Stereo expansion is typically static; in other words, it doesn’t respond to the incoming signal or change its setting - this results in a somewhat predictable stereo image. If you use dynamic or program-dependent stereo imaging, you can make your stereo width respond to the incoming signal, making it interesting.
One way to do this is to use a dynamic equalizer that’s also capable of mid-side processing.
I’ll create a band then assign it to the side image. Then, I’ll make this band dynamic. As a result, whenever the band is triggered, it’ll amplify that section of the side image - making for unique sounding stereo expansion.
Sometimes mastering engineers struggle with differentiating one chorus or section of the song from another section - this can be a frustrating problem for sure. If you want your chorus, bridge, breakdown, or other section of the song to stick out, simply automate that section's processing.
For example, if I wanted the last chorus to be slightly louder than the other choruses, I could go into my automation, find my limiter, select the gain function, and then increase the gain using automation just on that section.
I could do the same thing with my saturation, or maybe increase the high-frequency amplitude using an automated shelf.
You really open up a lot of possibilities in mastering when you use automation.
We’ve all most likely used a stereo imaging plugin at one point or another, but one of the first stereo imagers to exist, aside from panning, was a phenomenon called crosstalk. Crosstalk occurs when the stereo tracks of a tape recorder overlap each other slightly too significantly.
By having a little bit of the left channel in the right, and the right in the left, you cause mild phase cancellation which often expands the stereo image.
This shouldn't be confused with delay-based stereo imaging, which delays the signal to cause this phase cancellation, and should be avoided more often than not.
So, if you want classic-sounding stereo expansion, find a tape machine plugin with a crosstalk function.
When mastering, you can combine multiple forms of saturation to create a fuller and more impressive-sounding master. By combining saturation types you include multiple harmonic formations, which add together to amplify hidden parts of the signal - resulting in an impressively detailed and nuanced sound.
For example, if I combine tube and tape saturation, I’ll get a second-order harmonic from the tube, and a 3rd order harmonic from the tape - as well as other harmonics. Some harmonics may be identical as well, which just means they’ll add to one another.
Lastly, different forms of saturation employ different, nonlinear, compression curves. By combining these curves together you can get an even more unique nonlinear form of compression.
If you want to create as loud of a master as you possibly can without incurring distortion, set the release time of your limiter to exactly 50ms. By setting your limiter to 50ms, you retain transient detail and allow the signal to return to unity quickly after attenuation.
Shorter release times do this as well but also introduce distortion.
The reason being, each waveform has a particular wavelength - with lower frequencies having a longer wavelength and higher frequencies having a shorter wavelength.
When you set your release super short, you cut into low-frequency wavelengths, which causes distortion.
50ms is the absolute shortest release you can have without this distortion occurring.
Maximization and Limiting often refer to the same thing - essentially there’s a ceiling, and you push the entirety of the signal into this ceiling. But upward maximization is the process of taking only the quietest parts of the signal and pushing it upward while keeping the louder parts the same.
If you combine maximization and limiting, you getting the best of both situations.
First, push lower-level details forward with upward maximization. You can also use low-level compression which is a similar effect.
are all good options for this.
Then follow it with your favorite limiter.
If you’re mastering music for other people or clients, it helps to be able to handle a revision quickly and effectively - this is why I like to use additive EQ near the end of my chain. This allows me to make changes quickly without changing a lot about the signal.
For example, say a client wanted the vocal to cut through a little more in their master. If the EQ was at the beginning of the chain, then boosting 2kHz would change how the compression worked, how the saturation worked, and so on.
But, if I have it near the end of my chain, I can make this change without working how it’s going to affect other forms of processing.
