When I first started mixing, I loved the way that the plug-ins could dramatically alter the original recorded audio. Each time I opened up a new plug-in hours were spent scanning through the presets and adjusting every available control. On the one hand, this is a great learning process, finding out how a plug-in can affect your audio. On the other hand, gaining some technical know-how on what each effect does and where to use it in a mix is essential.
Audio effects are often thought of as the aggressive special effects that can be added to sounds to make them pop out and draw attention to themselves. In reality, most audio effects used in mixing and mastering are combined in very slight and subtle ways. Effects are generally used to thicken sounds, add a sense of increased stereo dimension, and probably most importantly help define instruments positions in the soundstage.
If you are new to mixing try to focus your effort on the fundamental effects to start with. Familiarise yourself and practice using EQ, compression, distortion, delays, reverb, and panning. Understanding these effects in depth will lead you down the right path when it’s time to extend your effects knowledge.
You may or may not have heard the terms wet and dry when it comes to audio effects. If a track, or instrument is dry, this indicates that there is no applied effect processing whatsoever. A wet track, on the other hand, has some level of effect processing used on it. The wetness of the track is most often applied as a percentage, between 0% (dry) and 100% (wet, fully effected).
Many plug-ins provide a dial that allows the amount of effect to be applied within it. Those that do not can be set up on an auxiliary track, and the amount of effect can be controlled with either the send amount from the original track or the fader level on the auxiliary track. The former method is often used when an effect will be shared by a few instruments.
For example, a reverb unit could be set 100% wet on an auxiliary track, and then shared by vocals, guitar, and percussion. The level of each track can be adjusted using the send level. As many audio effects require high processing power, the benefit here is that valuable DSP can be saved by sharing the unit.
Specific effects have the ability to put instruments back in a mix or bring them forward. Reverbs and delays can be used to make instruments sound further away. Instruments can also be made to sound further away by reducing their high-frequency content, or by taming transients with a compressor or transient designer.
Instruments can also be placed at the back of the mix by simply reducing the volume or altering the tone with equalization.
Remember that each instrument in the soundstage needs its own space. Space can usually be found between the extremities of the left and right pans, and by using the depth of the stereo field.
Sometimes in a mix it is necessary to bring specific instruments to the fore. This can be done in a few different ways, including adding a little high-frequency content, emphasizing transients, or by adding a very short delay. Adding a delay, which is not in sync with the tempo of the song is an excellent way to accentuate its sound. Distortion is another unique way to emphasize the sound.
Dynamic effects are used primarily to alter audio output in relation to audio input. These effects react, using a set of rules, to input levels and adjust output accordingly. Dynamic effects include compression, distortion, transient designers, and gates.
Let’s take a look at each of these effects in turn.
Most of us have heard the distortion effects created by guitarists using the distortion pedal. In physical units, distortion is created by pushing audio through components above their nominal rated output. Typically the electronic components are pushed outside of their comfort zone in a pleasing, musical way. Favorite components for producing distortion include vacuum tubes, triode valves, transformers, and solid-state operational amplifiers.
Probably one of the best ways to understand the effects of distortion is to take a look at the SansAmp PSA-1 plug-in supplied with most major DAWs (see Figure 1). The plug-in has five control knobs which can produce different distortion effects - preamp, buzz, punch, crunch, and drive. Each knob adds distortion to the audio signal in its own unique way.
Figure 1: SansAmp PSA-1
The effect can be used very effectively in mixing to add sympathetic harmonic distortion to add texture, or to help instruments cut througha mix.
Compressors are designed to reduce the dynamic range of an audio signal. In simple terms this means that peaks are reduced and troughs are increased while maintaining an average, nominal signal level. In many applications the compression can be applied transparently, i.e. there are little or no artifacts created during the processing. However, compressors are also known for their ability to add tone, color, and warmth to instruments by how they process audio.
Audio can be manipulated using the compression unit’s controls, threshold, attack, release, and ratio. As well as being able to alter the dynamic range of incoming signals, a compressor can affect the transients and sustain of instruments. For a full rundown and explanation of compression see the article here.
In a similar way to a compressor, a transient designer can affect an instrument’s attack and decay (sustain) on an audio track. Transient designers are a popular go-to on drum tracks but are also useful for rhythm and synth parts.
A Transient Shaper can sharpen and temper the attack of an instrument, or adjust the delay to be longer or shorter. When added to a multi-band filter this can be a gripping combination.
Using a transient designer can help breathe life into an otherwise uninteresting and lackluster recorded track. It differs from the compressor in that it allows direct control over the attack and decay.
Izotope produces a fantastic multi-band transient designer (I have this as part of my Alloy package), which is suitable for both beginners and advanced users. Another favorite of mine is the Softube two-band stereo “Transient Shaper”, and there are plenty of others out there to choose from.
