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You're welcome Jay. I've been left scratching my head about that one many a time too. Glad it's working now. Tom

It sounds like your modulation may be resetting to zero. Try recording some modulation before your strings come in and see if that fixes the issue. Tom

Have you tried setting multithreading to Playback Only, in audio preferences? I've found the new Live mode to cause the sort of issue you describe in some instances. Tom

Another approach is to use Logic's EQ in Side mode to cut the frequencies where the loss in mono is most noticeable (probably in the lows to low mids). Experimenting with a high pass at a 6 -12 dB slope would be a quick starting point (but a shelf or bell will be more targeted). Doing this has the effect of monoing the lower frequencies, while retaining the spacial information in the highs, resulting in a less dramatic difference between when listening in stereo or mono. (Tip: Placing a Gain plugin after the EQ and flipping either the L or R channel, will allow you to monitor just the side signal, to hear exactly how it is being affected by the side EQ. Of course you will need to bypass the Gain plugin once you are done) If you feel the overall piano sound is now suffering as a result of the EQ-ing, try using another EQ (in Stereo mode) to boost the areas that are lacking. A similar method to all of the above is to use an EQ in Mid mode and, while listening in mono, EQ the piano until it sounds less different to when listening in stereo. With a bit of back and forth using either method, you should end up with a piano that satisfies in both stereo and mono. Hope that helps, Tom

I don't use Flex Pitch much, but have encountered the problem you describe. From memory, I found that switching from Flexpitch to one of the other Flex modes (in the track header) would allow me to add or edit flex markers. When switching modes back to Flexpitch, these markers seemed to be retained (although not visible), allowing editing of timing with Flexpitch (as described by David), without the problems that you have described. Hope that works for you. Tom

Great news and well done! Hopefully she'll hold together from here.

... I guess you could say it's a sickness ...

Hi RedBaron, Glad we've got somewhere at least. Without a signal flow diagram though, I'm not 100% clear on what you are doing, so sorry if I'm missing what you're getting at, but are you saying that your signal is making two roundtrip journeys when you record it? If so, that's unusual (but I'm sure you have your reasons) and problematic, because Logic will only compensate for one roundtrip. In any case, instead of feeding your external processors directly from the Stereo Out (after the signal has left Logic), have you tried feeding them from an IO plugin inserted on the Stereo Out channel (or the 2Bus aux from method 5), via a spare pair of physical ins and outs (before the signal has left Logic)? As long as you remember to Ping from the IO plugin, so that Logic knows the latency of this external path, you should be good to go (with any of the realtime methods). As Ski said, the idea is to use the Recording Delay parameter to sync a loopback recording that is made through your regular IO path (with no plug-ins inserted in Logic). So if your regular IO path includes, for example, external processors and/or a mixing desk, then include those in the test. If the external processors are only inserted sometimes, then don't include them. Once this baseline is set, you should be able to leave the Recording Delay alone and use Logic's PDC, in combination with the IO plugin, to sync things up when adding in the external processors. When you get a chance, why don't you double-check that your baseline setting for the Recording Delay preference is correct for your system and then retry the realtime methods I've outlined. If things still aren't lining up for you, post a basic, but detailed signal flow diagram for your setup (labelled rectangles and arrows would be perfect) and a clear description of what you are trying to achieve when things don't line up, and when I get a chance, I (or someone else) might be able to help figure it out some more. ... my head hurts now ... Tom

You're welcome Eric! Curious indeed

Glad you found a solution to this. It is a strange one, but I think this may help explain what's going on: When recording from a physical input, Logic automatically compensates the position of the recording for the systems IO latency. When recording from a bus, Logic assumes that the routing is internal and that there is therefore no IO latency - so Logic doesn't compensate for this. My guess is, that Logic can handle either recording scenario in isolation, but not both at once. When trying to record simultaneously from a bus and a physical input, perhaps Logic defaults to its input recording behaviour and compensates both recordings for IO latency. This would have the effect of putting the input recording in the right place, but put the bus recording early by the IO latency amount and before the start of the recorded region. If this is true, then in your original test, changing the IO Buffer size should affect how early the bus audio is recorded; and your Input>Bus>Audio Track solution should result in both files being recorded in sync with each other, but early on the grid (by the system's IO latency). From what you and Eric have posted though, that doesn't seem to be the case, so there could be more to it than this. Anyway, it's probably worth double-checking your solution if maintaining sync with the grid is important in your particular situation. Tom

