• Plugins
• Bundles
• Tutorials
  • Introduction
  • Compression
  • Limiters
  • Equalizers
  • Mixing tutorial
  • Mastering tutorial
  • Stereo expansion
  • Analog vs. Digital
  • 32-bit vs. 64-bit

Principles

Let's start off by taking a look at the physical side of our hearing. Each of our ears, has an area of focus. We can clarify this by saying that each ear is recording sound coming from both a side axis and a smaller area in front via the pinna. Theoretically, for us to be able to hear the widest possible sound we can, each ear would have to receive an entirely different sound to the other. This can be achieved either by wearing headphones, or by positioning a monitor next to each ear, and playing something completely different into each channel. This can sound very disturbing, so we tend to use a stereo analyzer (e.g. MStereoProcessor), to show the spatial characteristics of a sound..

Stereo analyzer basically shows 2 diagonal axes, one for the left and the other for the right channel, with each point representing a sample. If it displays a vertical line, then both channels are completely identical indicating that the signal is monophonic. If a horizontal axis is produced, then the channels are completely out of phase, and one is an inverted version of another. The ideal shape would be a vertical ellipse, not too thin but not too wide either as this can sound artificial. If we return to our earlier example and play something completely different into each ear, then the stereo image would probably look like a circle, becoming more vertical or horizontal from time to time. This indicates that both channels are not correlated.

The goal of a stereo expansion plugin is to take a monophonic or very thin signal, and make it wider. Optimally, the resulting audio would also be mono-compatible, which means that if it is 'monoized', for example when it is played on the radio, then it wouldn't lose too much of it's frequency content. This can happen if a vocal is out of phase for instance, as mixing it into mono will attenuate it a lot or even remove it completely! So how is this done?

Stereo-mono encoding

One of the oldest methods is to simply extract the mono and stereo content and either amplify the stereo part or attenuate the mono part. The method is very simple:

stereo = l - r
mono = l + r

See what we mean? Mono is the sound which is present in both ears. Stereo is also present in both ears, but is out of phase. So the output can be represented as

output = 0.5 * mono + 2.0 * stereo

It's a little bit more complicated, but the idea is to suppress the mono content while enhancing the stereo. However, there's a catch: there must already be some stereo content present! Additionally, amplifying the stereo content too much, may result in distortion or extra noise. Nevertheless, this is generally the most common method used to control the stereo content. In MStereoProcessor you can control multiple bands separately, thus minimizing the distortion.

The advantage with this method is that it is completely mono compatible.So if you monoize the result, the frequencies that cancel each other would also cancel each other in the original audio as well. Simply put, the result will be exactly as mono compatible as the original.

Channel delay & phasing

What if we were to delay one channel by 10ms? Well, the output would not be correlated anymore causing it to sound 'very stereo'. Unfortunately this kind of stereo is nothing other than 'out-of-phase' which we normally try to prevent when recording, so it is best avoided!

This is because the brain interprets such changes in phase as differences in the position of the sound source, so you essentially move the sound position or change it's direction. What's more, the sound becomes very mono incompatible. Mixing it to mono, would remove lots of frequencies, making it sound weird or metallic, just like a typical comb filter. Some cheap effects units still use this method, but they are probably best avoided unless you have some creative use for them.

Phaser is an example of an effect that uses a similar approach creatively. Phasers selectively alter the phase content of the audio signal and as this can be done a different way for each channel, they produce a good deal of stereo expansion. Obviously the results are not really mono-compatible, so the effect is more commonly used on individual tracks, such as guitars and keyboards. Download the free demo of MMultiBandPhaser to try the effect for yourself.

Spectral modification & comb filtering

Another popular approach is to change the spectral content of both channels in such a way, that when summed together, both channels will cancel any changes made, therefore making it mono compatible. Imagine it as having 2 equalizers, one for each channel, and using several peak filters, but with opposing values on each channel (e.g. the 1kHz channel would be +2dB in the left channel, but -2dB in the right channel).

This method produces the same phase shift in both channels, ensuring that they are correlated, but with enough of a difference to make them sound stereo.

This is essentially a special kind of comb filtering, which can be found in a number of free or low cost effects. However, it also has several disadvantages not least of which is the sound, which can be too artificial, or 'weird' due to the way the filtering removes several frequencies from each channel. This can also produce spatial issues, which may move the source artificially in space, often appearing to be further away or present on both sides of the mix.

Despite all of it's shortcomings, this effect can initially appear to produce pleasing results although your ears will usually tire of it after just a few seconds. So if you really want to use it, do so with a lot of caution.

Multivoice methods - chorus, flanger, doubler...

Many decades ago the well-known Swedish band ABBA came up with a cool way to enhance their recordings - they recorded each instrument multiple times and panned them to the sides. Recording engineers still use this technique today, and it now forms the basis of the multivoice methods of stereo expansion.

The idea is to take the audio, modify it somehow so that it sounds like a new recording, and place it somewhere in the stereo field.

This method may be seen as a mixture of all the previous methods put together. It creates separate copies of the original signal, pans them, shifts them and delays them etc. The biggest problem is, how to make the other voices different enough to the original audio. If they are too similar, we get the typical artifacts already mentioned - comb filtering, phase distortion etc.

Flanger is a device that creates a single copy of the original audio, but with a slightly different pitch and varied time delay. The difference between the original signal and the added voice is minimal as is the variation in time. There is also a feedback line that induces strong comb filtering. This results in the typical 'flanging' - sweeps across the frequency spectrum.

Chorus on the other hand, tries to make each voice as different as possible, so that it seems like there are many people playing small variations of the same part. Chorus usually provides much more natural stereo expansion, and also doesn't suffer from the mono-compatibility issues as much when implemented properly (which is unfortunately not the case with most choruses on the market). You can check MMultiBandChorus and MMultiBandFlanger to see how a well designed chorus or flanger works.

There are also other ways to achieve the same type of effect, like doublers for example. These devices however, usually offer very similar functionality and so ultimately, a chorus with enough parameters can become a universal solution. Care should be taken when using this type of device as they manipulate the pitch of the audio, and therefore many settings can easily produce a vibrato effect, which is often not desirable.

Reverberation

The most natural method is to simulate what happens in nature where the sound gets reflected from the many things all around you creating thousands upon thousands of copies, each slightly different (but with exactly the same pitch). Reverbs can be very personal, everyone has individual tastes and a favoured type. The general aim is to make a reverb that sounds wide, but still natural. You can check MReverb and MMultiBandReverb to hear how they work.

Generally, reverbs are some of the most common spatial devices. Reverbs are used a lot during mixing, though as usual, there are some disadvantages. Because they create distinct copies of the original audio, they will either create a long tail, or the sound won't be wide enough. They may also allow you hear the artificial reflections.

Conclusion

We have shown several of the methods in use today to make (almost) monophonic audio wider. Stereo field adjustment is useful primarily for controlling the wideness and should not be used for widening as a general solution. Spectral modification and channel delays shouldn't be used at all unless there is some creative need for them. Choruses, flangers and other voicing methods are used a lot for separate tracks. And finally, reverbs are used for separate tracks and whole mixes. They usually provide a smaller amount of widening and the obvious space-effect, but the results are more natural, especially when used on whole mixes.