E_S_D Posted November 10, 2007 Share Posted November 10, 2007 Hi everyone, I'm new to Logic, and to this forum, so thanks in advance for any replies that you might give to my question(s). I used to use Cubase SX on a Windows-based system, but with the arrival of Vista I decided to switch to Mac, and at the same time change music software, to see if it brought anything new to my creativity... I'm having trouble getting my head round one particular aspect of Logic, namely working in the same session with audio files of different sample rates. I mixing a session of a live concert recording: some tracks were recorded into Logic at 24bit/44,100. Some were recorded onto an external recorder at 24bit/88,200. OK, probably not a good thing to have two different sample rates to start with, but that's done now... I have tried converting the 88,200 within the Audiobin. When I do this, and line the tracks up, for some reason, the 2 slowly drift out of time. After about 30 minutes of audio, there is about a 50-100 ms delay between the 44,1 and the newly converted 88,2 tracks. Is this normal? I've always been able to convert and switch between sample rates in Cubase or CoolEdit, no probems. It also appears that you can't choose which sample rate you want to export to within Logic. I wanted to import the 88,2 files into a new session, and export them as 44,1 files, to see if this solved the problem. The only options available are bit depth and file type. No sample rate choice? Am I missing something? I'm now back in a new session with a setting of 88,2. Now the convertion function within the Audiobin is greyed out, so I can't convert the files anyway. Sorry fr the long post, but this is really doing my head in, and I don't have a seperate audio editor installed on my Mac. I've tried Audacity, but it won't open 88,2 files... Thanks a lot for your help... Link to comment Share on other sites More sharing options...
marcel72 Posted November 10, 2007 Share Posted November 10, 2007 Try this: Open a session with the files that you want to convert the SR of (the 88.2kHz files in your example) and load the files you want to convert. Solo a region, and click the bounce button on the corresponding output's channel strip. The resulting dialogue will give you an option to specify the sample rate of these bounces. Choose the SR of the session you want to run your mix from. Load the resulting bounces into your mix session. The delays you are experiencing may not go away, tho. Although the clocks on the two systems should add up, they may not without some enforced sync during the original recording (ideally, you would use a wordclock signal to keep the two devices in sync through the recording process, when recording a single performance with multiple devices). Your only other option is to make some judicious edits during the session to try to realign your audio files, in which case you run a high risk of introducing phase anomalies. Do this once you are in a session that is composed only of audio with the same sample rate... Don't try to make recordings of a single performance with multiple devices without linking their clocks, and always use the same SR. I really hope that lesson didn't cost you too much. On the plus side, you'll never make that mistake again, will you? Link to comment Share on other sites More sharing options...
Spunkadellic Posted November 10, 2007 Share Posted November 10, 2007 two sample rates in a session....thats nothing ...i screwed up bigger!!!! recently i recorded a session into pro tools at 24/48 - just as i always do HOWEVER......my apogee clock was set to 44.1 so now i have 48khxz files that dont play the right speed when theyre at 48 theyre too fast converting them to 44.1 makes them even faster....IM SCREWED!!!!!! Link to comment Share on other sites More sharing options...
E_S_D Posted November 10, 2007 Author Share Posted November 10, 2007 Hmm, OK, fair enough. I'm a little surprised that the two seperate sessions don't add up at the end of the day. It seems like a big delay to me, even over such a long file. I tried the bounce feature - thanks for that, it's sorted out the issue wih how to export files in the sample rate of your choice. But yeah, the problem remains. Just too bad really. You reckon recording on two seperate recorders at the same sample rate would be OK? Cheers, E_S_D Link to comment Share on other sites More sharing options...
fader8 Posted November 10, 2007 Share Posted November 10, 2007 I have tried converting the 88,200 within the Audiobin. When I do this, and line the tracks up, for some reason, the 2 slowly drift out of time. After about 30 minutes of audio, there is about a 50-100 ms delay between the 44,1 and the newly converted 88,2 tracks. Is this normal? Absolutely. In fact I'm surprised it's only that much. You're lucky. As Marcel pointed out, 2 clocks left to their own devices will never run in close enough sync. It doesn't take much error in either one to give you this discrepancy. I think the editing solution Marcel suggests is your best bet. Take the new files that you've converted and chop them up into regions about 2 minutes long. Work with each one to get them lined up with the reference tracks. At least this will minimize any audible "musical" timing problems. I'm assuming that the 2 systems were recording different microphones so phasing isn't really an issue. Link to comment Share on other sites More sharing options...
fader8 Posted November 10, 2007 Share Posted November 10, 2007 You reckon recording on two seperate recorders at the same sample rate would be OK? No. Only if they have the same wordclock source. You'd end up with exactly the same problem. Typically when you set a device to 88.2, it's just dividing a 44.1 clock anyway. So the same drift can be expected. Link to comment Share on other sites More sharing options...
marcel72 Posted November 10, 2007 Share Posted November 10, 2007 two sample rates in a session....thats nothing ...i screwed up bigger!!!! recently i recorded a session into pro tools at 24/48 - just as i always do HOWEVER......my apogee clock was set to 44.1 so now i have 48khxz files that dont play the right speed when theyre at 48 theyre too fast converting them to 44.1 makes them even faster....IM SCREWED!!!!!! If you duplicate 'the setup', can't you play them back and bounce them out of PT? I guess you were using LE? The HD rigs I've worked on wouldn't let you do that, just like Logic wouldn't... Link to comment Share on other sites More sharing options...
E_S_D Posted November 11, 2007 Author Share Posted November 11, 2007 OK, thanks everyone, sorted that one out. Realignement + narrowing the stereo field + a bit of reverb to smooth it all out, and it sounds pretty good... I'll be recording everything onto the same machine next time... Link to comment Share on other sites More sharing options...
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