An audio gate can simply be thought of as a switch, turning the output on and off as audio passes through it. The gate can detect signal levels (volumes) and switch output on and off accordingly. For example, a signal below a gate’s detection signal level is not allowed through the circuit, and in this case the gate is quiet. Whereas a signal above the gates detection signal level is allowed through the circuit.
Additional settings on the gate provide control of the speed to open and close the gate, how long a gate will stay open, and how long a gate will ignore new signals before acting again. Typical controls found on a gate are the threshold, attack, release, hold, and range.
Gating can be very useful to reduce bleed on tracks, particularly close-mic snare and kick drum recordings. On DAWs gating can be achieved manually with clip gain, or by deleting unwanted audio.
Any effect that adds, alters, or manipulates the time base of a signal is known as a “Time Based Effect.” The most obvious effects in this category are delay, reverb, and echo. Time base effects also include “Modulation” but these will be covered in the next section.
What is an echo? An echo is the sound of something repeating. In nature we hear the effect of echo all around us. From inside a large cavern or cathedral, to our bathroom shower, and even sitting in our acoustically treated home studio. Echoes and delays of sounds feel very natural to our ears and are therefore effects that we like to add to our instruments and mixes.
In the digital and analog mixing world, we have control over the number, space between, and volume level of repeats. For example, we may repeat a sound one time, 5 ms later, and at equal volume. We may also repeat a sound 10 times, with varying spacing (5 ms to 20 ms), and at diminishing volume.
On an analog delay tape machine there is a physical limitation to the number of taps that can be incorporated into a design. A digital delay, on the other hand, has no such restrictions and in theory can produce unlimited repeating audio.
Delayed sounds can also be integrated into the stereo sound field. For example a hard right panned instrument, such as an acoustic guitar, can be delayed and hard-panned to the right. Depending on the amount of delay (milliseconds) this can create what is known as the Haas effect (under 35 ms), which adds a sense of depth or space, or an audible delay (over 35 ms) which can create a stereo-like effect.
This industry staple delay uses a single echo, usually with a very short (50 to 100 ms), and at an almost equal volume level, to add a sense of spaciousness on an instrument. This single echo replicates an early reflection of a natural sound. The longer the delay, the more space the instrument appears to have.
A Ping-pong delay has the effect of bouncing a signal from one side of the stereo field to the other. In this case the delay unit simply pans the delayed signal right and left. Most ping-pong delays allow the user to synchronize the delay timing with the BPM of the track. In doing so, this can create a very rhythmic addition to the original audio.
A similar effect can be accomplished by utilizing two mono delays. The original signal is sent to one of the mono delays where it’s output is mixed and fed to the second mono delay. In turn the second mono delay is mixed and fed back to the first mono delay. The amount (volume level) supplied to the delays are usually lowered each time so that the delay effect diminishes.
A flanger mixes two duplicate audio signals together while adding a delay to one. The delayed signal, generally below 20 ms, is varied to produce the effect. The effect exploits the use of comb filtering in its processing.
The origin of the term flanging is disputed. However, one explanation provides the perfect visualization of how flanging works. In the early days of tape machines, used for recording and mixing, tape reels were manipulated by hand to slow down and speed up the audio. Engineers placed their fingers on the “flange” of the metal tape reel to control the motion, and produce the effect.
In more recent times flangers have become available as hardware units, guitar foot pedals, and digital audio plug-ins. Flanger effects have developed over time, to include feedback and inversion of original and processed signals.
You can think of reverb (reverberation) as a repeated delay with absorption. Unlike a standard delay, reverberation is the culmination of multiple reflected sound waves bouncing in all directions until they finally arrive back at the source. Sound waves bounce off surfaces the same way that light reflects from a mirror.
As the sound waves bounce around varying surfaces they lose energy in different parts of the frequency spectrum. Some surfaces absorb high frequencies while others absorb mid or low frequencies. It is the multiple reflections and varying frequency absorption that produce the reverb effect. The reverb effect will continue until the initial sound has lost all of its energy.
As humans, we all easily recognize the effect of reverberation. Blindfolded, we would all recognize from the sound alone whether we were in an elevator, our living room, a cathedral, or a long and winding cave. Part of our internal recognition system allows us to distinguish early reflections. The early reflections in reverberation provide the audio clues about the size of the space and the type of material within that space. In particular it is the first reflection (pre-delay) that provides the most significant clue about size.
Most digital plug-ins provide controls to adjust pre-delay, early reflections, frequency absorption, and the length of the output. The controls can be used to craft the reverb sound most suitable for a mix.
Applying reverb to a snare can help make the instrument stand out (by giving it some space). A common approach is to use a plate reverb and add enough pre-delay so that the initial snare transient is largely unaffected.
The story of snare reverb would not be complete without mentioning gated reverb, a technique of gating the output of the reverb unit so that the reverb part of the sound is chopped off at both ends.
