Production P.E.M.D.A.S- The order of effects

[outdropt]

Some elementry math for you.P.E.M.D.A.S-
■ P = evaluate Parentheses
■ E = evaluate Exponents
■ M = Mutiply
■ D = Divide
■ A = Add
■ S = Subtract

Wikipedia-"In mathematics and computer programming, the order of operations (sometimes called operator precedence) is a rule used to clarify which procedures should be performed first in a given mathematical expression."


This can apply to Music production. Of course there are no "rules" in music, but if you place different effects in a row, this can affect how the sound.... sound's.haha

All effects will change based on the information given in the incoming audio signal, generally the better effects will use multiple sources of information to affect the sound. This can be volume, frequency, time, other audio sources, algorthimes/automation, ect. That is why you will see different results based on placement of said effect the audio sources rack of effects.

Ex. (Note that each effect can produce different results based on the algorithms set in the plug in as well as other user settings incorporated within the plugin's.)

distortion+filter= how many FQ are present based on the cut off of the filter

filter+distortion= the LFO modulates the "color" of the sound, as well as how the distortion effects specific FQ while certain others are present.

chorus+delay= a sound that phases but repeats that same phase modulation over a period of time that is set in the delay

delay+chorus= a sound that repeats at a set time but can phase modulate throughout the repeating sound


Now for something a little more complicated

Seprate, grouped audio channels+(Filter{2K+20K})+chorus+Stereo widener

Seprate, grouped audio channels+(Filter{600-2 K})+heavy distotion+mono

Seprate, grouped audio channels+(Filter{200-600})+heavy saturation+sidechain compressor+mono


Seprate, grouped audio chanels+chorus+Stereo widener+(Filter{2k + })

Seprate, grouped audio chanels+heavy distotion+mono+(Filter{600-2 K})

Seprate, grouped audio chanels+heavy saturation+sidechain compressor+mono+(Filter{200-600})


They both obtain different result.

In the first case, you have one source audio signal being taken and split into 3 identical channels, from their it is being filtered into a specific frequency range and then affected by stated effects.This can result in the overlapping of frequencies. This can be a desired effect, all VST's add their own characteristic to the sound. Some effects like distortion, reverb, saturation, and many others will generate frequencies that are not otherwise present. This can also cause muddiness in your mix so use this wisely.

In the second case you have effects that are using the incoming audio signal and affecting the sound entirely. The signal will then be filtered to a specified range. This is more controlled but filtering between certain effects like compressors, stereo wideners, distortion, ect can cause a noticeable change in volume, as well other characteristics of the sound that might be undesirable

.Again, there is no correct way. Whatever you think sounds good is the right way to go. Id like to know some effect racks you guys build...

[Aphile]

I've always done my splitting and hard cuts first. I figured that processing a select area of frequencies would yield cleaner results than processing all frequencies and trimming the outputs with EQ. This is a topic of high interest to me with very little documented online.

How do you guys do it and WHY?

[swerver]

I've always done my splitting and hard cuts first. I figured that processing a select area of frequencies would yield cleaner results than processing all frequencies and trimming the outputs with EQ. This is a topic of high interest to me with very little documented online.

How do you guys do it and WHY?

This. Processing frequencies that I am then going to EQ/cut away is illogical to my brain - that's not to say it's wrong though, it just makes no sense to me.

[Isley]

EQ first, if you process a sound then cut away from it you waste creative energy that couldve been used in a better way if you had the initial sound eq from the beginning. Coarse - fine, not fine - coarse, etc..

[NinjaEdit]

One order of operations I would suggest, as far as engineering goes, is inspired by good analog mixing desks. This would be done after creative effects/sound design, when it comes time to improve the sound transparently. Creative effects, by their nature, will occur in any order and combination.

Gain
Volume
Stereo image (pan, width, m/s)
Frequency response (filter, EQ, de-ess, de-noise)
Dynamic range (volume automation, gate, expansion, m/b compression, compression)
**Harmonic content (saturation, excitement)
*Maximisation (eg ISP limiting)

* Final maximisation is inspired by the mastering process. It is optional at mixdown, but would occur at the end.
** It is not obvious where harmonic content adjustment should occur, because all analog components will colour the signal.

