Audio and frame rates?

R

ronin99

Active member
I'm trying to get a better understanding of how audio and frames rates work together, take an H4N for example I could record video at 23.976 or 29.97 on a camera, and the audio file from the H4N would match to either one of those. I don't have to set a frame rate for that (I'm not concerned with jam syncing time code or anything).


However, what happens if your film is shot and edited at 23.976 but your sound designer does all his work in 29.97? Will there be a four frame sync difference when you get the audio file back? I'm not familiar with most DAWs, do you need to set a frame rate there before you begin work?
 
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Audio is not dependent on frame rate, but on sample rate. As long as your audio is recorded at 48kHz, and that's what both the NLE and the DAW are set to utilize (48kHz is the native sample rate for digital video, so the NLE should default to this), there shouldn't be an issue.

In productions with sync audio, the frame rate is just metadata and, though it does need to be consistent between sound recorder and camera, it has no affect at all on how the audio is recorded.

And yes, wherever time code is involved between video post and audio post, the system needs to be set to the same frame rate just to avoid confusion. And how will the DAW maintain time code values when importing an OMF or AAF into the session? And, assuming there's a BITC window in the reference video file, how is that going to help navigate the time line? What kind of sound designer doesn't follow video for frame rate?

The simplified version: frame rate doesn't affect the speed of audio playback at all, but inconsistent frame rate between audio and video systems can create problems with sync points in post and make time-code-based systems pretty much a headache to deal with.
 
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I'm looking at my Sound Device user guide which says that the recorder can record audio at the following sampling rates
- 44.1
- 47.952
- 47.952F
- 48
- 48.048
- 48.048F
- 88.2
- 96
- 192

When I do record myself (which doesn't happen frequently) I always use 48kHz - so far haven't had any problems. I tried to read about several rates around 48kHz (47.952 and 48.048). I actually (think :) understand (at least conceptually) the F mode. But could never get why 0.1% deviation on both sides from 48 are offered in the first place?
 
The reason for the 0.1% speed difference goes back aways. But instead of thinking of frame rates VS sample rates it's better to think of "US video SPEED" VS "film SPEED".

A little history will help. B&W TV ran at 30fps, which worked well for transferring too and from film, important since there wasn't really video tape. Then they decide to go to color. There was a big behind the scenes battle and the lesser system won. Mostly because RCA/NBC wanted to keep their customers and the better system would have made color broadcasts incompatible with older B&W TV's. To make this work though they needed to slow the frame rate slightly so they could fit the color information in the signal but not so much that the older B&W TV's couldn't sync to it. So 30fps became 29.97 fps. That is fine and dandy but now to transfer film to video you need to slow it down slightly.

That is not a big deal IF you are transferring a finished film, BUT since a lot of TV shows were shot on film and then edited on video you had a problem for location sound. If they record at 48k then in post someplace you have to clock it slower so it will sync. At this point the story splits. For film folks who were doing sound post to video tape you kept your location sound at 48k and you clocked slower while editing and then went back to normal when mixing because films were mixed to film so you are back to "film speed". But if you were working to a video final you needed to go through a conversion at some point. Often everyone stayed in "pull down" and you mixed through an analog board and that is how you got the SR converted. As everything was going digital some smart person came up with the idea that IF you over clocked the recorder a bit then when you played the recordings back at 48K they would sync with the video and nobody needed to deal with different sample rates in post. So many high end recorders for media work got some odd sample rates. You would only use them for very specific work flows. So IF you are shooting film and going to end up going to NTSC video then you over clocked so you would sync. IF you were shooting NTSC video and were heading for film you under clock so that, again, you will be in sync.

As long as you stay at the same "speed" you record at 48k and you are fine. Nowadays video can shoot at just about any frame rate so any well planned shoot should be shooting at whatever "speed" they want to end up at.

For extra credit...

Drop Frame TC was created to compensate for the difference between 29.97fps and 30fps. It throws skips frame numbers periodically because you can't have fractional frames so TC would read 29.97 and count it like it was 30fps and that meant that the TC would drift out of sync with "real time". By skipping a number every once in a while the TC would stay very close to real elapsed time. Very important for broadcast shows. It's basically the same idea as leap years.

As extra extra credit, you should be VERY careful about using the pulled down/up sample rates. Some "smart" NLE's will detect it and SR convert it on the fly thus defeating the whole reason you used it in the first place.
 
The odd-ball sample rats (47.952, 48.048, ect) are normally for dealing with 24-25fps in film pull up/down, music videos and such usually.
 
Just so you know, if you cut picture in one frame rate, do sound editorial in another, and then give final sound to picture to put them together, there will be drift. Happened to me on the very first short film I ever worked on. I was told pictures was a 30 when it was really at 29.97. I worked at 30. Drift ensued when we handed off the final mix. ALWAYS make sure your frame rates are discussed and understood. 23.976 does NOT equal 24, and 24 does not equal 23.976. I've had so many camera people say "yeah, I'm at 24", when they really mean 23.976, so when I hear that I always ask "do you mean 24, or do you really mean 23.976?", to which I usually get the answer of "uuuhhhh...I don't know. How do I find that out?". That's happened to me everywhere from MOCAP stages, to low budget stuff, to $200 million union features....
 
ronin99 said:
However, what happens if your film is shot and edited at 23.976 but your sound designer does all his work in 29.97? Will there be a four frame sync difference when you get the audio file back? I'm not familiar with most DAWs, do you need to set a frame rate there before you begin work?
Click to expand...

In order for the 23.976 material to be played correctly at 29.97, the material needs to be converted to 3-2 pullup sequence. What this does is take 4 23.976 frames, and spread them across 5 29.97 frames. (Which has been done since NTSC began and 24 fps movies converted for broadcast...).

