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Post by c0rtland on Oct 10, 2018 16:37:17 GMT -6
Hey guys. I'm running latest logic update. I have two symphony mk1 chassis 40x40. I'm sending 1k out 23 to a mult Back in 39/40 I'm getting around 15 samples worth of phase shifting (latency?) on the return between the two channels. I checked different outs and ins on the apogee's I also tried going in through my Burl b2. Could logic be doing this? Has anyone heard of this happening? Did I get slipped acid? |
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Post by popmann on Oct 10, 2018 18:45:17 GMT -6
It won't do it sample accurately, IME, no.
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Post by c0rtland on Oct 10, 2018 18:56:01 GMT -6
What do you mean 'it'? Logic? That is so f'd. So if I send a mix out 23/24 when it comes back in the l/r channels are 15 samples misaligned? That's bonkers. There is no way that is an acceptable margin of error.
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Post by wiz on Oct 10, 2018 19:10:18 GMT -6
I send stuff out and back in of logic all the time, sample accurate.
Whats your routing? Within Logic.
cheers
Wiz
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Post by the other mark williams on Oct 10, 2018 19:24:22 GMT -6
I send stuff out and back in of logic all the time, sample accurate. Whats your routing? Within Logic. cheers Wiz Same here. |
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Post by c0rtland on Oct 10, 2018 20:04:58 GMT -6
New empty project.
Track 1 / Native Tone generator plug -> 23/24
Track 2 / input spdif 1/2 (b2 bomber)
Also tried
Track 1 / tone gen -> 23 into mult
Mult into track 2 input spdif 1/2 bomber
Also tried
Inputs into two mono tracks instead of stereo track.
Also tried
two inputs in apogee 3/4 instead of bomber.
I came into this when we were aligning our atr 102. We saw it on the o scope and then found it in logic as well.
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Post by Tbone81 on Oct 10, 2018 21:16:07 GMT -6
How does logic handle delay compensation? Is there something activated on one channel that may be missing from the other?
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Post by wiz on Oct 10, 2018 21:44:00 GMT -6
New empty project. Track 1 / Native Tone generator plug -> 23/24 Track 2 / input spdif 1/2 (b2 bomber) Also tried Track 1 / tone gen -> 23 into mult Mult into track 2 input spdif 1/2 bomber Also tried Inputs into two mono tracks instead of stereo track. Also tried two inputs in apogee 3/4 instead of bomber. I came into this when we were aligning our atr 102. We saw it on the o scope and then found it in logic as well. Try this on a stereo track , instantiate the IO Plug in, set it to output 23/24 and the input to SPDIF 1/2 and ping the plug in, this will give you your latency offset and compensate for you. Now, set the output of this track to a Bus, say Bus1, send the output of bus 1 nowhere, Create Track 2, and set its input to Bus 1, and record away... it will all be sample accurate. You can't just use the return inputs... you have to send them via a track to a bus, for it to be sample accurate. If you still have problems, I will make up a project for you with the routing done and put it up for you to download cheers Wiz |
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Post by drbill on Oct 10, 2018 21:51:52 GMT -6
You might be "sample" accurate, but there are almost certainly inter-sample latencies depending on the hardware. Higher sample rates are better, but it's still there. 1 sample is enough to wreck phase coherancy.
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Post by popmann on Oct 11, 2018 0:05:23 GMT -6
In rereading...do you mean you have a stereo track sent out and back in that the NEW stereo file is 15 samples off the first, or that the L and R channel of the new files are 15samples off EACH other?
My answer was regarding the first. If it's the second and the L&R channels of the return are messed up relative to EACH OTHER...there's something seriously wrong. But, 15 samples on a loopback is functionally better than my LogicX ever got...on the same hardware that Mixbus and Cubase were spot on. Obviously YMMV.
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Post by c0rtland on Oct 11, 2018 5:45:51 GMT -6
In rereading...do you mean you have a stereo track sent out and back in that the NEW stereo file is 15 samples off the first, or that the L and R channel of the new files are 15samples off EACH other? My answer was regarding the first. If it's the second and the L&R channels of the return are messed up relative to EACH OTHER...there's something seriously wrong. But, 15 samples on a loopback is functionally better than my LogicX ever got...on the same hardware that Mixbus and Cubase were spot on. Obviously YMMV. Relative to each other. That's what I'm saying. So bizarre |
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Post by jeremygillespie on Oct 11, 2018 9:40:59 GMT -6
Have you tried sending out 2 mono signals with a signal generator inserted on each channel?
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Post by drbill on Oct 11, 2018 10:04:28 GMT -6
In rereading...do you mean you have a stereo track sent out and back in that the NEW stereo file is 15 samples off the first, or that the L and R channel of the new files are 15samples off EACH other? My answer was regarding the first. If it's the second and the L&R channels of the return are messed up relative to EACH OTHER...there's something seriously wrong. But, 15 samples on a loopback is functionally better than my LogicX ever got...on the same hardware that Mixbus and Cubase were spot on. Obviously YMMV. Relative to each other. That's what I'm saying. So bizarre YIKES! I misunderstood too. Are you splitting that stereo signal between two interfaces? That would be my first thought.... But if they are on the same interface, that's definitely strange. |
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Post by c0rtland on Oct 11, 2018 10:27:55 GMT -6
Relative to each other. That's what I'm saying. So bizarre YIKES! I misunderstood too. Are you splitting that stereo signal between two interfaces? That would be my first thought.... But if they are on the same interface, that's definitely strange. Same interface. I'm gonna have to do some more troubleshooting. |
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Post by wiz on Oct 11, 2018 15:31:25 GMT -6
Yeah, something weird going on there...
I misunderstood as well
Cheers
Wiz
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Post by svart on Oct 12, 2018 8:44:26 GMT -6
15 samples is a lot more than mismatched cable lengths or analog prop-delay can account for. It's got to be some kind of software issue, like a bypassed plugin on one channel or something.
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Post by popmann on Oct 12, 2018 9:53:36 GMT -6
I will throw out the tidbit that I learned the hard way about logic....if it's set to full compensation, you can't have a DISABLED plug in anywhere on a bus. On a track, is fine....but, on a bus, it will basically compensate for the disabled plug in that doesn't NEED the latency since you know--it's disabled, thus not processing audio.
I'm not sure HOW or why that might throw this off--just thought I'd interject because I used it for a year before I FOUND that....and I'm a manual reader.
Do you have another DAW at all? That would be my first step....same hardware/drivers/IO/ApogeeHardwareMixer....different DAW....if stereo loop works, it's Logic. If it's Logic, report back....you'll need to check some stuff in the Environment...but, if you're not familiar, don't go digging in there.
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