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Holger Lagerfeldt

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  1. I began on ProTracker on Amiga in 1990, then from 1993 Cubase 3.0 on Atari ST, then finally Logic 3 on Mac from 1998. Pro Tools I've used when I was forced to, and it's certainly capable, but I find it fiddly, grey, and unstable compared to Logic. Ableton Live seems incredibly messy and weird. I get that some people love it for electronic production, but I can't wrap my head around it. Cubase today... the interface is loathsome, but at least it makes sense to me, and the audio engine is slightly better than most, e.g. no zipper noise. If I had to switch today I'd go from Logic Pro to Studio One or the more customizable Reaper, but neither are a step up from Logic Pro. Sooo hard to switch and I instantly miss some essential GUI features from Logic Pro when I try e.g. Reaper.
  2. Actually I've learned the most from people asking me, as sometimes I get a question I didn't (yet) know the answer to.
  3. I think it's fair to wonder about these things as it's not very intuitive exactly why it's happening. Especially once you notice that different analyzers or equalizers report different values for the same input, then it's natural to go "something must be wrong". And occasionally something is wrong, just not very often 😉 The manual completely ignores this and just says the analyzer follows the same peak values/range as the equalizer which is a bit of a stretch in reality.
  4. Cheers! I just added a bit more info related to the windowing functions of some analyzers. I've refrained from the old and annoying "use your ears, not your eyes" - until now, heh heh 😉 But I guess it's applicable.
  5. It's not an error. It's most likely related to the FFT block size of the analysis. It's quite common to see exactly this behavior in various analyzers. Try taking the free Voxengo SPAN analyzer and send a 0 dBFS sine test tone through it. You'll see a similar visual "reduction" at around -5 or -6 dBFS. Same in the excellent and free TDR Prism, all in peak mode. Increase the block size in SPAN and the sine peak will begin to rise visually towards 0 dBFS, but you'll also notice how choppy the update rate becomes if you switch to e.g. white noise or a dynamic signal. Adjusting the block size isn't an option in most analyzers as they'll prioritize a smooth and useful display over a detailed but choppy one. I'm guessing the built-in analyzers in the Channel EQ, Match EQ, and Linear Phase EQ have differing processing tradeoffs, since their prime objective is EQ - not graphical analysis. Also, it depends on what type of windowing function (applied to the bin width/block size of the FFT analysis), such as Hann, Hamming, Blackman, etc. In some analyzers you can set this as well. The MultiMeter is specifically focused on analysis, but its tradeoff is a much coarser frequency resolution (notice now the sine should be a thin line but is displayed as a triangle), and it has no other processing to perform at the same time. Or it's simply offset behind the scenes or uses peak interpolation to fix it. Swings and roundabouts.
  6. Jaikoz by JThink is the most flexible tagger I know of. It can be used for professional work, embed anything from art to ISRC codes, and is pretty good for batch work. The amount of parameters you can add or edit is mind boggling, but it can be as simple as you like, too. GUI looks like early 2000s, though 😉 It's not expensive, but updates aren't free after the first year. https://www.jthink.net/jaikoz/ Skip the "Pro" version and go with the regular Jaikoz unless you need acoustic meta data, that's basically the only difference.
  7. Ah, so the inputs and outputs have different options in terms of gain. Inputs go from 0 to +20 dB of digital gain and outputs go from -100 to 0 dB of DAC trim. Yes, in this case you'd need to first make sure you're using the right type of cables (2 x TRS for balanced audio > 2 x balanced TRS or XLR and vice versa) and then match the I/O gains until e.g. -18 dBFS nominally is still -18 dBFS (or as close as possible) in LPX. Also try outputting a 0 dBFS test signal and re-do the loopback test (without the outboard) after completing this calibration, listening out for any harmonic saturation/overload/distortion due to digital overloading. It could be plug-in delay compensation doing its thing. Try recreating this in a new default (blank) Logic Pro project without any plug-ins open at all, except the test oscillator and the I/O plug-in.
