Chapter 6 - Space Manipulation

The sound in a stereo audio recording can be seen as being arranged in a three-dimensional "sound stage." A sound does not usually occupy a single point on any of these axes; rather, it is a three-dimensional "blob" in the sound space.
The X (width) axis of the sound stage is stereo position.
The Y (height) axis is pitch, with higher-pitched sounds appearing higher in the sound stage.
Finally, the Z (depth) axis is distance, with more prominent sounds appearing closer to the front of the sound stage. In this section, we will look at the tools that allow one to manipulate the sound stage; to move sounds forward, back, and to the sides in the mix. We will not consider how to move a sound up or down in the mix.

6.1 Panning

The most elementary tool for manipulating the X axis of the sound stage is panning. Panning can send a sound to the left or the right. It is useful for providing separation between sounds that overlap in frequency range. It is often best to maintain a balance when panning; for each sound that is sent to one side, send a different sound, similar in frequency content, to the other side. Furthermore, the central elements of the music should usually be kept in the center (for pop music, this usually means the drums, bass, and vocals).

Any elements containing significant amounts of bass and subbass frequencies should also usually be kept in the center, for several reasons. Bass frequencies are usually the loudest part of a mix, and if they are panned to one side, then that channel will be significantly louder than the other channel, reducing the net loudness of the mix. Furthermore, when playing back on speakers, it is difficult or impossible to localize bass frequencies, so the panning will probably not be noticed. (And, in fact, if the speaker system has a subwoofer, then the panning will simply disappear.) On the other hand, when playing back on headphones, the panning will be noticed, and it will sound extremely unnatural.

Another thing that can be done with panning is auto-panning effects. When you auto-pan a sound, you cause its panning position to change over the course of the track, possibly quite rapidly. Auto-panning can be a nice ear-catching effect, but if used tastelessly, it can be very annoying. The human ear has been trained by millions of years of evolution to pay particular attention to sounds that are in motion, and auto-panning can distract the listener from the task of listening to music with the task of following the moving sounds.

6.2 Stereo Sounds

You have heard the word "stereo," but what does it mean? Stereo sounds are simply sounds that have width to them, as opposed to mono sounds, which are narrow. Mono sounds occupy a single point on the X axis of the sound space, while stereo sounds straddle a range of the same space.

A stereo signal consists of separate left and right channels, with different signals in them. Most DAWs allow you to treat these two channels as a unit. You can also adjust the balance between the two channels using the "balance" control, which is analogous to (and usually identical to) the "pan" control for mono sounds. If sent to the left, the balance control will reduce the volume of the right channel while leaving the left channel alone; if sent to the right, it will reduce the volume of the left while leaving the right alone.

If a sound is in stereo, that usually means that there are two variations on the same sound, with one variation in each channel. Some sounds are stereo to begin with, such as natural sounds that are recorded in stereo. You can also take a sound that began as mono and turn it into a stereo sound. Essentially, all you have to do is to make the two channels different from each other. There are a number of ways to do this. Here are a few common methods:

You can add reverb. See Section 6.4 for details on reverb.
You can detune the left channel from the right channel. Since this is not possible to do with any standard mixing tool, it must be done before the mixer. This technique is seldom practical with recorded performances, but quite effective for synth patches and short one-shot samples.
You can EQ each channel separately. Usually you would cut the lows of one channel, and cut the highs of the other channel, using a high shelf and low shelf filter respectively, with the same center frequency. This would be done after any other "normal" EQing has been done on the mono source. This technique is rather subtle; you may want to combine it with other techniques if you are looking for a more dramatic stereo effect.
You can create a phase offset between the two channels. By delaying^[1] one of the channels by up to 40ms, you cause the signals coming from the two speakers to be offset, but still perceived as one signal. The sound will be perceived as coming from the side which has the earlier arrival time. This phenomenon is referred to as the "Haas effect."
There are a variety of effects plugins wfiicfi make a signal stereo as a side- effect of tfieir operation (for instance, many chorus effects). There are even plugins, sometimes called "stereoizers," specially dedicated to the task of turning mono signals into stereo signals. Most of them are, internally, based on variants and/or elaborations of the above techniques.

Stereo sounds generally sound bigger and richer than mono sounds, whereas mono sounds generally sound cleaner and punchier than stereo sounds. It is generally not a good idea to over-stereoize your mix. Stereo sounds take more space in the mix than mono sounds, and a mix with overuse or tasteless use of stereo effects can sound weedy and lacking in punch. The key to a good stereo image is to find a good balance between mono and stereo.

