April 27th, 2007, 08:47 AM   #2
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I don't think there's any way to automatically sync video and audio without genlocked equipment. I only use cam captured sound and have to re-sync half the time anyway.

I find a frame with clapper-like action at the beginning, sync it (usually 1 or 2 frames off), then go to the end, find another frame and stretch or compress unlinked audio as needed (usually 3 to 5 frames if the clip was an hour or so long).

Scenalyzer has an option to force sound sync that works. Doesn't seem like the new HDV capture tools do. Hoping the release of PP3 helps.
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Old April 27th, 2007, 09:42 AM   #3
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You didn't mention what your editing software is etc so it's hard to get specific on how to do it but the bottom line is you need to get the 44.1 kHz audio resampled to 48kHz without a pitch change. Some NLEs do it automatically on import but apparently that's not happening on yours. The video project is 48kHz by definition and during 1 second of video, 29.97 frames, it will read 48,000 samples of audio. That represents about 1.1 seconds of audio at a sample rate of 44.1 kHz, the audio is playing back faster than the video is being displayed with what was recorded in 1.1 seconds being played back in 1.0 seconds and what starts out together quickly falls out of sync. Resampling the audio to 48kHz before adding it to the timeline will - err, MAY that should be - fix it. Hard to know for sure because there are so many variables
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Old April 27th, 2007, 09:51 AM   #4
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What video editing software are you using? Vegas automatically compensates for clips on the timeline that are different sample rates, and I've never had any problem. Last year I used a CD recording of clean audio along with footage from the camera located in the balcony recording guide audio only since there was no time to run and tape down cables. I did a hand-clap sync mark before the presentation started. This combo held sync for the full hour but I imagine there was some luck involved in achieving that level of matching for that long. I didn't convert the CD audio separately but let Vegas handle it automatically. It worked fine both on the timeline and in the final render.
Do you know what sample rate or sync signal was used in the original ADAT 8-channel recording? That could have been non-standard and when converted to the CD could result in a slight difference from the real time events. If that was the case, then simply converting the sample rate won't make it match up. You would need to stretch slightly without changing the pitch.
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Old April 27th, 2007, 09:59 AM   #5
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What Steve said.

You needed to convert your sample rate to match your editor's project, or he needed to change his project's audio rate to match your audio rate. Either works.

If the sample rates were the same, then the drift due to no genlock shouldn't really be more than a frame or two every 10 seconds or so. I've personally shot entire films on film and wild DAT (no timecode, no genlock, just a regular el cheapo DATman) and didn't have problems with sync later, because I'd just pull up the audio when it drifted, same as you did. In fact, on most of the shorter takes, I didn't really even notice the lack of lock.

If everything works (looks good and sounds good) the way you did it, then so much the better.. Hand matching and conforming audio is still done every day somewhere on this planet, usually because people are in a panic because their genlock equipment malfunctioned or broke down, or someone forgot to slate...

As you can see, you pretty much faced up to the worst of it, and it looks like you fixed it too. It's not a huge huge problem, but it is a pain in the proverbials to fix.

So next time match up the audio rates before you sync your dailies.

But actually, you have a pretty marketable skill now, plus the experience to deal with what freaks a lot of people out these days, so congrats are in order, methinks.
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Old April 27th, 2007, 10:02 AM   #6
 
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matching sample rate is entirely dependent on your NLE/DAW system.
Some are smart enough to match them automatically, thus not only saving steps, but potentially avoiding complex problems.
For example, Sony Vegas and Sound Forge 9 force this correctly, even with BWF files.
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Old April 27th, 2007, 10:30 AM   #7
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Adobe all the way

Sorry, I forgot to mention - I used PPRO 2.0 for the video editing. When I tried importing the CD audio into PPRO 2.0 and noticed the difference, I exported the audio and took it into Audition 1.5, added the CD audio, upsampled it to 48 KHz and tried matching them in a multitrack project with the intention of deleting the camera audio and mix it down that way using the CD audio only, after syncing it all.

