you can easily encode M/S signals, apply your favourite plugins to each signal, then decode back to L/R again. the plugin does not need M/S capability built in.

 

Yes I know that (And I agree that basically you dont need anything more to do a regular proper M/S), but specific M/S treatment might be interesting sometimes.

 

what do you mean by specific M/S treatment?

 

I mean specific process that only a plugin with M/S options can do or simply a plugin designed to do M/S comp eq etc...

 

I use a ribbon for my Fig. 8 too. Have to try the exciter. Sounds interesting.

 

It adds some high air that most lower priced ribbons don't really have, but keeps the rich midrange textures that ribbons seem to do well. We used to work with a real Aphex at the local studio here, so I have a sentimental attachment that may cloud my judgement.

 

Do check Variety of Sound (free but 32bit) plugins if your DAW can bridge or if you use jbridge. Can't recommend enough.

 

Generally you want to be careful with MS processing, the same way you would be careful with multi-band compression. They tools that perhaps you can use to repair a particular problem, but not a panacea that is a given addition to any process. In short as with all things in any process, just cause you can do something doesn't always mean you should.

There is no fixed ratio, +/- 3dB is considered extreme and one must always be mindful of phase issues when filtering or compressing M/S (and it is not a new process Lateral / Vertical or Mid channel + Sides has been around for quite some time even before fancy computers.)

A properly digital recording for digital playback regardless of genra doesn't need really need M/S or side filtering if the real world of playback systems and mono compatibility were a non-issue. Back in the days when vinyl was cut there were some considerations the cutter would have to think about in regards to the physical momentum of the stylus and it's ability to respond to sharp changes in direction over long distances (bass.)

If your intended play back targets are club systems - mix and master with high dynamic range and place mono compatibility and phase correlation at the top of priority list. This might mean you want to HPF everything under 310Hz or even higher sometimes with a 6dBOctave non resonant filter and adjust the mid frequencies up with a shelf or corresponding depth. If you want a hybrid master you might boost with a HSF the side frequencies above 12K or even reach back from 25K with a steep HSF, or a resonant shelf on the side at 8k and high Q as to dip the sides around that frequency. There is no set practice it is an ear thing. All these things are fixes though, if you have the raw tracks, fix it there if you know how, and your time is better invested in fixing it in the mix.

Just remember F/M and that as SPL rises apparent loudness in optimal hearing frequencies rises disproportionately faster than perception of weakly but that those bass / sub-bass and infra bass frequencies are usually boosted and felt rather easily even at low levels on large optimized playback systems.... make the best compromise you can... that is actually what mastering audio is now, unless you are cutting vinyl, just go for a happy medium between large playback systems that can move a lot of air and are generally mono, micro playback systems that are resonant and maybe mono or stereo (phones speakers, cheap boomboxes and cheap stereos or computer speakers) and Hi-Fi systems that are expensive and will reveal deals or deficits quite easily.

Know your room, know your loudspeakers and know how they translate.

As far as MS processors, unless you are using a really old DAW almost everything has a Middle Sum + Sides functionality now. Check what you have there are no magic bullets.

 

Thank you a lot for all of these infos. I'm assimilating now!

I mostly work in FL12 for sound design and music writing, then deal with all the technical aspects with PT12. So yeah both of my DAWs have pretty good M/S capabilities especially FL's Mid/Side knob. Anyway, it's getting much better since some posts here and articles. I've learnt the basic concept of M/S during my sound technician grade, but it was always pretty dark to me, in the digital/DAW world. Now it's a bit more clear.

One last question, why sometimes when I solo things with brainworx meter for example, when referencing, I sometimes see S channel completey out of phase like hitting -1 all the time, M channel hitting +1 too. Even on some commercial "professionnal" tracks, and it really doesnt sound bad (well the solo'd sides are often much more flat and wide when this weird phase thing occurs). But on my work, I really never had a good sounding mix with out of phase sides. How's that?

 

Delay and reverb effects will affect phase reading on a correlation meter, and they will most likely do it in on the lateral channel. If you think of what a delay actually does you imagine the peaks on one side now getting closer to the troughs on the other side especially on for the smaller λ in the upper registers. This isn't much of an issue as long as you can no hear drop outs and your bass isn't phasing. In other words if you are not hearing dropouts in the combined stereo channel and a mono sum, you are fine. The soloed MID channel will always have +1 phase correlation by definition since it is a sum of what is left when you remove the sides and therefore a mono signal. Soloing the sides on a stereo signal will always have 0 or negative phase correlation since you are removing the center channel and now smaller waves will cross inverses more often.... if you can still hear it though then you are not out of phase, your ears are the best discriminator of phase issues along with a mono sum of your signal. Though it is useful to solo L/V channels sometimes to judge frequency presence they're not very useful outside of that.

Hope that helps.