You may have heard many varying descriptions of the difference between track compression and bus compression, usually including vaguely-defined, mysterious terms like “punch” and “glue” which don’t really help us understand anything.
Well, i have a kinda dorky yet effective way to think of the difference between track and bus (aka buss) compression for you. Continue reading →
So you’re working on a song and you’ve found a frequency which you want to adjust. Let’s say you want to nudge down 4.4k a little bit to reduce a bit of harshness. Now you ask yourself: what type of EQ should be used?
As time passes by, there becomes more and more dynamic EQ plugins available. Besides, in Ableton, Bitwig, and other DAWs, it’s easy to make any automatable EQ plugin act as dynamic by the use of envelope followers. On the other hand, when using a dynamic EQ plugin, there may be times when you want to use the bands as typical, with no reactivity. So, we have a pretty much open choice of whether to use a static or dynamic EQ on a track we’re working on. So what to do?
For this tutorial we’re going to posit an example scenario: a way to achieve the common practice of narrowing the bass content of a stereo track by scooping out the S channel’s low end—but this time, using analog gear instead of plugins. But what if we don’t have any M/S gear? Not to worry.
True Peak Limiting is a method by which a limiter adjusts for how the digital waveform will be reconstructed by playback systems which can result in actual peak levels above 0dB even when the digital peak level is technically shown at below 0dB.
Basically the way intersample peaks occur is that the quantization points of a digital waveform can at times imply a curve between them which goes up a little bit higher than those actual sample points, resulting in a louder-than-expected actual peak level coming though.
To imagine it, just draw two dots in your mind, and then instead of drawing a straight line between them, draw a slightly curved line. The two dots are at a height of zero, and the curved line connecting them is bumping a little bit above zero. See?
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.
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.
There’s some ambiguity between the terms gain, level, and trim. In general, they are used interchangeably. But sometimes they are different; for example, guitar amps often have both gain and level controls.
I won’t attempt to provide the ultimate absolute definitive official definitions of those terms. I’ll just tell you how i tend to use them. Continue reading →
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 →
Having recently acquired a new custom-built PC, various steps implemented to optimize its performance and usability were taken note of. While not an expert in such things, i figured to share these tips in case they may prove useful. Keep in mind that this guide is presuming that you already dabble in audio production and are upgrading to a new—from an existing—system.