... Also, just to clarify the whole "what gets delayed where" business of Plugin Delay Compensation as it relates to bus/auxes and outputs: In All mode, differences in latency are reckoned and compensated for whenever two audio streams are summed. For example, if two auxes feed a common bus/aux, then any difference in latency between the source auxes will be compensated for by delaying the output of the least latent aux by the appropriate amount - so that both audio streams are in sync when they reach the summing bus/aux. This will happen whether or not the summing bus/aux is routed to an output, another bus/aux or even to nowhere. So the key to when PDC is applied to an audio stream isn't whether or not it is eventually routed to an output, but whether or not it is being summed with a more latent signal. On top of this, the stereo output channel strips (in a multi output system) are also appropriately delayed to put the parallel audio streams that they carry in sync with each other. (And as mentioned in earlier posts audio and software instrument channel strips are latency compensated by negatively delaying the regions that feed them, so that each outputs its audio signal at the same time). Hope that helps, Tom

Hi RedBaron, As long as Plugin Delay Compensation is set to All (and the Recording Delay is set to offset the regular system roundtrip latency), the following methods all result in a perfectly placed loopback recording or bounce on my system, so hopefully at least some will work on your system too: Method 1 - Disable Software Monitoring - Record the loopback Method 2 - Leave Software Monitoring on, but enable Low Latency Mode - In the Arrange Window, select your record track and make sure that no MIDI tracks are inadvertently record-enabled - Record the loopback Method 3 - Realtime bounce (good if using external FX) Method 4 - Offline bounce (no good if you are using external FX) Method 5 - Set the output of all tracks and auxes currently feeding the Stereo Output to a spare bus/aux - Name the automatically created aux "2Bus" or something similar - Move any plugins (e.g. AdLimiter, etc) from the Stereo Output to "2Bus" - Create a send to a spare bus on "2Bus" and set it to 0dB - Delete the aux that was automatically created by the send - Create a new stereo audio track in the Arrange Window and set its input to the bus you are sending to from "2Bus" - Mute the new audio track (or remove its output assignment) and put it in record - If you like this method, then rename the bus that "2Bus" is feeding to PRINT (or similar) in the mixer under Options>IO Labels. This makes it easier to follow the routing and allows you to quickly print anything in your session by sending it to the "PRINT bus". I've tested these methods with five AdLimiters on the output, with each set to a lookahead of 200ms - so that's 1 second of latency - as well as using them in real world situations, so I hope they give you some joy too. Tom

You're welcome - glad you didn't do any damage to your table! I had a feeling this, along with some other transform bugs, was fixed in v9, but perhaps I'm mistaken. Tom

Very glad to hear it! Tom

Hi LagartijaNick, Glad to hear you're liking the tutorial! It looks like you are following the method correctly, so I think you are running up against an unfortunate bug that affects the Delay parameter of External MIDI Instruments when set in ms. I demonstrate the bug, and a workaround, in video 48, Bug Shop - Part 3, so take a look at that. In a nutshell though, when set in ms, it appears that an External MIDI Instrument's Delay parameter will override the delay of all External MIDI Instruments positioned lower in the Arrange Page that have a lower delay setting. So if you try putting the D4's External MIDI track below the AKAI's, I think you will find that things behave as expected. Most likely though, you'll want the freedom to rearrange the order of your tracks any way you like, so a better solution is to keep your delays set in ms in your template as place holders and switch them to display in ticks once you've settled on a project tempo. However, the best solution (for me anyway) is to do what I demonstrate in video 49, which is to scrap the Delay parameter altogether and put a Sample Delay (set to 124 samples in your case) on the input channel that's bringing the D4 into Logic. The only downside to this solution is that you'll get an extra 2.8ms of latency when playing live through the D4 - but that shouldn't be much of an issue. Hope that sorts things out for you (and if you are experiencing this bug, report it to Apple at http://www.apple.com/feedback/logicpro.html). Tom