The gated reverb on the snare became iconic in the 1980s as artists such as Peter Gabriel and Phil Collins brought them to the fore popular music. If you’ve never heard the effect check out Phil Collins’s “In the Air Tonight” from the 1981 album “Face Value”.
Once again reverb can be used to great effect for placing vocals in an artificial space. In my own mixes I like to get the drums and vocals to sound like they were both recorded in the same room. In reality they weren’t however reverb can be used to make them sound like they were.
Unless you’re going for a very dramatic effect reverb on vocals should be done very subtly. Even the slightest addition of the wet signal will turn a dry vocal into something very, very natural sounding. Filtering the incoming vocal reverb signal with EQ can also dramatically enhance the vocal sound without overdoing the effect - something that Abbey Road pioneered in the early days of producing records.
Strictly for special effects, audio can be reversed to produce exciting and novel additions to recorded tracks. Most modern DAWs provide functionality to reverse digital audio files. If you don’t have access to this functionality on your own DAW, the effect can be carried out using free software such as Audacity.
There have been many uses of the reverse effect on music produced since the 1950s. The Beatles and George Martin experimented with reverse audio on Strawberry Fields Forever, The White Album, and Revolver.
Modulation effects can be categorized into amplitude (volume) modulation, frequency modulation, and phase modulation. Each of which alters, and adds to the source audio. Modulation effects include flangers, phase shifters, and choruses.
Some effects use low-frequency oscillation (LFO) to modulate audio. LFO generally occurs below 20 Hz and an oscillating wave is used to alter the original signal.
The primary modulation effects are chorus and phaser. However, variations on these modulation effects give rise to sound-shaping tools such as pitch shifters, tremolo, harmonizes, resonators, and vibrato. All of which are available as hardware units, guitar pedals, and digital audio plug-ins. Some plug-ins have the option to choose between many of the modulation effects listed here.
The chorus effect is created by delaying several copies of the original signals. Each copy of the signal is increasingly further away from the original audio. The effect is used to artificially create a sense of more than one voice or instrument. Copies of the original signals are modulated using an LFO to add to the sense of diversity. The chorus effect is often used in mono recorded vocals to thicken the sound.
The phaser effect is created by duplicating the original signal and altering its phase. The effect incorporates an all-pass filter, and an LFO applied to the phased signal. When the two signals are mixed back together their phases are misaligned, producing it’s very distinct characteristic sound. This effect is often used on electric guitars and synthetic sounds.
We have looked at what are probably considered the main effects of mixing. As our definition of effects is not just for special effects, below are a few more to complete the list.
Panning allows mono tracks to be positioned anywhere in the spread of the stereo field. A mono guitar track, for example, could be placed hard left, center, hard right, or anywhere in between. Panning is particularly useful in stereo productions as moving an instrument off center can help to unmask the instrument being panned and the instrument occupying the middle of the stereo field.
A useful way to use panning is through automation, either as a special effect, for example a guitar screeching left to right over a few seconds, or as a way to move an instrument “back and forward” in different parts of the song. For example, hard panning a stereo saxophone part to the extremes during verses and narrowing the spread during the choruses.
Equalization, or EQ, is used to attenuate or boost frequencies in an audio signal. EQ is used as a filter to remove unwanted or problematic frequencies or to enhance key parts of the frequency spectrum. EQ is particularly useful for adjusting the volume of an instrument without increasing the whole frequency range.
Saturation is a musical form of distortion. When saturation is applied to a track small amounts of distortion are added to the instrument’s harmonics. In the analog days of recording and mixing saturation was inherent with the equipment used at the time. Analog equipment was (and still is) designed using vacuum tubes, transformers, tape reels and tape heads, and transistors. All of which naturally add harmonic distortion to audio passing through them.
The added harmonic distortion provides tracks with new frequencies which can help instruments stick out a little more. This obviously helps with masking but also provides instruments with a little more punch, warmth, or depth.
Saturation is particularly useful on vocals, drums, bass which has been recorded with the DI, synthetic instruments, and on the mix bus.
Most home studio owners mix in the box. That is, the recording, mixing, and effect processing is all done using software in the form of a DAW and plug-ins. However, it is certainly possible for a home studio owner to add external reverb, delay, compression, or other hardware units to a home studio set up. To do this requires a lot more technical know-how for routing to and from the hardware units, so the majority of people stick to mixing in the box.
Auxiliary tracks on a DAW a very useful for sharing single effects units with multiple instruments. The basic setup suggests that the effect is set to 100% wet, and the auxiliary track fader is at 0 dB. The amounts sent to the auxiliary track can be controlled by the send fader. As we said earlier using a shared effect can really help with processing power.
Sometimes an effect does not need to be shared with multiple instruments. In this case the effect can often be placed directly on the instrument track. Some plug-ins allow control of the effect amount using a wet and dry control or can be controlled directly on the plug-in’s parameters.