TOPIC:

Hi guys,

I'm shooting some 24p 4K footage on a Sony A7s2. And I am recording audio separately on a Zoom H4. When I go to sync them up in post, I am finding that the audio is every so slightly slower than the video. When synced, a 20 second clip stays bang on, but when the clip is 2, 3, 4 mins, the audio slowly goes farther and farther off time with the video.

Does anyone know what setting I have wrong on the Zoom / what I have to adjust to ensure I'm recording video and audio at exactly the same speed?

Phil

What are the audio sample rates of the two recordings? The Zoom should be set to 48K probably to match the video. I'm assuming that's what the video is shooting,. That's the most likely culprit.

Is your project set to 24 fps? My guess is that the camera is actually recording at 23.98 fps, so dropping it into a 24 fps timeline speeds upp the footage ever so slightly. Try creating a new 23.98fps project and trying out if it syncs up better with your audio.

It was indeed in stamina mode and recoding in 44.1Khz... Don't know much about sound. Does this impact the playback speed?

It's been switched to 48Khz... this will now sync up with 24p?

Thank you !

Sample Rate is the most common culprit of audio drift and sync issues. Sample Rate is how many "samples per second" the sound is recorded at. Like frames per second in video. Problem is, they don't get handled like video frames. All audio has to be the same Sample Rate to match up in the edit to keep everything in sync.

44.1 means the audio is being sampled 44,100 times per second. 48kHz is 48,000 samples per second.

Drop those into a Timeline, and the samples (in all NLE's) are all used, and timed out according to the Timeline. So that 44.1kHz per second of samples gets stretched out and re-timed (so to speak) as if it were native 48kHz.

It would be nice if NLE's could start to treat audio sample rates just like video frame rates and reconfirm them "properly" to keep sync in a Timeline of a different sample rate.

FCPX.guru wrote:

Drop those into a Timeline, and the samples (in all NLE's) are all used, and timed out according to the Timeline. So that 44.1kHz per second of samples gets stretched out and re-timed (so to speak) as if it were native 48kHz.

Can you give an example when this happens? I've never seen any NLE do this. Mixing 44.1 and 48 kHz audio in the same timeline works fine for me. Ripped CD tracks have always worked as expected in 48 kHz timelines, for instance. If a 44.1 kHz were retimed to play back at 48 kHz, that would speed up the track by about 8%. That means that sync would drift by one second every 12 seconds. A two and a half minute song would play back in just about 2:18. Plus the pitch would be way up. I've never seen this behavior.

FCPX.guru wrote:

It would be nice if NLE's could start to treat audio sample rates just like video frame rates and reconfirm them "properly" to keep sync in a Timeline of a different sample rate.

This is exactly how FCPX works, in my experience.

The way sample rate does affect sync is that different devices, using there own clocks, don't necessarily sync up perfectly. That is; 1 second on one device might be considered 1.002 sec on a second device; or 48000 samples/sec on device A is considered 47904 samples/sec by device B. When playing recordings from both devices in a single timeline, both tracks are played back at 48 kHz so they drift out of sync. So, it is the devices' difference in actual clock speed, more than sample rate settings, that cause issues.

Take two shots for a multicam, two different sample rates, they will drift out of sync. I see it weekly.

FCPX.guru wrote:

Take two shots for a multicam, two different sample rates, they will drift out of sync. I see it weekly.

I have heard this many times but I don't see it myself. I just imported a 27 min 39 sec 44.1 Khz mp3 file from a Tascam DR07 and a 44 min 23 sec 48 Khz wav file from a Tascam DR10L (both made of the same event) and they synced perfectly using an FCPX sync clip with no drift. I then created a project and dropped in the 48 Khz file which set the project audio characteristics to 48 Khz, then dropped in the 44.1 Khz file, manually synced the beginning and they were stayed in sync to the end.

However -- the indicated program length of each file differed from that shown in FCPX, assuming we trust the program length as shown by Finder, MediaInfo or Invisor. The difference was 27 min 39 sec 533 mS reported by Invisor or Finder vs 27:34:01 and 45 subframes in FCPX. The wav file was 44 min 23 sec 95 mS reported by Invisor & Finder, vs 44:20:10 and 0 subframes reported by FCPX. I'm not sure what that means. Subframe display is enabled in preferences.

We tend to assume the "timeline" time or program length shown by Quicktime player is true clock time, but in some cases this is not true. The timecode shown in FCPX is more accurately called a "time like scale" and just because a frame is labeled 00:00:01:00 does not mean 1 sec has passed. Rather something akin to 1 sec has passed, and the timecode is actually an indexing system for finding material. Apple explained this back in the FCP7 documentation: documentation.apple.com/en/finalcutpro/u...ction=6%26tasks=true

The bottom line is you cannot trust anything except what you actually measure, so the only way to be certain is play the file and time it with a stop watch.

