[Author85]

What everybody has said above is correct, but I'll add some more information. I've done many DVD reauthor's with alternate audio sources. My big thing is taking DVD with crappy AC-3 sourced audio and re-synching them to either better sounding PCM sources or soundboards.

First off, I don't know if you realize what a HUGE job it is to resynch a show. If you don't have timecodes to use (which is probably a 99.9% chance you don't) you are going to have to do it the old fashioned way of visually synching up the peaks on the waveforms on some kind of NLE. I've found from experience that DAT recordings are much easier to synch up than analog or cassette recordings. The reason is that DAT seems to record at a much more constant speed. Even if it is fast or slow, as long as it's not variable, you can apply a linear correction across large portions of the soundtrack. If it's analog sourced, watch out! I've had some analog sourced re-synch's that have taken 200 to 300 corrections across an entire concert.

The reason, IMO, is that in older style cassette recordings, the "record" speed seems to be pretty variable, directly dependant upon the battery supplying a constant and steady voltage source. More modern recorders, especially DAT's, seem to be less prone to a variable speed recording, and are thus easier to make timing corrections to in post.

In an NLE, I basically use the original audio soundtrack as a "guide source", since I know that it is synched correctly to the video. I then place my "new source" directly below the original "guide source" and make timing corrections to it to exactly match the graphical waveform peaks and valleys. It's all visual at this point.

Oh, also, be sure to turn your "quantize to frames" feature OFF, if you are using a video NLE for this. If you don't, then the editor will only allow you to make cuts or corrections at each video "frame", which is about 1/30 of a second. With "quantize to frames" turned off, you can make cuts and/or timing corrections at any point on the timeline, not just at every 1/30 of a second. This is EXTREMELY important, as most timing corrections cannot be done at just 1/30 of a second intervals. If you want your audio perfectly synched you must do this. If you are using an audio editor for this project, then chances are your program will default to not making cuts every 1/30 of a second, so it won't be an issue at all.

Once you graphically have the waveforms for both the "guide source" and "new source" lined up visually, then you can play both audios on the timeline at the same time to fine tune your synch job. It will be very obvious when the synch starts drifting as drumbeats and other sounds will sound off-beat. When this happens, you know you have to go back to the "new source" timeline to make some kind of timing correction, to get the source back in synch with the "guide source". This is what you have to do, sometimes every 10 to 15 seconds, sometime every 10 to 15 minutes, depending on the playback speed of the "new source". As you can see, this can get quite repetitive and time-consuming.

Now, you can make a very half-assed synch job that will sound like crap, and be noticeably out of synch, by simply "slapping" the new source on the timeline and trimming the ends without doing the above mentioned work. But because the VAST majority of audio recordings weren't recorded at a constant speed and hence don't playback at a constant speed, you won't get anything you would want to trade. If you want an absolutely perfect synch job, you must use my method above.

So, that's how I do it, and I always make an absolutely perfect synch, down to the microsecond, for my audio re-authors. If anybody has an easier way, please let me know, as I haven't found a way to do it any better without the benefit of audio/video timecodes.

Good luck!