problems of synchronisation in audio

deleuze3

Well-known member
Hi, i have a question about recording audio, because i have a problem of synchronisation.

I shot a video of a show the other day.

The show consist of 2 rappers who sings in microphone over a "music track" that plays in big speakers.

I recorded the audio separately from the video.
I plugged my pcm-m10 audio recorder in the console.
Sadly, the audio-recorder only recorded the audio of the voices of the rapper (there was no audible music from the audio recorder)

So, i decided to ask the rappers to give me the "music track" they use for the show in wav format.

I tried to mix the voice audio i recorded from my audio recorder with the music track in wav, but it doesnt match well, its out of synch.

I wonder if the fact that the music was played live and were going out of the speakers make the speed of the music slower than the wav file they give me.

I tried to look if the khz and the bits were different between the music track and the voice track, they were, but i mix them at 44.1 khz and 16 bits.
So the khz and bits shouldnt been an issue then.


Someone can help me about this ?
Thank you in advance,
Julien
 
problems of synchronisation in audio (2)

problems of synchronisation in audio (2)

I also had a problem of synchronisation when i was recording a karaoke i did by myself.

Here how i was proceding :

I had earphones to listen to the "music track" (in mp3) and i was singing out loud over the music track.
I putted a video camera in front of me to record my singing.

When i edit in adobe premiere i putted the singing video-audio and i add the music track in mp3 from my computer (the one i use when i was listening in my headphones)

Strangely, i tried to mix the singing audio with the music track mp3, but it didnt match there was a problem with synchronisation. One of the file was too fast or too slow, something like that.

Can someone help me to understand better how come there was a problem of synch ?
Thank you in advance,
julien
 
Long story short, audio for video runs native at 16-bit/48kHz. If you try to place 44.1kHz audio in your NLE (in this case, Premiere), it's going to have some speed problems. Further, MP3 is not an editing codec. Your MP3 file(s) should be converted to 16/48 WAV or AIFF.
 
thanks!
But the problem i had at the show was not linked with video.
It was an audio out of sync recording issue.
Not sure if i explained myself clearly, its long to explain.
Thanks
 
Well, you recorded their vocals with a handheld recorder, and tried to mix the source music provided by the artists with the voice recording, and they weren't in sync.

You said that the two files did have different resolutions, but that you mixed them together at 44.1kHz. Which one wasn't 44.1, and how did you mix them? Either way, Adobe Premiere wants 48kHz. Further, your handheld recorder's clock likely isn't a perfect 44.1kHz or 48khz (whichever you used), so you can expect to see some drift over longer clips.
 
I understand Premiere wants 48 khz, thanks.
The wav music file was in 44.1 khz and 16 bits while the voice (recorded with hanheld recorder) was at 48khz 32 bits.
I opened the 2 files in adobe audition and rendered both at a projet at 44.1 khz and 16bits (i understand now that my project has to be in 48khz for premiere, but even so, there is still problem of sync.

The recorder clock must not be perfect (the audio-recorder pcm-m10). Its a 17 min track by the way.
Thank you very much, its highly appreciated
 
The recorder clock must not be perfect (the audio-recorder pcm-m10). Its a 17 min track by the way.
Thank you very much, its highly appreciated

That can be remedied by using time compression/expansion to compress or stretch the voice track to the length of the music track.
 
do you know what function should i use in adobe audition by the way ?
Thanks,
all this is interesting!
 
I don't use Audition, so... no, I do not.

However, if you go back to the source files and convert them both to 16/48, it may be easier to do in Premiere. Again, I'm not an Adobe user so I don't know the functions, but I do this in Final Cut Pro and in ProTools every now and then.
 
If you played the music wav file at the incorrect sample rate, it would play at a different speed.

I assume the music file is 44.1kHz and the video track is 48kHz (?).

I would start again - but first sample-rate convert the music file to 48kHz and then everything should be at the same sample rate and should then work fine.

(or vice versa if it's the other way round)
 
1) Get the backing track in WAV, not MP3. MP3 is not a production format, it is a "dumbed-down", release-only format. In your particular case, it is even worse because you will possibly be subjecting the track to further digital manipulation. And damage to the audio (from MP3 compression, or from rate conversion, or from time manipulation) is cumulative and sometimes increases geometrically each time you diddle with it.
2) All audio for video is 48K sample rate. If you can't get audio clips at that sample rate then they must be converted to that rate. ANY/ALL rate conversion comes with potential loss of quality (including time anomalies in bad cases). Rate conversion is to be AVOIDED if at all possible.
3) Premiere (and most NLE apps) will do rate conversion automagically for you if you aren't paying close attention to what you are doing. The rate conversion done by Premiere (or any NLE) is not necessarily the best, and you are tyically much better off doing whatever rate conversion is necessary explicitly with a proper audio DAW app so that you can control the parameters for best results.
4) Rendering a project in 44.1 just seems like a REALLY BAD idea. Dunno why anyone would want to do that?
5) We typically shoot more than one camera, even if it is just a locked-down "wide shot" just to have some variety in the video. 17 minutes of a single camera shot is what home movies look like. At any place where you switch between camera shots, you can "pull-up" the video to match the audio track.
6) When we shoot "music videos", the AUDIO track is the MASTER and the video is "pulled-up" (frames deleted or added) to conform to the audio. It is MUCH easier to make slight changes to the video without horrible consequences than to make changes to the audio track.
7) If the time problem is linear over the length of the clip (17 minutes continuous??), you could simply grab the tail end of the audio track and "stretch" or "compress" it in length in the Premiere timeline to match the video. This will force Premiere to re-sample the clip to conform to your new length. However, re-sampling is RARELY a good thing and making SLIGHT changes to length is PARTICULARLY PRONE to introducing undesirable audio artifacts. This method is NOT RECOMMENDED, but only included for completeness.
 
thanks everyone, those are precious information,
have a great day,
i put all these information in my reminder doc !
Thanks
 
4) Rendering a project in 44.1 just seems like a REALLY BAD idea. Dunno why anyone would want to do that?

It's the CD music standard, thats why folks go there. It's only a bad idea if you need to use it in a video project.

deleuze3 (does that mean that there are two others??),

Lots of recommendations but maybe a few questions would go farther to help you out.

If you start the track in sync does it drift consistently out of sync or does it drift in and out of sync?

If it a constant drift then you should be able to "fix" it by time expending/compressing the recording by some small amount.

Generally when things are sample rate converted they will stay in sync. But some applications will not do a true conversion and just play back the samples at a faster/ slower rate. 48 > 44.1 is a very small change but you should stll be able to hear the pitch shift if you listen closely. Drift by a non stable clock (with todays technology) is probably going to be a fraction of a second over 22 min at most. A speed change because of a conversion that wasn't is going to be much greater. So if your drift is large then maybe all you need to do is go back and make sure the conversions are done correctly.

If the drift varies then you kind of SOL. You can chop it up and resync bit by bit but that is a real pain.
 

4) Rendering a project in 44.1 just seems like a REALLY BAD idea. Dunno why anyone would want to do that?

It's the CD music standard, thats why folks go there. It's only a bad idea if you need to use it in a video project..

Yes, of course, I do both audio and video projects regularly.
But rendering out a VIDEO project in 44.1K just seems worse than pointless.

If I am using 48K source material for CD release, I produce everything for the CD in 48K, and then convert to 44.1K as the final step.
 
thanks everyone ! It's well appreciated.
By the way it was constant out of synch, i.e. getting more and more out of sync, so i will use the expand, compression function
 
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