  8. You're welcome! Which channels on your Motu sound card are you using for this purpose, i.e. the I/O of signal from Logic to your outboard? It looks like input channels 1 & 2 also have a preamp and phantom power option, which is currently activated on channel 1. I assume you're using output channels 1 & 2 for monitoring to your speakers. Assuming you're therefore using input channels 3 & 4 for the return from your outboard and output channels 3 & 4 to send to your outboard, I suggest setting all of these channels to identical values, such as "0 dB". I.e. input channels 3 & 4: "0 dB" and output channels 3 & 4: "0 dB". This should take care of any offset you're experiencing. Motu doesn't label the values (naughty, naughty!), so it's not clear whether this is simply a gain offset or something else (perhaps it's stated in the manual)* It's a simply digital gain, it turns out. I suggest setting it to 0 dB and using balanced TRS cables. You need to make sure you don't overload your analog gear. However, in some cases you actually migh want to drive your gear a bit harder in order to intentionally saturate/drive the transformers or tubes. If you decide to go higher than "0 dB" on the line I/Os then make sure all of the relevant I/Os have the same values - but be aware that you risk overloading digitally (=nasty) if you're already close to full scale (0 dBFS in Logic Pro), since these line trims aren't analog on the Motu. Please post a screenshot of the output settings in the card as well, just in case. Also check the "LINE [...] MIX" pages to make sure no weird offsets are happening there! Faders should normally be set to unity (0) or not at all, depending on the design. *EDIT: I read the Motu manual now, it says: "All analog inputs and outputs can be calibrated to support a variety of standards, including EBU-R68, SMPTE RP-155, +4dBu, -10dBv, 2vRMS and 1vRMS. The line inputs are equipped with +1 to +20 dB of digital gain, adjustable in 1 dB steps."
  9. A) First you need to check your Master fader, which is the master VCA/offset fader in Logic. This is not your Stereo Output fader. By default its placed in the right side of the transport bar in the upper part of the screen and as the last fader to the right in the mixer. Make sure the Stereo Output and the Master fader are both set to unity (0) and that no plug-ins on the Stereo Output reduces the output level. Logic will always output without any level change if both of these faders are set to unity. But any offset value on the Master fader will affect the actual output of the Stereo Output in either direction. B) Your sound card has an analog calibration or trim level, both for the output (DAC) and input (ADC). These can be set to a variety of levels, such as +24, +22, +20 or +18dBu (typically balanced/professional) or +6, +4, +2, or +0dBV (consumer/unbalanced). These are either set manually in the sound card's control software or physically with jumpers inside the sound card - or perhaps it switches with auto-sensing. In some cases it's not user selectable, but then the input and the output will be nominally the same. If your sound card has a -4 dB drop on the output activated, which is quite close to the -3.8 dB you mention, that could be the explanation. You wrote: C) "To test I am using a test oscillator plug in, set to output -18db. I have a meter reading 0DB" I assume you mean a test oscillator outputting -18dBFS. If a meter (meter on the oscillator channel fader? meter on the Stereo Output?) reads 0 dBFS (I assume you meant) then you've got an +18 dB boost going on, if we're talking dBFS. Please explain your chain in details? D) If all else fails, make a brand new empty Logic Pro project and repeat these steps to test what's going on. Make sure you send the signal out (DAC) and back in (ADC) without any hardware in the chain. This is a DAC-ADC loopback test of your sound card. Depending on the quality of your sound card you shouldn't see more than a ±0.2 dB change in peak level with this test and the correct settings in Logic Pro. E) Also check your sound card's internal mixer software, if it has one. This can also cause such offsets. Make sure all faders are set to unity (0).
