plaidcurtains

When my projects get crowded, I get this message often, and I'm usually able to work through it. But sometimes it happens every time I press play without fail, no matter how big or small the project is. So sometimes I can't do anything. It doesn't seem to have much to do with the amount of work my computer's doing - I could be working in logic pretty smoothly for hours, then leave my computer for a half hour and when I get back Logic stalls so much i can't use it. I have an old mac, but it's not like the entire computer's frozen - it's just Logic, and it seems to happen randomly. Maybe my projects are too big for the computer I'm using and my disk is just too slow, but the way it seems to happen randomly makes me think it could be something else? Is there anything I can do to mitigate this within Logic? Say I have 10 tracks in my actual mix and another 10 muted"working" or "maybe" tracks - does muting them stop them taking up working memory? If I tidied them away in a folder, would that help? If anyone has any tips to save on working memory, that'd be a great help.

For example, can I program, say, a flanger to turn on only when the pitch shifter is at +5 semitones, and turn off when it's set to any other value? What I actually want to do right now is trigger the reverb to freeze (I'm using Chromaverb for that currently) when the volume reaches a certain level. The next time the volume reaches that level (input gain will peak, dip, peak etc), the reverb should drop the previous freeze and freeze again. I'm aiming to output only the wet signal and use the reverb as a live instrument. This is slightly different to the above example because overall volume is a measurement rather than a setting on one plugin, but I'm interested in both scenarios.

Ahaha thankyou

I often find that I'd like to be able to record subtle changes with plugin. If it sounds good when I, say, turn distortion up and down in real time, I want to keep that effect, and make the gradual change happen on its own. Is there any way to program plugin changes over time on a track? Perhaps it's possible to record plugin changes live? Or to program abrupt plugin changes by choosing a point in time when channel strip/plugin settings change? I can imagine a few different ways this might be possible, just not sure if they exist. I figure there must be a way to make effect plugins fade in and out, or make plugin settings turn up or down without having to change them live?

OK, got it. Thankyou for all your help man, you really helped me to learn some of the basics here!

Ok, I'm p much 100% on the role of the compressor, high pass, loop gain now! So the same compression works differently on signals with different amplitudes - it uniformly cuts them to a certain volume/signal strength/amplitude. This means it's not a gain setting, bc gain is always proportional to the input volume, or adds a set amount of dB? Though it does change the signal strength, so the difference across the compressor, aka the gain (disregarding makeup) must change depending on the input volume. So it's not so much a set gain but a changing one. The gain isn't the constant, deciding variable, but it can be measured. (?) I didn't really understand the distinction between volume and gain so I've been watching yt videos - in one video, the guy says gain is a measurement of signal strength, so it refers to everything that moves through the DAW, and volume is a measure of sound, aka air moved by speakers or headphones. But an input signal does have a volume, or at least an amplitude or signal strength, so I wasn't really happy with this explanation. A different guy also says gain is generally measured by signal strength, e.g. voltage, power or impedance, and the relative difference between the input and output is the gain. According to him, volume is just loudness, and gain can sometimes result in volume. So gain... isn't really a property of the signal itself, but a measurement of the difference between the output volume and input volume. Volume is a property of the signal output. Amplitude is a property of the signal. You could say both amplitude and volume describe "signal strength". But gain is a measurement of the difference between two signal strengths, not a measurement of a signal strength itself. Gain is relative, volume is not. So when people say "gain means input" and "volume means output", that seems wrong, lol. Isn't it more accurate to say gain acts on input and volume measures output? Both the input and output signals have a volume - neither actually has a gain. Gain measures and determines both the change in signal strength and the change in volume. Is this all right? And the reason the faders in the mixer change volume not gain is because there's no "processing" going on there - the signal volume does change, but it's not called gain bc there's no input and output going through a processor? Plus you can change the vol faders any time - they aren't ordered like plugins.

Ah ok, thanks.

Thanks for staying with me on this topic - I mostly get what's going on now! I have some confusion over what you mean by scaling to 100%, but I think I figured it out below as I was writing the question... So this means it affects the gain and not the volume? I'm a little confused about the 100% here. What exactly is this measurement? You referred to 100% feedback earlier, but if the compressor scales to 100% I think it must be a gain value - are you referring to loop gain? But then I'm thinking if the compressor scales to 100%, 100% must be decided by the compressor, but I'm not sure exactly which setting(s) would decide it. And does the compressor scale either up or down? I get that it doesn't boost specific frequencies, and I get that it attenuates down to the threshold level, but I don't know exactly what you mean by scaling to 100%, especially when the makeup is turned off - none of the research I've done has mentioned a function like that. It implies more than just attenuating the signal with threshold and ratio. I would think it would imply boosting up to a certain level (as well as attenuating down to that level) but as far as I know the compressor doesn't do anything like that with the makeup set to 0. When the high pass filter is turned on and the higher frequencies can be heard when they could not before - cutting the lower frequencies out does make the whole thing quieter, so the remaining frequencies get louder on their own (because of the 2.6db send, not because of the compressor) as much as they are allowed to by the compressor settings? (And this is the 100%?) So do you mean the compressor scales only by attenuating, and in this particular situation this always means 100% of the signal that would possibly be allowed by the compressor will get through, because the gain is always getting louder. But this doesn't mean the compressor itself does any scaling up or boosting?