6.2.1 Phase Cancellation

Stereo processing can often create problems with "phase cancellation." Phase cancellation occurs when you have two or more instances of the same frequency. When you sum two instances of the same frequency, you might expect to get a louder version of that frequency, and indeed that is often what happens. Other times, however, you will get a quieter version of that frequency, or even silence. To understand why, envision adding together two sine waves of the same frequency. If their peaks and troughs are perfectly aligned (i.e., they are "in phase"^[2]), then the sum will be a sine wave of higher amplitude. If they are offset somewhat (i.e., they are "out of phase"), then the sum will be a sine wave of lower amplitude. If the peaks and troughs are perfectly misaligned, then the sum will be a flat line at zero (silence).

Phase cancellation has two consequences. First, it will hurt the sound somewhat when in stereo, robbing it of its punchiness. Second, and possibly more importantly, the sound will become quieter, or even disappear, when the mix is summed to mono. You certainly don't want your lead instrument to suddenly disappear when someone decides to convert your mix to mono! For this reason, if you are using stereo sounds, it is good practice to periodically listen to your mix in mono to verify that there are no major problems with phase cancellation.^[3]

Many sounds that were recorded or synthesized in stereo have problems with phase cancellation. The phase offset technique (item 3 above) also creates phase cancellation. Problems with phase cancellation are particularly noticeable in lower frequencies, because there are fewer frequencies in that range and they are typically louder.

Indeed, any kind of stereo effects in the bass range are rarely effective, for one reason or another. Reverb (1) muddies up the sound. Detuning (2) creates beating, which results in the low end periodically disappearing and reappearing. Separate EQing (3) is, in this case, equivalent to bass panning, with all of the same problems, since it makes the low end louder on one side. And, of course, phase offset (4) creates phase cancellation.

6.2.2 Left/Right Processing

In order to have control over the stereo characteristics of a sound, it is often desirable to split it into two separate mixer tracks: one track for the left channel, and one for the right. This is called "left/right," or "L/R," processing.

Doing L/R processing requires three or four tracks. First you have the "source" track. This track's output is routed to two tracks: one "left" track and one "right" track. The left track has its pan/balance control set hard left, and the right track has its pan/balance control set hard right. If desired, these tracks are then both routed to one "destination" track, where they are mixed together into the final stereo sound. (This last track is not necessary unless you want to do further processing on the combined sound.)

In the case of a mono signal, this will give you two copies of the same signal, with one in each channel, that can be manipulated separately. In the case of a stereo signal, it will isolate the left and right channels, so that they can be manipulated separately.

L/R processing is a good tool for doing any of the stereo processing techniques described above. You can also narrow the stereo width of the material using L/R processing; by moving the pan/balance controls of the left and right channels towards the center, you can make it progressively more mono.

6.2.3 Mid/Side Processing

There another way, besides L/R processing, to do stereo processing on sounds. It is called "mid/side," or "M/S," processing. M/S processing involves two audio channels, just like L/R processing, but rather than having a left and a right channel, it has a center and a side channel.

An M/S version of a signal can be produced from an L/R version of a signal using nothing more than an audio mixer. To do so is kind of a pain; fortunately, there exist plugins to do the conversion from L/R to M/S and back again. I would recommend that you use one if possible, but also read the following explanation of how to do the conversion by hand, in order to gain a better conceptual understanding of what M/S is.

The mid channel of an M/S signal is half the sum of the left and the right channels. The side channel is half the difference between the left and the right channel. Or, more concisely:

M = {L + R)/2
S={L- R)/2

You can extract the M/S channels from the L/R channels of a sound by first splitting it into separate L and R channels, then mixing these together into the M channel, and creating the S channel by mixing together the L channel and a phase-inverted^[4] version of the R channel. Both channels should then be lowered 3dB.

In order to make use of an M/S-encoded signal, once you are done processing it you need to convert it back to L/R format. The L channel is the sum of the M and the S channels. The R channel is the difference of the M and the S channels. Or:

L = M + S
R= M- S

To convert an M/S signal to L/R, create the L channel by mixing together the M and S channels, and the R channel by mixing together the L channel and a phase-inverted version of the S channel.

Thus, your final signal chain looks like this: convert from L/R to M/S, do processing, and convert from M/S back to L/R. Having set up the signal chain, you have a wealth of options for stereo processing. By lowering the volume of the S channel, you can reduce the stereo width, making the signal more mono, as you could do by bringing down the pan controls in L/R processing. But you can also increase the stereo width, making the signal more stereo, by lowering the volume of the M channel.

Beyond that, there are a wealth of different creative possibilities for making use of M/S processing. By applying separate processing to the mid and the side channels, including EQ, compression, and the other space manipulation techniques that will be discussed later in this section, you can dramatically and creatively shape the stereo character of your sound.

