Hello All!

Let me lay out my situation.  Looking to talk through the available options with some experts!

I am currently re-editing all nine seasons of a television series that began shooting/editing/airing in the early 2000's.  We were fairly good at archiving the projects and media, but as you can imagine we have a few holes of missing media.  Since most of our material still links, we are continuing to edit in the original NTSC 24p project (not a 23.976 project as it did not exist back then) on newer, updated Avid versions (v.2019 and v.2021).  All of the master clips still exist, so we are batch re-capturing offline picture. I'm looking for help in recreating what we did with the sound files so that we don't have any drift issues.  

Our work flow:

Picture was down and cross converted to DVCAM (29.97i) from original HD-CAM masters (23.976p) and digitized into the 24p project with pulldown.

Audio WAV files from the sound mixer was duplicated and converted with a program (that no one can recall what it was called) and then imported into AVID.  Because those files were not used in the final mix, we did not keep those converted files).  BUT we do know what the program was doing.  On a first pass it would change the sample rate of the audio file to 47.952 kHz (pulling it down), and then on a second pass it would then change the metadata of the audio file back to 48kHz so the avid wouldn't apply any sample rate conversion on import.

I'm wondering if anyone knows of a program that can recreate the above work flow.  Or if there is an AVID workflow that I can do that would achieve the same result. Obviously, if we were starting from scratch we would just import into a 23.976 project and move along.  But we need to import to existing master clips so our sequences/groups etc are about to be brought back online.

Also, let me know if there it might also be smart to ask this question in an audio portion (i just gravitate towards this thread is we've always worked on Mac AVIDS and you guys can usually answer all the issues we have).

Thanks!

-Rob

Pretty sure you can do this with Wave Agent by Sound Devices. I don't really understand the need for a second pass. You either slow it down or you don't. When you import you can tell MC to NOT convert source with "pulldown rates" (like 47952 and 48048).

Was the picture shot at 24.000fps? Was sound recorded at 48000? Digital? Or Analog?

Back in the day, Avid did not give you the option to convert or not convert sample rate on import.  It would read the meta data and would change it if it didn't match the project sample rate.  So that is why we had to change the meta data back to to 48000.  The picture was shot on 23.976 (I belive) and sound was digitally recorded at 48000.

But if the shoot was 23.976/48000, the audio files would have needed to be restamped as 48048 to pull them up to match the 24fps playback of the picture -- as far as I can wrap my head around this.

The audio was shot at 29.97/48000.  The changing of the sample rate settings were created before I arrived on the show.  So we just used the previous settings and everything worked.  I do recall it was resampling to 47952.  I have downloaded the program you mentioned, but need to put it on an older machine to see if it will work.  I tried Isotope's RX8 resampling, but after changing the sample rate the file still said it was 48k.  Maybe that's fine?  I just need to play around with it I guess.