  10. I generally zoom in a couple of different ways: Ctrl key + mouse lasso the area I want to zoom in on. You can do this multiple times in a row, going further and further in. At each step Logic stores the zoom spot/level, so you can zoom out in steps by Ctrl+clicking in an empty spot. Unless you move away or zoom further in or out again, whereever you are is "locked" and stored as a navigation snapshot. If it's a specific region or selection of regions, I select those regions (by lassoing them with the mouse, usually) and press the key command for "Zoom to fit Selection Vertically and Horizontally, store Navigation Snapshot". This key command can also be used in conjunction with Command+A (select all) for a quick overview of the whole arrangement. There's also a dedicated key command for this combination: "Zoom to fit all Contents, store Navigation Snapshot". Alternatively, there's the smart "Toggle Zoom to fit Selection or All Contents", which basically combines the two on a toggle basis. I believe it's placed by default on the Z key. Occasionally I use the regular horizontal and vertical zoom in/out key commands, but as a power user this is a bit slower. The slowest is trying to use the mouse on the sliders.
  11. Thanks! The solution in that thread seems to be consolidating audio? In my case there is just a single audio file imported into Logic Pro and keys don't work, so I don't think it's the (exact) same problem. However, the next time I experience this bug I'll see if simply selecting consolidate audio will make Logic snap out of it. Since bouncing and adding to project audio seemed to fix it today and consolidate files seemed to fix things in the other thread, I have a feeling that the bug is related to something with files or project audio. I.e. a certain kind of file interaction may force an update to project audio... Why Logic doesn't accept keystrokes is beyond me as there's no apparent connection there, so perhaps more is going on. In any case while it's a good thing to save your project before starting in earnest this bug can happen in an existing, already saved project.
  12. On occasion, though quite rarely, I'm unable to save a project or input any keystrokes in Logic Pro. It could be in an existing project or - worse - a new, yet unsaved project. The mouse works and the GUI can be used, i.e. I can press the play button in the transport, but e.g. the spacebar won't work. File dialogs or other dialogs work if I can activate them (apart from the save function which is blanked out) but text input boxes don't work and Logic won't accept any keystrokes. I've got no other software running in the background, nothing that should interfere with the keyboard. The combination of not being able to save and Logic not accepting keyboard commands is an odd one. Sometimes Logic will snap out of this catatonic state if force it to bounce to the desktop (and perhaps other things will work, too), but it's function related, not related to time or waiting it out. I'm not sure if this is a Logic Pro only problem or more of a macOS problem, as I seem to recall this kind of behavior once or twice in other applications. Since I use LPX up to eight hours a day, it makes sense I experience it mainly in LPX. Mac mini M2 24 GB RAM, Ventura 13.6.3, Logic Pro 10.8.1, US language, Danish region
  13. If you have Izotope RX you could batch automate this perhaps, that's what I would try. My OCD requires me to say that dithering has been turned off in your screenshot. Is that because it's your second pass on an already 16 bit dithered file?
  14. The usual way, unless it's done inside a single plug-in compressor, is: 1. Find an available Aux track and insert a compressor For parallel comp settings, check out the Studio VCA https://www.logicprohelp.com/articles/logic-pros-compressor-circuit-types-r17/ However, don't change the mix/wet/dry mode - keep it 100% wet. 2. Send from your source track, e.g. the vocal or vocal bus, to the parallel Aux. 3. Set the send amount to unity (0.0) 4. Click & hold on the Bus send and change the send type from the default "Post pan" to "Pre fader". You'll notice the send knob and the Bus label switch sides on the channel. Sending pre-fader means the parallel compressor input level won't change even if your source channel fader changes. 5. Compress and set the Aux fader level as desired to control the blend amount. You can equalize the return (or send, by inserting the equalizer before the compressor on the Aux) signal to manipulate the parallel signal more. Consider using the linear phase EQ for any equalization when parallel processing. Distortion and limiting can work well on the parallel Aux, too. As CwC says, the above method allows you a higher degree of freedom than a simple inserted compressor with internal mix/wet/dry.
  15. I'd swap the linear phase EQs for regular minimum phase ones. Linear phase/FIR tend to smear the sound with pre-filter echo. Rarely do they make sense unless you have a particular use for linear phase, such as parallel processing or processing on an already limited file. Perhaps the Multipressor should go before the full frequency comp(s) as multiband can work well in catching dynamic frequency issues, before hitting the broadband compressor. Also, I'd get rid of the additional normal limiter in front of the AdLimit, no reason to have two limiters unless e.g. the first is closer to clipping and the second is softer.
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