We know that the new(ish) circuit models for Ableton Live’s Auto Filter effect add some tasty analog-modeled distortion, but what about their characters? Which of the circuit models adds the most aggressive distortion?
If it takes forever to scan plugins every time you open Live (due to certain poorly-coded plugins taking longer than others and bottlenecking the process), you can set it up to skip plugin scan. Keep in mind that if you do this, you’ll want to perform a manual scan (from preferences) any time you install or update any new VST plugins.
How to do it? Add “-NoVstStartupScan” to your options.txt file
That’s it! Live will no longer do a scan every time it opens.
Ever since i applied this option, the stupidly vast size of my plugin collection doesn’t adversely affect my workflow. Live startup time is way snappier. When i occasionally buy or grab a free plugin, i just run a scan after installing it. Every once in a while you might have to do a “deep” scan (hold alt while clicking the scan button) if something doesn’t show up when it’s supposed to.
Note that from Live 10.1 onward, the plugin rescan button is located in the new dedicated plug-ins tab in Live’s preferences, rather than under file/folder, as previously.
Hopefully this nifty tip is inapplicable to you, as that means your plugin scan is not bloated. However, for those of you that are having issues with slow startup of Ableton Live (or in case it eventually becomes an issue) this may be a viable solution.
I found some old notes i had taken (for my own reference) about the parameters of Ableton’s Gate audio effect plugin and figured others could use the info as well, so i’ve polished ’em up, expanded on them a bunch, and dropped them here for you.
I’ve found that getting to know exactly how each parameter works with gate plugins leads to attaining desired results exponentially easier, and in particular learning how the more exotic parameters such as lookahead and hysteresis work facilitates sophisticated gate optimization.
There are not many controls for digital plugins where i prefer stepped options as opposed to continuous values, but the Q factor of EQ slopes is one of those that generally i do. Not really sure why i prefer them that way; i suppose there are a few reasons.
Regardless, i made myself (and you!) up a nifty spreadsheet of “Go-To Q Values”, based on logarithmic steps between the basic Q value(0.70607) and the minimum or maximum ranges. These Q values are optimized for Ableton Live’s EQ Eight, but can be applied to most any parametric EQ, when you want no-brainer go-to Q values to fall back on.
Just disable VST plugins during times in tracks when they are not being used.
Automate the nifty device activator (aka “on/off switch“) of a device (or a rack of devices) to shut off during times of silence. Set the automation to switch on a little bit before audio starts, and to turn back off a bit after the audio is completely silent again.
So there’s this issue with fading between chains in a rack in Ableton Live which i haven’t heard much mention of. Perhaps you’ve noticed it?
Say you have a rack with two chains, let’s call them “A” and “B“.
Starting to Set Up our Chains
Using the Zone Editors in the Chain Selector Editor of the rack, we’ll place A on the left (full at 0), fading out the other direction. Vice versa for B, placing it on the right (full at 127) and fading out towards the left.
A … fades to B
Now let’s map the Chain Select Ruler to the first Macro.
Macro Control to Fade between A and B
Macro 1 will now give us a nifty fader knob between whatever processing we place on chain A versus whatever we place on chain B.
Kind of like every human being has a height value, every audio clip has a peak level value. So what can we do with that information, beyond knowing that going above 0dB usually isn’t advised? In our eternal quest for ultimate audio quality, the inclination can be inherent to record loudly—as close to 0dB—as possible, and thereafter maintain that peak level. We fear that by mixing with track levels that are too quiet, we might be losing fidelity, some harmonic detail in the saturation floor or something.
While it is true that recording analog signals as loudly as possible without signal clipping to begin with will indeed minimize noise floor, the benefits reaped by maximizing peak level for individual tracks tapers off as you get closer to zero. At what peak level are audio sources louder than they really need to be? At what level are they too quiet, that they might need to be boosted excessively later? Using an algorithm based on the energy of how sounds stack together, i devised a set of go-to ranges for peak levels based on track counts. Since peak levels can vary fairly wildly based on content, you are given minimums and maximums (instead of single target values). Juicy details ahead…Continue reading →