Why All the Analogue Emulation?

Slate Virtual Tape MachineThe age of digital audio is the age of perfect stability. Of flawless copies and non-destructive editing. After decades of analogue audio designed to be ultimately transparent, digital has finally delivered on that.

As such, enthusiasm amongst senior engineers is commonplace. You’d be hard pressed to find an audio engineer of a certain age who’s not incredulous at the vinyl resurgence. In theory, even the stock plug-ins that come with your DAW should be able to deliver perfect sounding audio.

So what’s with all the fuss around analogue emulation in plug-ins? Looking at the biggest sellers in digital pro audio you’d be forgiven for thinking the recording industry hasn’t moved on since about 1973. Surely with all the imperfections of analogue equipment, the binary perfection of digital should be a blessing, shouldn’t it? Why spend so much time and money on analogue emulations?

Familiarity of analogue

One argument’s that often trotted out is the simple familiarity of analogue equipment. Firstly amongst engineers who ‘grew up’ surrounded by 1176s, dbx 160s, Neve desks and so on, but also amongst listeners. They may not consciously recognise it, but even the least tech-savvy listener is familiar with the sound of analogue recording from all those Rolling Stones records in their mp3 collection.

Engineers and listeners alike are used to the sound analogue equipment makes. And for engineers who learnt on analogue equipment, combining the efficiency gains of digital with the same workflow and sound as analogue means they can immediately know what a plug-in will sound like (or at least should) and keep their old habits. The classic pieces of kit serve as useful signposts when choosing which processor to use.

Ease of use

One of the great advantages of digital is the unparalleled flexibility it provides. EQs that can target frequencies down to a single Hz, compressors tweakable to the microsecond; these processors don’t exist in the analogue world.

However, that can also be a downside. As anyone who’s ever used any real analogue equipment will tell you, the limited set of controls can actually be a joy. It speeds up your workflow as you’re not inundated with endless options, and can quickly narrow down choices to ‘this one or that one’ rather than micro-tweaking how many decimals of a decibel you’re cutting 53Hz by.

There’s something to be said for replicating the limitations of analogue within the digital realm.

Non-linearity and distortion

The area of analogue emulation that’s received the most attention lately is the non-linearity and associated distortions. Slate Digital are particularly hot on this.

Simply put, digital is linear. No matter what level the audio is at, it will be captured and played back exactly the same. Analogue is non-linear. The higher the level, the more distortion is introduced. And this distortion isn’t necessarily just harmonic (i.e. multiples of existing frequencies). Up until very recently, plug-ins were unable to replicate this.

These are essentially the imperfections that make analogue sound different from digital. And the sonic character of classic pieces lives in these imperfections. But analogue emulations have come a long way, and plug-ins are now capable of very good approximations.

Is it worth it?

Let’s say you’ve just bought Cubase. It comes with all the plug-ins you could possibly need: compressors, EQs, multiband compressors, limiters, reverbs etc. Every tool in the audio box. Do you need to spend money on a UAD card or Slate plug-ins?

Technically no. Technically you can make perfect sounding mixes using just these plug-ins. Technically, you can come pretty close to analogue sounding anyway if you know your theory well enough (by tweaking EQ the right way, using compression to emulate tape compression etc.).

However, as a mastering engineer, I’ve worked on enough mixes to know when someone hasn’t used analogue emulations (or analogue). And a lot of the work actually goes into dirtying them up. They come in perfectly clean, lacking depth and life. They sound small. The perfection of digital is why adding some form of distortion or harmonic excitement is now common in mastering – to compensate for the missing layers of distortion inherent in analogue mixes. Distortion that makes mixes sound big and exciting.

And I have to say, analogue emulations do make mixing easier. Not just in the operation of the plug-ins themselves, but because using them makes mixes come together quicker and with less effort.

It’s true that there’s a lot of marketing hype surrounding analogue emulations. But it’s not all hype.

The Beginner’s Guide to Mid/Side Processing

Mid/side processing is a powerful technique that can be extremely useful in the right circumstance. But it can be a little difficult to get your head around and understand exactly what it’s for. Here we look at what mid/side processing is, how to achieve it and what to use it for.

What is mid/side?

Mid/side – sometimes called sum/difference – is a technique whereby a stereo recording is changed from the usual left and right to mid and side. The ‘mid’ or ‘sum’ is the mono portion of the recording (everything the left and right signals have in common) and the side is the stereo portion (everything else).

Using mid/side, we can more easily zone in on different parts of the stereo recording for EQ or compression say, or even manipuate the stereo image itself.

Manipulating M/S with free plugins

The easiest way of constructing an M/S matrix is to use a free plug-in like Voxengo’s MSED. This let’s you chain the relative gain of the mid and side signals to effect stereo width (another free plugin for this is included in BlueCat’s free gain package).