Anyway, at 23.976, there are 2002 audio samples per frame @ 48K/s, and for 4 frames that would be 8008 samples.

For 29.97, there are 1601.6 samples per frame @48K/s, and after 5 frames, one has 8008 samples.

In other words, by the time all is said and done in the 3-2 conversion the number of audio samples across 4 frames at 23.976 equals the same number of samples for 5 frames of 29.97.

And all is well in the world...

On running 23.976 material at 24 fps without compensation, the reasoning is thus.

There are 2002 samples @ 23.976 from the above calculations.
There are 2000 samples @ 24 fps.

So after 1 second of play @ 24 fps, the audio is 'short' 48 samples. After 10 seconds, it is short 480, after 20 seconds it's short 960... which is almost 1/2 a frame, and so can impact lip sync, or other audio/visual cooccurances.
 
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@Noiz2 Thank you for your very informative answer. Looking at that 0.1% I suspected where it all came from; however, I still had an uneasy feeling that there were a chance I could do something wrong. Now I know that with I do, I can go ahead with 48kHz!
 
I remember learning about all this backstory a long time ago...but man...this stuff still gives me a headache.

To keep it sane I ONLY record audio at 48khz and I make sure to ask EXACTLY what the delivered frame rate is, like was mentioned above.
Never had any problems when sticking to this.
 
Some of this goes way over my head but I appreciate the knowledge. I think Dave answered what I needed to know regarding post is that there will be drift if we edited in 23.976 and the post sound does 29.97.

Unfortunately, this is the exact position I'm in.

The so called "sound designer", an amateur, can only work in 29.97 (yes, I know, hire someone else, a professional, I would if it was up to me and the person hired wasn't related to the boss), so is my workaround to take the mixed down WAV file i eventually will receive from this person into wave agent and make it 23.976? any other ideas on how to keep sync?
 
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One can change the sample rate to compensate for the time difference. A change to 48.048 should not be a noticeable pitch change with dialog. Though for 23.976, no change should be needed, unless the editor forked it up
OTOH, it would drift w/ 24 FPS film.
 
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ronin99 said:
Some of this goes way over my head but I appreciate the knowledge. I think Dave answered what I needed to know regarding post is that there will be drift if we edited in 23.976 and the post sound does 29.97.

Unfortunately, this is the exact position I'm in.

The so called "sound designer", an amateur, can only work in 29.97 (yes, I know, hire someone else, a professional, I would if it was up to me and the person hired wasn't related to the boss), so is my workaround to take the mixed down WAV file i eventually will receive from this person into wave agent and make it 23.976? any other ideas on how to keep sync?
Click to expand...

Then tell them to get Pro Tools HD. It can handle any standard frame rate. I've never ever heard of someone only being able to work in 29.97 that was actually doing real work. Honestly, if I were in this situation, I'd hand off what I recorded and let them know they need to convert it to 29.97 since the sound designer can only work with that frame rate. It may seem jerk-ish, but having been down similar roads like this before, here's why.

If I say that I'll do the conversion, now I'm responsible for making sure it works properly wherever it is going, and if I don't feel confident in the person that I'm delivering the files to handle whatever I give them, no matter what it is, then I'm going to get thrown under the bus no matter what. So, If you rolled at 23.976 on production, then that's what you hand off. You tell them that and then collect your paycheck. There are post production tools to help with post production that they can use. Now, if the post guys were capable and nice and didn't get their jobs because of nepotism, then I'd be happy to help with whatever needs to get done. I've been burned too many times by amateurs, so once bitten twice shy. I won't be a complete jerk. I'll point people in the direction of the stuff that'll help them, but that's about it.
 
Always good to work out your shooting and post workflow before you start shooting.

NTSC drop-frame has been a nightmare since digital started to come into play in the early 90's and with so many options available everyone needs to know who is shooting in what format and if any pull up or pulldown is required as it can be done on the framerate or the sample rate.

We have it easy here in the Uk as it is generally either 50i, 25p or 24p at 48k but you will still get some musician who is working at 30 frames 44.1 and throws you their score at the dub stage.
 
Have they added omf import/export back in?

Aside from the HD version, you needed to add the complete production tool kit to allow that before.
 
David W. Jones said:
Have they added omf import/export back in?

Aside from the HD version, you needed to add the complete production tool kit to allow that before.
Click to expand...

Not as of PT10, at least. They added pretty much all of the Production Toolkit functionality into the standard software.
 
ronin99 said:
The so called "sound designer", an amateur, can only work in 29.97 (yes, I know, hire someone else, a professional, I would if it was up to me and the person hired wasn't related to the boss), so is my workaround to take the mixed down WAV file i eventually will receive from this person into wave agent and make it 23.976? any other ideas on how to keep sync?
Click to expand...

What tool is this person using? what tools do you have?

Since I use Adobe products, Premiere can take in any number of formats. The Adobe sound tool, Audition, can also take in any number of formats.

I don't know anything about Vegas, but I believe it also can take in any number of formats... so, unless they are using some freebee tool, why the insistence on 29.97???
 
"I don't know anything about Vegas, but I believe it also can take in any number of formats... so, unless they are using some freebee tool, why the insistence on 29.97???"
Yes Vagas can have different frame rates. Different audio sample rates and bit depths as well, even on the same track.
 
Rick R said:
The odd-ball sample rats (47.952, 48.048, ect) are normally for dealing with 24-25fps in film pull up/down, music videos and such usually.
Click to expand...

Not so Rick. 24 <-> 25 is a 4% change, the 0.1% change is for NTSC video <-> 24 frame film.
 
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