[Artie_Fufkin]

In general, I usually have a limiter at the end. EQ/filter before the limiter usually so it's not compressing the signal based on something in the low end that I didn't plan on using anyways.

reverb, saturation, and many others will generate frequencies that are not otherwise present.

Does reverb actually do that? I can imagine short, metallic sounding reverbs do this, but otherwise they shouldn't, right?

[MoonUnit]

In general, I usually have a limiter at the end. EQ/filter before the limiter usually so it's not compressing the signal based on something in the low end that I didn't plan on using anyways.

reverb, saturation, and many others will generate frequencies that are not otherwise present.

Does reverb actually do that? I can imagine short, metallic sounding reverbs do this, but otherwise they shouldn't, right?

Yes they do. Think about it, if you send your reverb that's on a synth to it's own separate track, it sounds different from the synth even though it is BASED on the present frequencies of the synth. Reverb is smothering effect... this helps it sound natural even if it is only an algorithm.

---

I've fallen into a habit of EQing after compression and recently realized that it's not always the best idea. In general, EQ after compression is a good idea because the squash and boost of the compressor will reintroduce some frequencies that you have spent time EQing. EQ is something you should always be tweaking too - compression is more "set and forget." if the EQ is before compressor and you are constantly adjusting the EQ, it's going to effect how the compressor reacts and may dooty the sound.

THAT BEING SAID

EQing before compression can work wonders if you know how to balance it. Setting a compressor and then placing an EQ before it and boosting the lows can help tighten up the low end of a sound. Some famous engineers have even been known to place a compressor POST FADER and then driving the faders into the compressor can help boost the energy of a mix.

[Artie_Fufkin]

How? I'm not really familiar with reverb algorithms. What do they do, exactly?

[deadly_habit]

I think the only real PEMDAS with effects is not to cause a delay, feedback loop, or clipping, but still sometimes you want that so there is no set in stone rules.

[BassQemistry]

I grab an instrument rack and I split it into 4 channels: Dry signal, low band, mid, high. I usually Add light saturation and compression to the low. I also make the low mono. I sidechain the kick to the low band, so there isn't any gain reduction in the rest of the sound. I fill the mid range up with a good amount of distortion, saturation, EQ, compression, sidechain the snare, maybe sidechain the kick SLIGHTLY, ect. ect..The high band I process with, flanger, phaser, saturation or distortion, harmonic exciter, chorus, subtle delay to make it wider. Nothing is set in stone, I do what my ears like best. However, I always compress the whole bus to glue together the different frequency bands.

[deadly_habit]

I grab an instrument rack and I split it into 4 channels: Dry signal, low band, mid, high. I usually Add light saturation and compression to the low. I also make the low mono. I sidechain the kick to the low band, so there isn't any gain reduction in the rest of the sound. I fill the mid range up with a good amount of distortion, saturation, EQ, compression, sidechain the snare, maybe sidechain the kick SLIGHTLY, ect. ect..The high band I process with, flanger, phaser, saturation or distortion, harmonic exciter, chorus, subtle delay to make it wider. Nothing is set in stone, I do what my ears like best. However, I always compress the whole bus to glue together the different frequency bands.

so you make typical frequency split reeses a la noisia etc.

[Augment]

I've always done my splitting and hard cuts first. I figured that processing a select area of frequencies would yield cleaner results than processing all frequencies and trimming the outputs with EQ. This is a topic of high interest to me with very little documented online.

How do you guys do it and WHY?

This. Processing frequencies that I am then going to EQ/cut away is illogical to my brain - that's not to say it's wrong though, it just makes no sense to me.

With some basses/midranges, I tend to not eq at first. F ex if a bass/midrange has lots of sub, I'll overdrive it a bit first maybe, to get some harmonics of the sub part, then EQ. Doesn't happen too often tho, but if it needs some fat in the lower mids or something, I'll do it

[NinjaEdit]

Reverb algorithm.

You don't have to highpass before compression if your compressor can filter the sidechain, eg with Fabfilter Pro-C or Cockos ReaComp.

Somewhat taking back what I already posted, having some guide effects chain is useful, but a better approach, I think, is more abstract than concrete: Listen for problems, then solve them.