MSED will also act as an M/S encoder and decoder. Set it to encode and the left portion of the signal will become the M and the right, the S. You can then use any plugins that allow you to manipulate the L and R channels separately (but that don’t include an M/S matrix themselves) to manipulate the M/S signal, before sticking in another instance of MSED set to decode to convert it back to stereo.

How to construct a mid/side matrix without plugins

As easy as using MSED is, for additional flexibility (and understanding of what’s happening) it’s worth constructing your own M/S matrix at least once or twice.

The principle is simple – the M signal is everything common to the L and R signals, and the S is everything that is not. So, to construct your own M/S matrix:

  1. Split your stereo recording into 2 mono tracks – 1&2. Pan centre.
  2. Make duplicates of each one – 3&4. Pan centre.
  3. Send each pair to a separate group – buss 1&buss 2
  4. Invert the phase on track 4.
  5. Create 2 new buses – M&S
  6. Send buss 1 to M and buss 2 to S
  7. Pan M hard left and S hard right
  8. Invert the phase on buss S

This way you can now use any plugin that works in mono to manipulate the M&S signals separately.

Common uses of M/S processing

M/S processing is useful whenever you need to really zone in on a particular part of a stereo recording and the stems aren’t available. For example, if there’s an edgy sounding guitar at the edge of the field you can use an EQ cut on the S channel only to smooth it out.

How successful this sort of thing is depends on where instruments are panned, sometimes you just have to leave it be. Some common uses of M/S processing though are:

  • Stereo width
  • Increasing/decreasing stereo width by adjusting the relative gain (more S = wider). This has fewer artifacts than using ‘stereo widener’ plug-ins that often make use of comb filtering. Another approach is to use a high boost (8-16kHz) on the S channel to increase top end width.
  • Mono-ing bass
  • Tighten up bass by using a high pass filter at around 100Hz on the S channel.
  • De-essing
  • Use a de-esser on only the M channel (for lead vocals) or S (for backing vocals) in a complete mix.

As you can see, the sorts of problems that M/S processing can address are usually better dealt with earlier in the chain – but when that’s not an option, mid/side can be tremendously powerful.

The Beginner’s Guide to Buss Compression – Part 2 – Master Buss Compression

Last time we looked at the basics of buss compression – what it is, where to use it, and what compressor to use. This time we’ll be looking at master buss compression: a subject that flummoxes many budding engineers but that is essential for cohesive mixes.

What compressor should I use?

Compressors built for the master buss are subject to much adulation, mythology and devotion. Everybody has their own favourite so the answer really is to download as many demos as you can and try them out.

The same rules apply as for mix buss compressors (and really they’re generally the same models/plug-ins). There are the fast and ‘grabby’ models in the tradition of the SSL G series and the more gentle ‘vintage’ style.

Probably the most popular software master buss compressor at the moment is the Waves SSL G-Master Buss. But a cheaper alternative (just as credible, and preferred by many, including myself) is Cytomic’s The Glue.

Famous hardware compressors used for the master buss include API 2500, SSL G Series and Smart Research C2 in the fast and grabby vein, Manley Vari-Mu and Thermionic Culture Phoenix in the slow and gentle tradition of the infamous Fairchild 670.

Try out various software emulations and you’ll quickly get a feel for what style of compressor you prefer.

Why compress the master buss?

Master buss compression is most often cited as necessary for ‘glue’ i.e. a sense of cohesion. It’s often talked about as ‘what makes a record sound like a record’. Good use of master buss compression will give a mix definition, punch and depth.

When you first experiment, you’re listening mainly for all the parts gelling together, sounding like one piece of music rather than separate parts. As you get more confident you’ll learn to use slight pumping to introduce movement and punch – but be careful not to overdo it. As with all use of compression, too much can result in everything sounding flat and unexciting.

How do I set it up?

Needless to say, the two different styles react very differently and so require very different settings. This again is subject to much individual style. It’ll take a few mixes for you to zone in on your own personal favourites, but when you get them you can save it as a preset so it’s all set up when you start mixing. Always mix into your compressor rather than slap it on afterwards.

When mixing, I tend to set up The Glue with the following settings:

  • Threshold -14dB
  • Attack 30ms
  • Release 200ms
  • Ratio 2:1
  • Sidechain HP filter at 100Hz

I then mix so that the needle is just about pushed by the kick. If I want a gentler compression I use Stillwell Audio’s Bombardier using the following:

  • Threshold -20dB
  • Attack 40ms
  • Release 200ms
  • Ratio 2:1
  • Feedback/forward depending on the song
  • Again, there should be no more than 2-3dB of gain reduction – just enough to achieve depth and glue.

    The advantage of having a set threshold to work to is that you end up with mixes that are roughly the same level, without having to turn it down to prevent clipping. It gives you a good starting point for your kick and vocal to act as reference levels for everything else.