Syncing the beginning is not the issue, I can do that easily either in PPRO or in Audition, the problem is the drift.
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Old April 27th, 2007, 11:08 AM   #8
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Quote:
Originally Posted by Ervin Farkas View Post
...
Syncing the beginning is not the issue, I can do that easily either in PPRO or in Audition, the problem is the drift.
DATs often are not right on the money with their sample rates, one of the reasons file-based recorders are largely supplanting them. It's hard to know what gyrations that recording went through before you got ahold of it. How severe is the drift you're seeing?
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Old April 27th, 2007, 11:26 AM   #9
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The drift is about one second every five minutes.

I may have access to Vegas 6 - will this older version do the automatic audio sync or only the new version 7? I once used Soundforge, I forgot what version and dumped it in favor of Audition.
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Old April 27th, 2007, 11:50 AM   #10
 
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Any version of Vegas can do this, yes.
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Old April 27th, 2007, 12:14 PM   #11
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Vegas, here I come... you might have just won a fan over from the the Adobe side...
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Old May 1st, 2007, 03:31 PM   #12
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Did I understand this right?

OK, what seems to be too good to be true, it usually is...

From the replies I understood that I can drop my video+audio in Vegas, add a second audio and the software will automatically match the two audio tracks so they are now in sync, even if there is a difference in length.

Or... did you guys said that Vegas will only compensate for the sample rate? 'Cause if that's the case, than my Adobe software does that automatically as well, I have no reason to go to Vegas.

I might have been too hungry and didn't properly digest what I ingested... got happy too soon?

Thanks,
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Old May 1st, 2007, 05:14 PM   #13
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Quote:
Originally Posted by Ervin Farkas View Post
OK, what seems to be too good to be true, it usually is...

From the replies I understood that I can drop my video+audio in Vegas, add a second audio and the software will automatically match the two audio tracks so they are now in sync, even if there is a difference in length.

Or... did you guys said that Vegas will only compensate for the sample rate? 'Cause if that's the case, than my Adobe software does that automatically as well, I have no reason to go to Vegas.

I might have been too hungry and didn't properly digest what I ingested... got happy too soon?

Thanks,
It only will resample the file to the project rate, it won't adjust the length in the process. If the file is 30 seconds long when played at 44.1, after resampling to 48 it will still be 30 seconds long. The change in length takes place when the file is at one sample rate and is played back at a different sample rate WITHOUT resampling. Let's do the math - a 30 second file sampled at 44.1 kHz contains 1323000 samples. If you play that at 48kHz, it will play in 1323000/48000 or 27.5625 seconds, about 2.5 seconds too fast (and will also have a slightly upshifted pitch). But if the file is RESAMPLED to 48000, it will now contain 1440000 samples and will play in 1440000/48000 seconds or 30 seconds, the same length as the orginal 44.1kHz file.

If you play your picture by itself @ 48kHz and it's, say, 30 seconds in length and you play the sound by itself at 44.1kHz and it too is exactly 30 seconds in length, resampling the sound to 48kHz will enable you to sync them in the same project. But if you play the video on its own and it's 30 seconds and you play the sound on its own at 44.1 and it's 35 seconds, resampling won't help you, something else is going on. (Well, resampling will still help but it will only make it easier to solve the problem - in and of itself it's not likely to fix it.)
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Old May 1st, 2007, 07:38 PM   #14
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I still think something was not exactly at a normal sample rate either in the original 8-track recording, its editing or its conversion to the CD they gave you. If that was the case, then there will be a difference between the CD audio and your video+audio regardless of sample rate conversion (either automatically in your editing software or as a separate step).
But it should still be fixable with a slight stretch that doesn't change the pitch.
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Old May 1st, 2007, 08:15 PM   #15
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What Jay said. A 1-sec-in-5min drift isn't a 44.1 vs 48KHz issue - it's one or the other source being very slightly out of spec.

You need to apply a 0.9966% shrink to the longer one (or a 1.0033 stretch to the shorter one) and then sync them up by eyeball. Various NLES, Soundforge etc can all do this.

By the way, those numbers are for exactly 1 sec in 5 min. Work out your exact delay on the full sequence then adjust accordingly.
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