6.3 Delays

A delay, in its simplest form, creates two copies of the input signal, with the second one offset by a fixed time interval from the first. A delay has three controls:

Time: This parameter controls the length of the time offset. Many delays allow you to synchronize this parameter to the tempo of the music, and set it to a musical note length. If yours does not, you can set it "by ear" to a value that synchronizes with the tempo.
Dry /Wet: This parameter controls the balance between the volume of the delayed ("wet") copy and the non-delayed ("dry") copy. 0% silences the wet copy. 50% creates an even balance between dry and wet. 100% silences the dry copy, leaving only the wet copy.
Feedback: Turning up this parameter will result in a certain amount of the wet copy of the delay being fed back into the delay's input. This will result in repeated copies, or echoes, with decreasing volume. 0% feedback will make the delay create only two copies, as previously described. 50% feedback will make the delay create repeated echoes, with each copy being 50%, or 3dB, quieter than the one before it. 100% feedback will make each echo as loud as the last one, meaning that every sound that goes into the delay will echo ad infinitum. Feedback values greater than 100% will result in each echo being louder than the previous one, meaning that the sound coming out of the delay will increase in volume until something breaks down.

Delays can create two different general types of effects, depending on the delay time. With delay times below 30-40ms, the different copies of the sound will not be heard as separate; therefore, the delay will simply modify the character of the sound without creating the perception of multiple copies. With longer delay times, the delay will create the perception of multiple distinct copies. Here are some of the uses of delays:

Comb Filtering: The main effect of a short delay (under 10ms) with no feedback will be to cause interesting phase interactions between the two copies of the signal. The signals will cancel out in parts and combine to cause amplification in other parts. This will cause a complex sonic transformation referred to as "comb filtering." Turning up the feedback will create a more belligerent effect. Comb filtering can be a useful creative tool. It can make some things sound bigger and fuller. It can also be quite annoying.
Haas Effect: If the dry signal and the wet signal of a short delay are panned to different locations in the stereo field, then the comb filtering effect will give way to a stereoizing and localizing effect known as the "Haas effect." This effect is described in more detail in Section 6.2.
Rhythmic Delays: Once the delay time increases beyond 30-40ms, you start getting into the territory of rhythmic delays, where the dry and wet copies are perceived as distinctly separate sounds, arriving one after another. Rhyth- mic delays have a variety of uses. More prominent delays, where the delayed copies are readily audible, can add groove and complexity to rhythmic sounds. Subtler delays, where the delayed copies are not readily audible, can create a general effect of sonic enrichment. Use rhythmic delays on sustained sounds to "embiggen" them, or use low-volume rhythmic delays on an auxiliary send channel to fill out a sparse mix.

6.4 Reverb

Reverberation, or reverb, is a tool used to simulate sound of a natural acoustic space. When a sound is produced in a space, the sound that reaches your ears is heavily influenced by the space itself. In addition to reaching your ears directly from the sound source, the sound repeatedly bounces off the various surfaces in the space, and all of these "reflections" also reach your ears. Reverb units simulate this reflection behavior.

6.4.1 Purposes

Reverb is a highly multi-faceted tool that can be used for many different reasons. Some of those reasons are:

Manipulating Depth: Putting reverb on a sound tends to send it back in the mix. So, if you want a sound to fall into the background, you can achieve that by putting more reverb on it. (Or, alternatively, if you have two sounds that are competing for attention, and you want to bring one of them to the front, you can put reverb on the other one to send it to the back.)

Filling Out a Mix: Reverb adds additional sounds to a mix. These sounds can fill in the holes in the mix, giving a fuller, richer presentation. In particular, reverb is usually a stereo phenomenon, and so reverb will widen the stereo image of your track.

Tying Sounds Together: You can use reverb as a kind of "mix glue," regularizing the sounds of a bunch of different elements so that they sound more like they belong together.

6.4.2 How It Works

The output of a reverb effect consists of three sound components: the dry signal, the early reflections, and the tail. The dry signal is simply the unmodifled input signal. The early reflections are the flrst dozen or so reflections; they are both the loudest and the flrst to occur.^[5] The early reflections sort of merge into the main sound, sounding much like a wetter version of the same. The tail is the remaining reflections, which become a sound in their own right, that can last well after the sound has flnished.

Reverb units are a bit more diverse than, say, compressors or equalizers, but there are still some standard parameters. We will therefore examine a few common parameters and what they do.