It is true the audio data is sampled at two different rates but the playback should always be true to real time, IOW 30 min. of 44.1 Khz material plays in the same time as 30 min of 48 Khz material. When those two sample rates are combined on a timeline they must be conformed somehow. The software could stretch or shrink one or the other by 8.8% to make the samples line up but FCPX does not apparently do that. If it did, then at a normal playback rate one or the other audio track would be tone shifted and the program length would be 8.8% different.

I don't really know what FCPX does to conform 44.1 and 48Khz audio sample rates but it's not stretching or shrinking the track length by 8.8%.

One possible explanation is the smallest FCPX time granularity is actually 1/80th of a frame at the subframe level. You can't actually see individual audio samples. At 29.97, each 1/80th of a frame is 0.417 milliseconds. In each of those sub-frames, at 48 Khz there are 20.02 samples per sub-frame. At 41 Khz there are 18.393 samples per sub-frame.

So in either case there will be a periodic "leap sub frame" situation where occasionally a sub-frame will contain more samples than others. From this standpoint it's not really necessary that each audio sample line up vertically with adjacent tracks because you can't access that level of granularity anyway.

I think in many cases audio drift is caused by drifting timebases in each device. The root source of the timebase is an inexpensive quartz oscillator, just like a cheap quartz wristwatch. Oscillators don't have perfect frequency stability, which is why two quartz watches will gradually drift apart. Actual surveys show some quartz watches drift by 2 sec per day (much worse than the spec), so two watches with opposite drift could move apart at 4 sec per day. For two audio recorders this equates to 100 milliseconds per hour, easily enough to cause a lip sync issue on longer programs.

It is possible to make precision temperature-controlled crystal oscillators that have low drift, or use GPS-corrected timebases, but I doubt most cameras and audio recorders use those. That said, the two Tascams I mentioned above are inexpensive, yet they didn't drift any in 30 min. Maybe most of them are fairly stable but older or sick units develop drift.

I’m new here and realise this is an old post, but I came across something similar today.

Filmed a project with the camera settings set to 4K, 50p (video), I used my Rode Wireless Go II kit to record the audio. The receiver was plugged directly into the camera via xlr to jack cable, recording at 48hz.

In Final Cut, the audio recorded directly into the camera via the xlr/jack cable syncs perfectly.

However, I was using my pair of Go II mics in split mode (recording two mono channels, two mics, one on each of my two interviewees) and one mic slipped and turned onto the body of one interviewee which meant their thumping heart beat ruined both their answers and the answers of the other participant coming in via the receiver. No problem, I thought, I’ll download the backup audio recorded on one of the microphone which remained safely in position, however….

When I imported the back up recording from the mic into Final Cut, I experienced extreme drift, 2-3 seconds after 5 seconds or so of playback.

My FCPX Project was set to - audio @ 48hz, and video @ 4K at 24p. The audio would not sync. However, after exhausting all other options, I decided to start a new project - audio @ 48hz, and video @ 4K but this time at 30p, suddenly the audio drift was no longer an issue and the backup recording remained perfectly in sync.

If the frame rate has nothing to do with audio and audio drift, why did changing the frame rate from 24p to 30p fix my issue?

Frame rate in some cases can affect sync drift due to rate conforming. The previous discussion focused mostly on audio sample rate and whether the timecode format was drop frame or non-drop frame. Neither of those are video frame rate.

You shot video at 4k/50p and initially imported that to a 24p project and tried to sync with external audio. 50p is a 2x multiple of 25p, so that must be rate-conformed to the 24p timeline. The algorithm for rate conforming between 24p and 25p is a 4% speed change with audio pitch compensation. That rate conforming caused the video to be out of sync with the audio since the video was playing 4% slow. There is no other good algorithm for rate conforming between 50p (or 25p) and 24p. The advantage is it plays smoothly with no motion cadence issues. The disadvantage is the audio sync issue you observed.

You then put the 4k/50p video in a 30p timeline. That also requires rate conforming similar to 25p in a 30p timeline. In that case the NLE (FCP in this case) cannot use a speed change but uses frame discarding. That does not get out of sync with external audio but causes an uneven frame cadence. You can see that if you single step in the timeline -- certain frames will be discarded, inducing a slight discontinuity.

Ideally you want to know the distribution frame rate of the final product and the cameras should shoot in that frame rate or a 2x multiple of that, and you edit in that frame rate.. E.g, for 23.98 distribution you shoot in 23.98 or 59.94, and edit in 23.98. For 25p distribution you shoot in 25p or 50p and edit in 25p. That avoids audio sync issues and produces smoother motion.

