I’ve read a solution on the forum that could fix the problem I’m having, but I still need a little clarification.
The issue is,
Audio sources are being recorded from different computers, they start off in sync, and later on, one of the tracks seem to magically unsync, one happening faster than the other.
I read that speed could be the issue.
I’m fairly audio and tech savvy, but this one goes over my head.
Is this an issue with the recording process or the playback while editing everything together? And how can I fix this easily?
Every soundcard (or USB microphone, etc.) has it’s own clock (oscillator) to generate the sample rate (i.e. 44,000 samples per second). There is no “perfect” clock and “cheap consumer soundcards” tend to be worse than [u]audio interfaces[/u].
And how can I fix this easily?
An audio interface on both ends should do it. “Easy”, but not free… And, if you are recording with microphones, you can’t use a “computer microhone” with an audio interface. Stage/studio mics are not interchangeable with computer mics so you might need to upgrade your mics.
How long are the recordings? How long before you notice it’s out-of-sync? If both people are using a good audio interface you should be OK with recordings of reasonable duration.
If you record and play-back on the same hardware you won’t notice a problem. If you record and play-back on different hardware you can get timing shifts and pitch shifts. The pitch shift is often a bigger problem with musicians because the timing usually doesn’t deviate that much over the length of a song.
If both computers have regular soundcards, it’s hard to know which one to blame (or which one is worse).
This can also be a problem if you record with a USB microphone and play-back on your soundcard. For example if you record a guitar backing track, then play the backing-track through the soundcard while recording a vocal. If there’s a clock problem, you can never get the pitch & timing to match because you’re singing at the wrong pitch and tempo to match the backing track which may OK, but it’s being played-back differently than it was originally recorded/performed.
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Pro studios use a super-accurate (expensive) [u]master clock[/u] and (expensive) interfaces/DACs/ADCs with master-clock inputs.
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BTW - “Latency” is a constant-delay. It’s normal for audio through a computer and it’s not a problem for recording & playback because there’s always a delay between recording & playback… It’s only a problem when trying to monitor yourself while recording and hearing a delay in the headphones.
And you reminded me of one of my favorite songs! Head over heels no time to think
Seems like the whole world is out of… sync
Yes, one of my cohosts (the one with the issue) has a fairly old computer. I’m embarrassed for him, so I will not include the accurate age of the PC.
I personally use an audio interface, which would explain why I don’t notice this with mine, but…
He is using a Blu Snoball mic.
I usually don’t notice the latency until roughly 45 minutes through our hour long, or so, episodes during editing.
With this being a free, non-crowdfunded podcast, I would enjoy not HAVING for him to spend a bunch of money on a quality mic, which can be pricey.
I know one of you said changing the timing in settings (I’ve researched this, and yes, it does seem like a quite finicky task), and that seems like it could take some time to get it exact, and might not solve the problem fully.
I know purchasing a new microphone and audio interface would be the quickest and easiest solution, but I’m just trying to weigh my options here!
Yes, one of my cohosts (the one with the issue) has a fairly old computer…
…He is using a Blu Snoball mic.
The age of the computer isn’t necessarily an issue. Plus, the USB mic has it’s own built-in clock (it essentially has a soundcard/interface built-in) so his “old soundcard” isn’t the problem. Unless, depending on your workflow, if he is using his regular soundcard for playback and trying to synchronize with an existing “backing track”, the backing-track at the wrong speed can mess him up. The same thing could happen to you if you use your regular soundcard for a backing-track.
I usually don’t notice the latency until roughly 45 minutes through our hour long, or so, episodes during editing.
I did some quick estimations/calculations. A “typical” quartz crystal has an accuracy of 100ppm. If I’ve done my calculations correctly, that’s a “worst case” error of about 1/4 of a second after 45 minutes. That’s enough to mess-up music and to become noticeable with audio/video sync. And it could be twice as bad if one system is slightly-fast and the other slightly-slow. So… I’d say that’s in the “normal ballpark”. That also agrees with my original understanding that two “good” soundcards/interfaces will stay in-sync for the length of a concert, but may drift-out over the duration of a concert or movie.
There are methods for tweaking crystal frequency, so for example a wristwatch maker can get better accuracy. And some larger equipment uses a little heater to keep the crystal temperature constant to avoid thermal drift.
But, I’ve never seen a clock/timing accuracy-spec for an audio interface or soundcard.
The old “clapperboard” trick is usually good enough for synching recordings with different sound cards / audio devices.
When you start the recording, create a “clapperboard” sound that can be picked up by both microphones (you might need to be a bit creative when working out a way to do this, but you’ve not given enough detail about your set-up for me to be more precise).
When you reach the end of your session, add another “clapperboard” sound in both recordings (at exactly the same time).
You can now look at the clapperboard sounds at the ends of the tracks, and work out exactly how much longer one is than the other. Then use the “Change Speed” effect to adjust the length of one to match the other (changing the speed only of the section between the start and end clapperboard marks).
And you reminded me of one of my favorite songs!