Eddy

Hello, I haven't worked with one of my hardware MIDI keyboards in a long time, but recently tried to do so -- when I did, a strange thing happened: I recorded in a part, sounded fine (no latency or anything), and also played back fine, in time with the track in real time etc. Again no latency issues. But when I clicked on the region I recorded, it looked like I played everything behind the beat -- almost a 16th note late. But again when playing the region with it looking like that, the track sounded fine/aligned/on the beat etc. If I pressed Quantize to line up my notes with the grid, then things sounded off (ahead of the beat now). I do have several third party plugins active on other tracks, but this is just an External MIDI Instrument track I'm working with at this point; and just FYI all VST/third-party software instruments don't have this issue (i.e. they look and sound aligned, or at least as played where I can quanitze/line things up etc. if I want to). I'm attaching a screenshot -- you'll see that the MIDI region notes are not lined up to the grid, even though I played as exactly as I could, and the resulting audio file bounce/export that I did, is indeed lined up and sounds fine, you'll see the start time of the WAV file is ahead of the start time of the MIDI notes.

Thanks for the reply, yes I do use import all the time when working with audio files of various types -- what I'm really trying to do here is basically use Logic as a real-time music player, that I can just route the audio through iTunes then process with whatever plugins I choose (I know programs like Fidelia can do this, just looking at other options using Logic instead). So the issue is with real-time behavior, using input monitoring, rather than importing and playing back (which is of course possible, just means having to import every song that gets played, an extra step I'm trying to avoid)...

Hello, I just came across something that I can't explain, and wondering if anyone here has experienced something similar: my objective is simple, just route iTunes to play through Logic in order to process and/or capture the audio coming from iTunes. Use Soundflower, right? Seems simple enough, I selected Soundflower the output for system sound in System Prefs, selected Soundflower (2ch) as the input device in Logic, and my regular interface as the output. Everything "worked," however when I look at the meters in Logic, it is showing that the audio is clipping (actually just going into the red, but sometimes, as much as a dB or more, usually more like 0.4 to 0.7 dB in the red). I didn't really hear any clipping artifacts, but thought to try actually pressing record and capturing some of the audio, and here's where it's weird: the resulting waveform is normal, no clipping at all! Just like most modern recordings, the audio I was playing does go up to 0.0dB, but no clipping whatsoever... how could this be happening? Even putting a meter like Voxengo's SPAN is showing the clipping numbers when I'm just monitoring, but not after it has been captured and is playing back... Thanks for any thoughts!

Hey redlogic, Thanks for the reply, just now able to get back to this -- yes they're there in the Finder, after I load the rest of the project I can re-load the samples back into Ultrabeat and it will start working again. Seems like a weird bug or something, I've encountered it both on my 2008 MBP and my 2012 Mac Mini, so it's happening on both computers...

Hello, I've searched other threads but I haven't found the specific problem I've been having: I created a track with an Ultrabeat Drag & Drop Samples kit to start with, I have 5 samples loaded into Ultrabeat currently, I saved the setting as filename (whatever) -- if I close the project and open it again, Logic asks me to locate the first sound instead of locating it automatically; however, when I do try to locate it, when I'm to the correct folder the window just looks blank, I don't know what happened to the list of samples, so I go ahead and press "open". It happens one more time, same actions, then goes to the project, however now all of the samples are gone (it didn't even ask me to locate the other samples, they just disappeared). Any idea what's happening and how to fix it? Thanks!

Hello, I just noticed this issue after I purchased BFD Eco and tried playing some MIDI data on a track -- the drum hits play back fine from the MIDI track, but when I go into the Piano Roll Editor to select them, every time I click on a note it sounds way too loud, like going into the red on the meters (for example, a snare hit that was playing back at say -3db, when I click on it to change the velocity etc. it plays back at like +4db, clipping the meters). Is there some kind of MIDI thru issue happening, or some reason the note would get triggered twice? Thanks in advance for any help! Eddy

Just remembered to check on this -- thanks for the reply! I thought there might be a way to actually invert all of the individual waveforms at once, but it looks like Logic would rather have the Gain plugin do it, which is no problem... just wondering if it was possible.

Hello, I've been looking for something about this but haven't found it yet: I know you can use the "invert" function in Sample Editor to flip the phase (invert polarity etc.) of a region that is selected, but apparently it is not possible to select multiple regions and have them all inverted at once... or am I wrong about that? I'm hoping I am, I'm trying to do that with some multi-mic'd instruments that have already been cut up into regions. Thanks in advance for any help with this!