Because cinema is 24.0p and US TV is 29.97i or 59.94p, they use 3:2 pulldown to conform that for broadcast, where (in the progressive case) each 4th progressive frame is duplicated. In the interlaced case, they use frame blending: en.wikipedia.org/wiki/Three-two_pull_down The worst conforming path (from a frame cadence and motion standpoint) is 29.97p to 23.98p

Hello Friends... I recorded a live concert using my Canon R5 at 8K - 29.97. and I recorded the audio off the soundboard at 32 bit 192 sampling. Never had a problem in the past syncing up. Today, when I went to sync up, I noticed every 20 minutes, the Soundboard audio was 1 second off from the video audio. Is there any way to fix this. I use FCPX on a MacBookPro M1. I am desperate as this is a concert of a Grammy Award artist and I don't want to let her down... Cost no object... Thank you...

It sounds like you have a timing base discrepancy between drop-frame (29.97 fps) and what you may have used for the soundboard recording (30 fps). That difference is 3.6 seconds per hour. Trying to fix would require knowing all of the settings for the sound recording device, including the make and model of the recorder. There are some ways to address this.

Make sure the FCP timeline is 29.97 fps. If so, use a tool like MediaInfo or Invisor to verify the original camera file and audio file have the expected parameters. apps.apple.com/us/app/invisor-media-file...or/id442947586?mt=12

If you aren't sure, past the results of those here.

29.97 video can be either Drop Frame (DF) or Non-Drop-Frame (NDF). Audio doesn't really have that attribute but some recorders encode that metadata. Another possibility is there's a DF/NDF mismatch and that's causing a sync problem.

1 sec drift over 20 minutes is 30 frames per 20 minutes or about 1.5 frames per minute drift. That implies that after 2 or 3 minutes, a lip sync problem would be visible to close observation. 3 minutes would be 4.5 frames off, and that should be easily visible. Does that happen?

Another cause is timebase drift. On any camera or recorder, the fundamental timebase is generated by a quartz oscillator, similar to a quartz wristwatch. Those are manufactured with a certain plus or minus tolerance. However a second of drift per 20 minutes is a lot. That implies the cause may be elsewhere such as DF/NDF or something else.

The audio sample rate normally is not a factor for this.

If you are looking for a quick fix, you might be able to retime the audio to stay in sync with the video. It would only take a minute adjustment, such as 0.001. I'm not sure the FCP UI will reliably accept that many digits of precision, but Resolve will.

joema wrote:

29.97 video can be either Drop Frame (DF) or Non-Drop-Frame (NDF). Audio doesn't really have that attribute but some recorders encode that metadata. Another possibility is there's a DF/NDF mismatch and that's causing a sync problem.

Joe, isn't 29.97 fps always supposed to be drop-frame, by definition? I've only seen an audio recorder set to 29.97 non-drop-frame when it couldn't record 23.976.

Good points about timebase differences/drift vs. timecode mismatches...

On our Sony A7SIII and FX6 cameras, 29.97 and 59.94 can be either DF or NDF. They would normally be DF, but if synchronizing timecode with other cameras shooting 23.98 (which is always NDF), we have to set the cameras running 29.97 or 59.94 to NDF.

If DF/NDF is mismatched at 29.97, that is 3.6 sec per hour or 1.2 sec per 20 minutes, which is roughly what joeray is seeing.

However I thought that would only make a difference if syncing on timecode, not sync by audio waveform. If he's only tried sync by TC, maybe try sync by audio.

If that doesn't work and if the video is 29.97 DF and the audio (which normally does not have a true frame rate) contains some metadata stating 29.97 NDF, the video TC can probably be rewritten to NDF using the 3rd-party QTChange utility: www.videotoolshed.com/handcrafted-timecode-tools/qtchange/

Obviously, the metadata should be checked on both files and the files duplicated for safety before trying anything.

joema wrote:

On our Sony A7SIII and FX6 cameras, 29.97 and 59.94 can be either DF or NDF. They would normally be DF, but if synchronizing timecode with other cameras shooting 23.98 (which is always NDF), we have to set the cameras running 29.97 or 59.94 to NDF.

Okay, cool, that jibes with what I said.

joema wrote:

If DF/NDF is mismatched at 29.97, that is 3.6 sec per hour or 1.2 sec per 20 minutes, which is roughly what joeray is seeing.

That was my initial thought...

Thanks for the clarification, Joe!

@joeray, one way to address your issue, if applicable, is to use the Elastic Wave functionality in DaVinci Resolve. You could import a camera clip (with onboard audio, or just the audio) and the soundboard-based audio, then go to the Fairlight Page (or possibly the Edit Page). Then, you could apply the Elastic Wave function/effect (control-click-hold or control-rightclick-hold and choose Elastic Wave). Elastic Wave won't affect the pitch. You could then bounce the fixed audio and import into FCP...