Dry, Early, Reverb: These three parameters, or similarly named param- eters, will allow you to set the relative levels of the dry signal, early reflections, and reverb tail. More reverb and early reflections relative to the dry signal will make the sound source seem farther away.

Decay: This parameter controls the length of the reverb tail. A reverb time around 1 second will simulate a fairly typical-size room. A 2-3 second reverb time will simulate a concert hall. Higher reverb times, as long as 7 seconds, will simulate even more reverberant spaces, such as a big, empty cave or a cathedral. Reverb times far beyond that are rarely found in the real world.

Pre-Delay: This parameter controls the time delay between the dry sound and the arrival of the flrst early reflections. A shorter pre-delay simulates a smaller space, and a longer pre-delay a larger space. This makes sense because sound will take less time to hit the walls and return in a smaller space than in a larger space. A normal room will have pre-delay below 50ms. A larger space may have pre-delay as long as 150ms.

Room Size: This parameter controls the density of the reflections, both in the early reflections and the tail. A smaller room size will create denser, more tightly spaced reflections, and a larger room size will create sparser, more loosely spaced reflections. This makes sense because it takes longer for sound waves to travel across larger rooms, and therefore reflections are created with lower frequency.

Damping, Cut: The objects and surfaces in a room, besides reflecting sound, also absorb sound. In particular, soft surfaces are known to absorb high frequencies. Most reverb units will therefore have parameters to change the frequency response of the room.

6.4.3 Convolution Reverb

The previous section applies to so-called "algorithmic reverbs," which create reverberations via mathematical simulations of rooms. In recent years an entirely different technique has gained popularity for simulating reverberation, called "convolution reverb."

Convolution reverb works by taking a recording of an "impulse" (any short, loud sound with wide-ranging frequency content) sounded in a room, and the resulting reverberations. It then processes this "impulse response" to extract the reverberatory fingerprint of the room, allowing it to recreate the same reverberation on any input signal.

Convolution reverb is cool because it allows you to take the acoustics of any space and simulate them in your computer. You can use pre-recorded impulses of the best-sounding spaces in the world, and you can also record your own impulses in any nice or interesting space that you happen to be in.

There are also more creative uses of convolution reverb. You can, for instance, strike a pot or a pan and use the sound as an impulse response. You can create the effect of playing a sound in the "space" of a kitchen implement.

6.4.4 Mixing With Reverb

The most common way to use reverb is to place a reverb unit on an aux send track^[6], and send small amounts of each mixer track to this send track, to infuse each sound with a small amount of reverb.

Generally, each sound should have a large enough amount of reverb that the reverb is audible when the sound is playing in solo, but a small enough amount that the reverb is inaudible when the sound is playing in the mix. Adding reverb to a sound also increases its volume, so you may want to turn the sound's main level down a bit after adding reverb.

When you use reverb in the above way, you wih not hear any reverb tails when playing back the mix (unless the mix is very sparse), but if you mute the reverb channel and then unmute it, you will hear an expansion of the stereo image and a general enhancement of the overall sound quality.

Kick drums and basslines should usually have only the tiniest amount of reverb on them, if any, as the reverb muddies up the bass range. Alternatively, you can put a high-pass filter before the reverb, so that the reverb only affects the higher frequencies of these sounds.

Not all sounds need to have reverb on them. Often, even if a sound is dry, if there are other sounds in the same frequency range that have reverb on them, then that will be sufficient to create the impression of reverb on the former sound. Therefore, if you want a sound to rise to the foreground, you can leave it dry or mostly dry, and let the other sounds carry the reverb. A common approach is to leave the drums and lead sounds mostly dry, and drench the background instruments in reverb.

Don't be afraid to use multiple reverb units, either. I often use two reverb units. One of them is a short decay unit to which I send the drums, to fill them out and help tie them together. One of them is a moderate to long decay unit which I use for everything else. (I send a little bit of the drums to this unit, as well.)

You can also use additional reverb units tailored to work well with specific important sounds, such as vocals or lead instruments. For instance, long pre- delay can reduce the masking effect of reverb on a sound, allowing you to put a lot of reverb on a lead sound while keeping it in the front of the mix.

[1] See Section 6.3.

[2] Sometimes also referred to as "chip shop."

[3] There are also stereo analyzer plugins that can point out phase cancellation in your sound.

[4] Inverting the phase of a signal simply means flipping it upside down. Many DAWs have phase-inversion buttons on their mixer strips; if yours does not, you will have to use a plugin to perform the phase inversion.

[5] More advanced reverb units often allow you to control the timing and amplitude of the individual early reflections.

[6] When using a reverb unit in this manner, be sure to turn down the dry effect level.

Next - »« - Previous