andyjohnmm

Thank you very much. Andrew

Hoping I worded that subject title correctly.. Recording a vocalist, I ran the song many times and let her sing it all the way through - onto the same track - and the "overlapping" takes appeared as we went. Now doing a comp, and running into an issue. For the most part, it's easy for me to choose and highlight the proper section of a take and move on, though some sections of takes I would like to use but they are slightly off time-wise. Is there a way for me to move the "takes" around that I'm highlighting, thus placing them correspondingly in the comp track at the top? Many thanks.

I see, yes, okay. Using the direct monitoring from the interface, or in effect "no" monitoring by just hearing what's happening in the room outside of the headphones, is it safe to say that file will be placed sample-correct using - in this example - a 512 buffer size?

I see, yes. Recent test, using 512 buffer size, with direct input monitoring from interface, the audio seems to be placed correctly. Thing is, if I do the ping test or a test using a recorded transient - it seems to suggest that I should be setting the delay slider to place the audio early. Are you saying this test is not applicable if I'm using direct monitoring? And the audio clip samples in drum machine designer - I can find information on regular track clips being affected when changing tempo of session, but what about the samples I've assigned to cells - do they stay at the original quality and pitch I originally recorded them at if moving the tempo around? Thanks again.

I see. Okay. Using direct input monitoring from the interface - is there a chance the audio recording would somehow end up being placed early that way more so than monitoring from logic? Also, do samples I created and am using within drum machine designer remain unaffected if changing the tempo the session? Thanks.

Okay. Will attempt that. Wondering, is there any instance where Logic will put a recording early - "ahead" of the grid - even when the delay slider is set to zero? Really focusing in on getting the claps on beat, with slider set to zero, though they still seem consistently placed early, just slightly. This is at 512 buffer setting.

Okay - unsure how to proceed. Are there other tests or methods by which I can solve this issue? I still don't know if Logic is placing my audio correctly. Thank you.

Tried the slider at 0 samples. Click is blaring in the phones and my clap timing is fine - still seems all hits are consistently being placed ahead of the beat, early, all of em.

With the 512 buffer size selected, I did a "ping" test with the i/o plugin and adjusted the delay slider accordingly (to -50 samples). Using the direct input monitoring from the interface, with Software Monitoring off, It seems the recording is still being placed early (clapping to the click). Same thing happens when I use Software monitoring and no direct input sound at all. Is it possible the recorded audio is still somehow being placed early?

Will turn this off and attempt a test with 512 buffer size and using direct input monitoring. Thanks.

Yes, went through the "heavier" tracks and froze them. So that is in place, though, I'm still wondering if I say, set my buffer to 512, and do a test with running a signal out and back in - and delaying the recording with the slider one way or another to accommodate the discrepancy - will this be a final say in how I should have that slider set - and then using that buffer size from now on I'll know to leave it there and be comforted knowing my recordings are "true" and displayed/playing correctly down to the sample? All of that using the direct input monitoring on my interface? I'd like to be able to continue using it. Many thanks again.

Thanks to all replies. Being deeper into the session, with multiple software synths and many tracks, running the session with the buffer that low causes overload. Is there a way to get a definitive delay adjustment using my direct monitoring from the interface (which I'm most accustomed to) but use say, 512 as my buffer selection? Seemed the audio was actually ending up ahead of the beat, not late, as previously stated (even with the delay slider set to zero!). The Large session can still run at that 512 selection, though I'm unsure if my recordings are actually what I'm recording, if you will. They seem close, but how can I be sure they are dead on, using the direct monitoring from the SSL+? Very much appreciate your knowledge on this. I have been reading other posts etc., Though I'm still unclear on the matter, admittedly. All

Frustrating stuff. Noticed my recordings were being displaced late weeks back. Did the "ping" test based on suggestions here and got a result of +58 samples. Set delay slider to -58. Now things seemed too early. Did an actual cable out of 1 to input 1 with a transient and got a similar result so I've left the slider as is. Seems my percussion and claps are all just slightly behind the beat though (Monitoring the input from SSL2+ interface and click up nice and loud - it's not my playing that is the issue). Can't seem to find a once and for all answer to getting the audio track laid down to be the actual performance. Could it have something to do with my buffer size always set to 1024 because I'm monitoring from the direct input and don't use the monitoring from the program? Thanks very much.

As well, no plugins are on the channel selected for recording or the Stereo Out.

Recently noticed that audio recordings are being recorded "late". Seems what I'm hearing in the phones (not a return from program, but input monitoring from audio interface - SSL2+) is fine, though after the take when I zoom in I see the actual recorded audio is a few samples late - always. I never monitor from the program, only from the input blend on the interface. I thought I was going a little mad with the performances seeming fine then upon playback seeming different, or just slightly off for lack of a better explanation. Tried reducing buffer i/o to minimum (32) and still the same issue. Tried a test recording midi kick drum on a track, then feeding the soled output back into the interface channel 1 - same late recording. It is a small delay, though any at all I am not okay with. What am I missing here? Or better, what can I begin to look at in terms of adjustment? Using an SSL2+ interface into its own USB port on an older (2012 but modded with new SSD and 16gb ram) MacBook Pro. Thanks all ever so much, Andrew