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Changing and/or disabling frame rate on audio files
Hi everyone, I’m new here and I’m italian so please forgive me for any misuse of the english language.
I’m starting to edit a 16mm short film which recently came back from telecine.
I’m currently in the process of syncing the audio files (WAVE) with the clips and I encountered a problem.
The film was shot at 25 fps but the audio files were recorded with a 24 fps timecode.
I know that fps on audio is just a stream of data that doesn’t modify the actual recording speed of the files but I’m having sync problems with a 25 fps PAL timeline.
The audio is synced at the beginning then starts to lose the sync gradually.Is there any way to disable/remove the framerate/timecode from the audio files, ideally without re-encoding?
I apologize in advance if this question has been answered before, I did use the search function and I didn’t find anything.PS: this happened to me last year on another project and I “solved” the problem exporting the audio with Compressor (thus eliminating any timecode). But this time I have 3 to 4 audio tracks in each file because the sound was recorded with various equipment (radio microphones, mono boom and stereo boom) and when I attempt to re-encode with Compressor (obviously) everything is mixed in a single track.
PPS: I thought about conforming the footage at 24 fps with Cinema Tools but I suppose this could be a problem when I send the EDL to the telecine for the onlining, color grading and final re-capture at 2k resolution.
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9 Replies
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[Marco Circosta Garcia] “I know that fps on audio is just a stream of data that doesn’t modify the actual recording speed”
FALSE
[Marco Circosta Garcia] “The audio is synced at the beginning then starts to lose the sync gradually.”
Answer to above.
Steve Eisen
Eisen Video Productions
Vice President
Chicago Final Cut Pro Users Group -
Ok, sorry, I thought the audio recording itself wasn’t frame based, otherwise I don’t understand how could I solve the problem by re-encoding it without changing the sample rate, like I did in that previous case I mentioned. That time I literally didn’t do anything to the audio other then re-encoding it in the same format (48Khz, 24 bit) and when I imported it on Final Cut it didn’t have any frame rate/timecode anymore and I could sync it without any problem.
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Just to sort a few things. Film shot @ 25? Telecine @ 25? Wav files recorded how?
You say you have wav files with 24fps timecode. Was this because the recordist had a device that recorded broadcast wav with embedded time? If so and the rate was set to 24fps then FCP does take notice of timecode and frame rates and is effectively varispeeding the audio.
Audio is real time and as long as the frame rate of the camera, telecine and sequence settings are all 25fps then you need to reimport the audio but do so in a way that FCP doesn’t stamp or read the frame rate as 24fps. So copy all the audio to aif files and before you import them into FCP make sure your easy setup is on 25fps. The aif files should have a slight name change so when batch converting, add a prefix. Do not change the sample rate as the sync drift is due to FCP interpreting a frame rate.
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You absolutely understood my problem. Both film and telecine are at 25 fps. The recordist did exactly what you said.
[Michael Gissing] “So copy all the audio to aif files and before you import them into FCP make sure your easy setup is on 25fps. The aif files should have a slight name change so when batch converting, add a prefix. Do not change the sample rate as the sync drift is due to FCP interpreting a frame rate.”
If you mean export all the audio files with compressor to aif files and then import them in FCP I already tried it: it works but it merges the 4 tracks of the WAVE files into a single stereo track. As I said above the WAVE files have 3 to 4 tracks, each one recorded with a different microphone at the same time.
Is there a way to do this without merging the tracks? -
I usually do this with PC software like AVTransfer which can batch convert polywavs. Check if Compressor is set to mixdown rather than pass the files through untouched. Otherwise there is probably free software out in the world like Audacity that can do this.
Sound Devices also makes software called Wave Agent that can manipulate polywavs, possibly restamping timecode metadata. Again that is free PC software.
Finally an alternate method may be possible via manipulation of an XML. This tutorial from Matt Lyon talks about fixing incorrect timestamping of audio in FCP and also the second half explains how to import audio assets without incorrectly stamping an incorrect frame rate.
https://library.creativecow.net/articles/lyon_matt/
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I solved it, thanks to Michael for providing me with the right imput. I used an easier method though, discovered after messing with Matt Lyon’s tutorial a bit.
Here’s what I did: I simply selected the file in the bin, choose MODIFY–>TIMECODE… and then I unchecked the SOURCE TIMECODE.
Doing this simply disables the timecode from the audio files, apparently solving my problem. -
Great result and thanks for reporting your method.
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Frame rate doesn’t alter the recording speed. The problem is that the editing software is adjusting the PLAYBACK speed. Should mean that the problem is easier to fix by changing the metadata in the sound file.
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Timecode is metadata. Actual sampling rate of audio files is 48 000 Hz instead of what would translate to 24 Hz for film (or 23.976 Hz for HD). The problem comes from FCP. It compensates using playback speed to match conflicting timecode metadata. Change or remove timecode metadata on audio files to solve the problem. Wave Agent from Sound Devices can easily do this.
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