I've dragged some chord diagrams from the part box into my score and assigned chords to them: I double clicked on them and highlighted the little speaker button green, which triggered a chord sound at that moment, but it didn't make anything show up in the piano roll and nothing plays when I play the track, plus when I close and re-open the window the button is grey again. Is there a way to make these trigger midi, or any other way to get chords from the chord library to trigger midi versions? It would be so much quicker than drawing them in in guitar tab form, as I have been doing.

Thankyou for the explanation - so the limiter on the stereo out is ultimately keeping the levels from rising past a certain loudness, and the compression is balancing the levels under that point? And compression keeps the output level the same by boosting? I've been reading some beginner articles about compression and I gather that any transients above the threshold will be attenuated, and it vaguely says the less dynamic areas will be boosted. This could imply it is the compressor that keeps the output level constant - so when there is some gain reduction in certain frequencies, does the compressor always automatically increase the gain of the others to keep the input level the same as the output level? Or is it that only happening in the context of a 100% feedback loop, because the higher frequencies are ever-increasing anyway, so with the HiPass limiter on and the gain of lower frequencies diminished, the higher frequencies make up the level imposed by the stereo out limiter? Hope this makes sense!

I've tried this and it sounds really cool, still trying to figure out exactly what's going on! I used myself singing and playing guitar as the source audio, and I've found a pretty good horror film soundtrack/singing bowls type of sound. There are undertones that sound choral too... it sounds alternately angelic and demonic. I have to thank you for making me sound like a coven of witches. So you route the original track to Bus 1, then from Aux 1 you route it back through itself AND through Bus 2 to Aux 2? And you send it back from Aux 2 thru Bus 1, so I guess that also goes back to Aux 2 through Bus 2 and ultimately is sent from Aux 2 to stereo out. Is this how it works, am I right? I figure there might be something about "bands" that I'm not getting, as I haven't researched those yet. I think I'm sending the input through the Aux 1 plugins 3 times in total, and through the Chromaverb on Aux 2 god knows how many times? Wondering if it loops endlessly, as Aux 1 sends to Aux 2 and Aux 2 sends to Aux 1... Also, the HiPass seems to be the key to literally all the overtones - If I turn it off, they just go away and leave the bass hum. Why? Are they letting through frequencies that wouldn't otherwise have been there, and if so what was disallowing those frequencies? It doesn't seem to be any of the other plugins... it's it's just interesting that a filter would so clearly add sound rather than taking it away.

I haven't forgotten about this post, will get back to you once I've tried this out!

Ploki, that's exactly the case w/r/t the tails. (It's she though lol.) If anyone's interested in the fixes: I had success with bouncing in place, including the audio tail in the region and then fading it out. I also tried adding an Enveloper plugin and turning the release slider down to the lowest value. This is my preferred fix, as not bouncing it seems to keep more of the original quality of the resonance. Adding a compressor and fiddling with the ASDR settings or creating a sample track and changing the release in EXS24 didn't do anything - I think that's because the audio tail wasn't "supposed" to be there, aka it was more of a side effect than a direct action on the signal, if that makes sense. I think only the Enveloper worked because of this: " In contrast to a compressor or expander, Enveloper operates independently of the absolute level of the input signal, but this works only if the Threshold slider is set to the lowest possible value." (Courtesy of apple support.) Not 100% on what EXACTLY this means, but I think I get the gist.

Thanks for the advice everyone - I ended up bouncing the sample in place and have managed to get a pretty seamless repeating effect, using snap to zero and fade to get rid of the clicks. I bounced a looped version and used the second loop as the repeating segment. The first loop wasn't suitable because it started up from silence and I wanted a continuous effect, and the second loop wasn't too affected by the building tail. It sounds ok, slightly different but acceptable. I might try to loop the sample without bouncing and duck reverb. I can't control the length of the tail by tweaking any reverb settings as I don't have any, but I'm hoping I can make the tail shorter than the region with ducking? I will also try the release slider thing and report back. David, I'm pretty sure it's the EQ making the tail - I have 5 different Channel EQ settings turned on, and no other FX. I deliberately tweaked the EQ for extreme distortion (so much that I'm getting a pretty clear note out of what was originally a recording of a clap). I've checked with these on and off, and it definitely seems to be the combined EQ creating an audible tail (visible when bounced with the audio tail in the region). "wail" is the original sample, the track below it contains the bounce.

I've been using the locators to effectively loop my audio while I tweak the sound. I'm happy with how it sounds that way, but when I actually pull the sample out in a loop (it doesn't matter it it's the audio version or the sample track, it does the same thing), the reverb builds up until it becomes a roar. I don't want to get rid of any reverb because I like the way it sounds within the individual sample (besides, I didn't add reverb and I think the "tail" is just from combined Channel EQ), I just don't want the cumulative effect. How can I get rid of the reverb tail when looping, and just have the sound begin and end at the region parameters every time? (Technically x number of loops makes one region but hopefully you know what I mean.)