(Sorry, the OP has been detained until now, I'm back ) Wow, thanks gentlemen for all of your responses -- I respect what each of you has posted in prior times on prior topics, so this is great to have a collection of entries from so many. I think I'm understanding what you're saying, and it is indeed true I think that the VU meters in Satson are only there for visual reference to be different from the digital peak meters in Logic (although they do indeed change when you start tweaking other knobs in Satson, like gain, HPF and LPF, the saturation modes, etc.). I think it just confused me for a moment when I tested the tone and it was the exact difference in number from "+6" to go down to "-8," (hence "14") when I had the setting for the meters on 0dbVU=-14dbFS. I know it's not that big of a deal, I'm sure many people don't "need" VU meters when mixing etc. But I will say that they have been very helpful in my mixing workflow so far, they help me put things in the "sweet spot" much faster, I'm mixing with more conservative levels which translate to better mixes and mixes that are easier to master, etc. You're right, Eric, thanks that's the idea, I'm trying to get a sense of the signal in a "virtual console" situation, going into the console, then being patched to compressors, EQs, effects etc. (many of which are emulating hardware themselves, so I realize I'm going into the Matrix here)... Thanks for the reply Lagerfeldt I appreciate your comments! (As I've appreciated your posts in the past, your guides to digital recording and mixing etc.) -- Satson does have a Gain knob as one of its primary functions, its purpose is to go at the beginning of the chain, then on to EQs, Compressors, etc. having a reference of 0VU=-18dbFS (or -14, etc.). I agree with you, it makes good sense to have things consistent! And there's a definite change in sound if you want it with Satson's architecture, it's supposed to contain some console saturation, which you can push even harder if you want, I don't really do it but some may... otherwise many times it's just leveling out a signal to pass along to the next part of the chain (I'm still learning all of this, but I want to say some of these "analog modeled compressors" react differently to different levels going into them, does this make sense that you might want to adjust levels with Satson (or a simple Gain plugin would work I know) before going into the compressor (whether or not it has an input knob, which many of them actually don't) Maybe the initial question does actually boil down to the simple "floating point" issue where as David said the signal can go beyond 0dbFS in the channel strip and not clip, I need to understand that better, I'll read up some more and it will probably start to make more sense. Anyway sorry for putting in three days of replies into one post, thanks again for all of the replies, and feel free to hijack my post to talk about other audio issues as long as they're entertaining

Hello, I know there have been some other threads about Logic's meters, but I think this topic is a little different: I've been observing what's happening on the meters for Logic's channel strips after using my Sonimus Satson plugin (a console emulation that has a VU meter that's calibrated to 0dbVU=-14dbFS or -18dbFS). If I set Satson to the -14db setting, and then play a steady tone that shows 0VU on the Satson meter, the Logic meter reads -8, not -14 like I thought it would -- does this mean that for channel strips, 0dbFS is actually +6 (in the red) on the meter? Does this have something to do with the "floating point" thing? And then on the master bus, in the red is really "in the red", meaning 0 on the meter is really 0dbFS? At least that's what it looks like, especially when putting a limiter on the master bus. Thanks for any help on clearing up the confusion!

Never mind -- I realized it was one of my plugins that needed to be re-authorized -- problem solved!

Hello, I've been looking to see if this problem has happened elsewhere, but I haven't seen it yet -- just got a new Mac Mini, installed Logic clean and upgraded to 9.1.8 and Mountain Lion 10.8.2; worked out some initial issues, everything is loading and coming up fine now, expect the fist "real" project I tried to play back, there were intermittent audio dropouts/ periods of silence every minute or so, just for a half-second, no more than that. Anyway, I never had that happen in my older 2008 MacBook Pro with Snow Leopard... any ideas? I will say that I'm having to daisy-chain from the FW800 port: first my G-Drive Mini (which is FW800), which is my sample drive; then to my Maxtor FW400 drive (for audio and projects); then my MOTU Ultralite interface is last in the chain. Could that be causing issues? Or is there some buffer setting/journaling on or off/other preference that needs to be checked or unchecked? Thanks for your help!

Thanks Nigel! So close, it seemed like it was going to work, and it actually even changed the volume knob for a small interval of time, then stopped working... I'm still figuring out what all of the controls mean on the modulation menu within Kontakt, but it looks like it should work, I don't know why it's not. The upper left part of the Kontakt screen is showing activity, and the foot pedal is definitely showing CC11 from 0 to 127 in the Activity Monitor in Logic; it seems like either Kontakt is getting conflicting signals or just not processing the expression pedal movements. Still going to work on it...

I guess the other question could be, What is the best way to record volume changes on a VI like Kontakt? I'd rather not do volume fader automation, I'd like to leave that for the mixing stage, this would just be for tracking live, but having the volume changes recorded as MIDI volume data.

Thanks for the replies gentlemen -- it's working exactly as you said, showing up in the Hyper Draw part of the Piano Roll, recorded volume changes on the ES2 patch, etc. What I was trying before was a random patch in Kontakt Komplete, which probably isn't set up to record volume changes with the expression pedal. I guess the solution there is what you were talking about Atlas, putting a transformer object in the environment to change CC#11 to CC#7 -- haven't done anything like that before, is there a post on here somewhere that explains it? I'll also look in the manual... Thanks, Eddy