Need help with mismatched sample rates

[Deadtide.com]

Okay, I'm completely lost and in need of additional grey matter to help me sort this out. I'll keep it short, so you don't get bored:


? I wote music in ProTools at 44khz, 24bit

? I bounced a track AIFF at 44khz, burned it as an audio CD and send it to my singer in London.

? He takes the CD to a studio that uses Logic "in OS 9" (not sure what version).

? Engineer says "How do you want this set up?"

? Singer says, "Best quality," not really understanding what the question is.

? No clue what the engineer did and have no way to contacting him.

? I get back a data CD with the separate vocal tracks for mixing.

? I imported the vocal tracks into ProTools and see that they are at 48khz instead of 44khz. Timing is off, pitch is off. I fought for over an hour the other night before giving up and can't seem to find the magic combination of settings and/or conversions to get the tracks to line up. I can get the pitch correct, but the tracks are way to too short. I don't think I ever got the speed correct...

? I have a sneaking suspicion that the tracks my singer sang over were sped up over in London, which is why I can get the pitch correct, but not the tempo/length/speed.


Does anyone have any advice on how to get this fixed without resorting to time stretching and/or pitch correction? I neither know nor understand exactly what happened, so I'm having trouble coming up with solutions on my own...

Thanks again!

[cassettefetish]

Build this:
http://negativland.com/audiogadgets.html

(Scroll down to "how to build a high fidelity audio device)

[Deadtide.com]

Build this:
http://negativland.com/audiogadgets.html

(Scroll down to "how to build a high fidelity audio device)

There was nothing on the page as you described. Not exactly sure how it would have helped if it were, either....or are you just being a smartass?

[cassettefetish]

well, not really.. build one of those devices, mail it to him, have him call you and you can get a pretty decent recording through the phone.

I guess I never would bother with all the converting and stuff if the singer sang it differently, originally. I understand it's a big hassle to re-do, but the studio screwed up, have them re-do it if they played the original file too fast. (It shouldn't be too hard to just play a pre-recorded song!!)

Maybe just send them a regular audio cd this time?

--Nick

[brew]

basically that studio f'd it up. they should have kept everything at the same rate, who knows why they didn't.

they may have just played your 44k files at 48k, and the singer sang over those sped up files, so you're stuck with a higher pitch. you could play the 48k vocal back at 44k and record that pass to 44, then bring it into your 44k session, and that should line up. but of course the singer will sound different a bit, because now they will be slowed down.

maybe the song won't sound so bad if you play your 44k at 48k...

if, however, they just saved the file at the wrong sample rate, you could convert to 44 and it should work out, but that doesn't sound like your problem.

basically you'll have to think through what they may have done, and try to reverse engineer as best you can. other than that, have them do it again. you can sample rate convert in wavelab or other software.

[kcrusher]

You can use ResEdit (either in OS9 or in classic mode in OSX) to edit the sample rate of those files.

Lauch ResEdit
Open the audio file
You'll get a window with 3 boxes - double click on the one marked 'STR'
You'll then see the sample rate - change it from 48000 to 44100.
Save
Reopen your session and all should be well.

[HoneyBear]

Protools is pretty good at converting between sample rates (especially in OS X) so I don't think the problem is on your end. You could try creating a new session at 48k and importing your old files into it (they will convert up) and then bring in the 48k vocals. I don't know enough about Logic to know whether the file has some sort of time stamp which might help with the alignment and I don't know if the track you were sent starts at the very beginning (making it much easier).

A suggestion for the future: when attempting this kind of work add a couple of clicks to the top of the track you are sending. You'll have a couple of clicks on the track you get back and will save yourself a bunch of headaches.

One other thought - was the file sent to you sent as data or as audio? If it was sent as audio instead of data this would have some strange effects (although I'm not sure if speeding up is one of them). If you can play the file on a CD player then it wasn't done right.

[joel hamilton]

The resedit thing works really well. I have used resource editors to hack all kinds of files to make them work in sessions, including sample rate as was talked about above. If you dont understand resource editing, dont fuck with it. You can really screw things up. Work on a copy of the original file in case you screw up as well.

resedit can be the most kick ass tool, or if you try to get really ambitious with it, it can be pretty screwy. I have never messed up with it because it was instilled in me from the start to work on a copy, not the original files.

Then again, only use this if the import is not working for you. Like if the speed is screwy, and you dont have a clock source (man I wont shut up about clock sources these days!!!:)) then try editing the file itself via resedit.

If you do have an external clock source that you can select the sample rate on, you can open a 44.1 session, and pull it up to 48k. That means if you "import and convert" files that are meant to be played back at 48 they will sound normal, and the ones created at 44.1 will play faster.

Do the same thing in reverse (48k pull down to 44.1) and see if they work that way.

OR:

Resedit.

Or:

Record the tracks to tape, or DAT or ADAT or anything that IS NOT connected to your computer. Make sure they sound right in whatever session with whatever sample rate. Capture on some device not connected to your computer via digital. Load back in the sounds in a single session, with one sample rate of your choosing.

Yes, it is another conversion, yes it is not optimum, but it takes the headaches away and makes it work. All you would have to do is line up the vocals.

There is probably enough headphone bleed if you limit the crap out of the track while you are sliding it around. Use something like L1 to KILL the vocal track (remember the plugin latency, i think 64 frames for L1) and lsten to the headphone bleed away....

/rant

[@?,*???&?]

Boy, I'm glad we are in this user friendly digital age!

I worked a session with Nick Launay once for this band out of Chicago called the Lupins. This was back in the analog days with two 24-track machines. Nick flew the lead vocal into the master from a demo reel the band had at a completely different tempo. He punched in at the console as I kept nudging the offset on the Lynx units for the tape machine lock. We eventually got the vocal over onto the master and it sounded fine. One would never have guessed it was actually at a different tempo originally. Alot of work, but I almost wonder if we don't create more than we should- especially now that we are working in this DAW world.

[kcrusher]

There is one complication that may be prevent you from doing a simple sample rate conversion and that's this:

If the sample rate of the file is TAGGED differently than the actual sample rate of the file. Let's say I have an audio file that's 44.1k, but it's tagged as 48k, it will sound fast, as it's playing a 44.1k file at 48k. If I then try to do a sample rate conversion, it will STILL sound fast, as the SR conversion essentially is still playing it back at 48k (because it's tagged that way) and resampling at 44.1.

So, depending on what the actual problem is, you may not be able to use SR conversion and using ResEdit would be the only choice. Oh yeah - you CAN quickly change the SR tag using Pro Tools in OSX by going to Windows>Show workspace, locate the audio file and under the 'sample rate' column, just type in the correct sample rate.