Does audio time code frame rate effect playback in FCPX

[James Lackleter]

It’s a little over 5 minutes

[Jeremy Garchow]

That should be enough to show any drift.

You are saying these same clips won’t sync in X without a resample?

Or are you asking why it syncs in Avid after the 23.976 import?

It’s possible a video clip at 24.0 frame rate and an audio clip with 23.9 tc rate can sync if the recordings are unadulterated. The tc of course, won’t match after a minute or so.

[James Lackleter]

Im trying to figure out why it’s syncing in Avid but requires resampling in FCPX. It’s the exact same settings so it’s a little baffling to me.

[Jeremy Garchow]

Can you tell us about your media exactly?

I’d like to test it.

[Jeremy Garchow]

Ironically, I was dealing with standards conversion today (23.976 to 25 for broadcast).

Curious to see what FCPX would do with this media, importing the 24p file to a 25 Project cause a noticeable shift in both time and pitch.

I imported the time coded audio separately, and played it against the obviously speed and pitched changed video with embedded audio. The files did not sync.

The audio remained in time and pitch, while the video didn’t. So the audio tc doesn’t seem to effect playback in FCPX, but video definitely does, and it seems rather buggy.

So, if your video is 24.0 and your audio is 23.98 and you are in a 23.98 timeline, FCPX might be changing the speed of the video without really reporting it. Resetting speed doesn’t do anything, hitting “automatic speed” screws it up a bit more (it goes the wrong way). Changing the speed to something like 96% (FCPX reports 96% but I know it’s really ((24000/1001)/25) which is just short of 96%. Turning off the “preserve pitch” checkbox actually pitched the audio back to normal.

I didn’t create the 24.0 vs 23.9 media that you have, but perhaps you are having similar issues, or a different combination/permutation therein.

But to answer your question, the time coded audio was not changed, only embedded audio.

Jeremy

[Walter Soyka]

[james Lackleter] “I’m wondering why the audio is syncing in Avid without resampling. Same deal, importing 24.00 FPS footage into a 23.98 timeline. Audio is syncing perfectly without resampling, even though I’m pulling down the frame rate. Any thoughts on why?”

What is your exact Avid workflow here, step-by-step?

Walter Soyka
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[James Lackleter]

I think FCPX has jumbled information and automatic pulldown/up is messing up the sync. FCPX seems to have problems with WAVE files in general, I did convert to AIFF which solved nothing.

Let me make a new test with Avid without using the proxies I made in FCPX with the Redcode, because I suspect FCPX pulled the footage and is not reporting it.

I will use RedCine, export the 24 fps clip and 23.98 and load it into 23.98 Avid Project with unaltered BWAVE and see what happens.

[Andreas Kiel]

I’m a bit late here.

But here some notes which makes things hopefully clearer.

Resampling and changing playback speed are two totally different things.
Samplerate for audio is the same as fps for video (not dpi for video) so let’s call it sps.

You can record audio at any sampling rate. But let’s take 48 kHz and 32 kHz here.

Record 60 seconds. Once recorded both files as stand alone files will play 60 seconds.

But forced to play back in a NLE timeline with a fixed sps of 48 the first file will playback correctly. The second one will play at 2/3 of the time (32/48). The other way round the 48 sps file will play longer in a 32 sps timeline.

Now for those poor guys who have to work with NTSC:
A real world second is not a second in NTSC video world. Its 1001/1000 seconds.
So selecting a 48k sps audio preset in a timeline preset means that a real world audio of 60 seconds is only 1000/1001×60. That means that those 60 seconds are only 59.94 seconds long. Its too short.

One way – if you stay strictly in NTSC – is to record audio with 48048 Hz. This way a real world 1 second audio will match a 1 second NTSC video.
Next option with BWAVE files is to change the playback settings in the BEXT header of the file. This won’t resample the file!!!

Now about the TC. BWAVE doesn’t have a TC in a way you’re used when handling video. It has a timestamp in the header. This timestamp contains the Samples after Midnight. Based on that the NLE will convert this to TC which can be displayed. So in NTSC world this might give a wrong interpretation.
For example if you have a “Start TC” of 00:10:00:00 (thats what you see on your audio recorder) it means that using 48kHz the timestamp is at 600×48,000 = 28,800,000 samples. Since the real sampling rate in the timeline though is 48048 Hz you have to divide the source stamp by this number. This results in 9.99000999000999 minutes. So for a 23.976 fps project this is a TC of 00:09:59:10.

So setting the Bext header and/or iXML chunk to the correct playback speed will keep everything intact. Again this is without resampling!
Using an AIFF there aren’t these options.

So you can edit all the stuff without resampling when using BWAVE. But resampling will happen when you export a final movie.

I wrote a bit 8 years ago when FCP started to read BWAVE metadata. Many of the things are outdated but the article explains a bit what happens with BWAVE.

-Andreas

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