Dynamic processors are used to control the level of individual tracks or an en-tire mix. Th ey can be used to increase the perceived loudness of a recording or to make a track more dynamically manageable.
Compressor
A common use for a compressor is to tame or tighten a track in which the dy-namic range is too great. For example, a lead vocalist might take a boisterous approach to one phrase and a subdued approach to another. It might be im-practical to ride the fader level in order to bring the subdued phrase up in the mix without letting the boisterous section overpower. A compressor can, to an extent, automate this process. Compression has earned a bad reputation since some engineers feel that contemporary popular music is overcompressed and lacking in dynamic nuance. However, some amount of compression is neces-sary so that a recording will retain its clarity on a variety of playback systems.
Interestingly, our ears even compress sound. Neuroscientist Daniel J. Levetin writes : “Th e inner hair cells have a dynamic range of 50 decibels (dB) and yet we can hear over a 120 dB dynamic range. For every 4 dB increase in sound level, a 1 dB increase is transmitted to the inner hair cells.”1
Compression can be counterintuitive at fi rst glance. A subjective descrip-tion of the applicadescrip-tion of compression to a recording is that it “makes things sound louder.” However, a compressor actually works by lowering or attenuat-ing levels. Th is seeming contradiction can be understood by considering the example of a singer whose performance is too dynamic to work with a given mix. Where an adjustment to a fader level would increase the volume of the soft phrases and louder phrases, a compressor puts a limit on louder signals while providing a mechanism to increase the level of soft er signals. Simply put, there
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will be less dynamic contrast between loud and soft passages when compression is applied. Figure 10.1 will help to clarify this concept.
FIGURE 10.1 Compressing a signal vs.
increasing overall level soft loud soft loud soft loud soft loud
increasing level applying compression
Components of a Compressor
Th e four primary components of a compressor are compression threshold, compression ratio, attack time, and release time (see fi gure 10.2). Each of these controls is interdependent so it can take time to develop a feel for the adjust-ments that are necessary to compress a track to best advantage.
FIGURE 10.2 Simple compressor
Compression Threshold
Compression threshold refers to the input level at which compression will occur.
When the threshold is reached, the signal is reduced by the compressor. Th e compressor will not respond to levels below the threshold so a high-threshold setting may have the eff ect of turning the compressor off .
Compression Ratio
When a signal reaches the compression threshold, compression is applied and the amount of compression is determined by the compression ratio. Higher ratios provide more compression and less dynamic contrast. By way of example, a compression ratio of 2:1 means that the output will increase 1 dB for every 2 dB increase in the input signal. A 4:1 ratio means that it will take a 4 dB in-crease in the input to inin-crease the output by the same 1 dB.
169 Attack Time
You can think of attack time as a compressor’s reaction rate. Higher values mean that it will take more time for the compressor to respond once the threshold level has been reached. Longer attack times will retain the initial attack of an instrument or voice but may cause the compressor to be ineff ectual. Shorter attack times are useful in maximizing signal level but may have a detrimental eff ect on tone. As with other compression parameters, attack time must be care-fully adjusted to fi t a given input signal.
Release Time
A compressor will stop compressing a signal when the input falls below the com-pression threshold. Release time refers to the number of milliseconds it takes for the compressor to return to its normal state.
Knee
Some compressors provide an additional parameter called the compression knee.
Th e knee relates to the way the compressor functions as an input signal ap-proaches the compression threshold. In a hard-knee compressor, compression is immediately (and fully) applied when the threshold is reached. In contrast, a soft -knee compressor gradually applies compression as a signal approaches the threshold. As Bob Katz states in his Mastering Audio book, “soft knee can sweeten the sound of a compressor near the threshold. For those models of compressors that only have hard knees, some of the eff ect of a soft knee can be simulated by lessening the ratio or raising the threshold, which will result in less action.”2
Th e samples on the companion Web site will help to illustrate some of the common applications of a compressor. In the fi rst example, a subtle amount of compression is applied to an electric fretless bass. Notice how the compressed track has a tighter sound because the dynamics are more consistent.
In the next example, compression is used to make a snare drum “pop.” In this instance, the compression ratio is fairly high and a touch of reverb and EQ is added.
Compression is particularly useful when applied to a vocal track. In the third example, the singer uses many subtle nuances that might be lost in the mix without the application of compression.
Multiband Compression
A multiband compressor is a useful tool that provides the capability to apply compression independently to specifi c bands of frequencies. For example, com-pression might be applied to the band of frequencies from 5 kHz to 20 kHz in order to impart a sparkling quality to a recording without aff ecting other bands of frequencies where compression might have a detrimental eff ect. Similarly, a multiband compressor could be used to tighten lower frequencies without
EXAMPLE 10.1
EXAMPLE 10.2
EXAMPLE 10.3
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making the recording overly bright or sibilant. Notice how much more vibrant the next musical excerpt sounds. Th is is due to the way multiband compression is utilized on the main bus.
Limiter
A limiter essentially functions as a compressor at an extremely high ratio. Whereas a compressor compresses signals proportionally, the limiter reduces any signal over the threshold to the value of the threshold. As such, a limiter is typically used as the last processor in a main mix to keep signals from exceeding 0 dB.
Limiters can also be useful during the tracking phase. Although it is not possible to use a limiter to remove distortion that is recorded to a track, a lim-iter can prevent a loud attack from overdriving a channel. In the next example, a limiter is applied to a bass track to prevent the small amount of distortion that is heard when the signal overdrives the channel.
Expander
An expander is similar to a compressor, but instead of reducing dynamic range, the expander increases dynamic contrast. When a signal reaches the expansion threshold, the output is increased by an amount that is determined by the ex-pansion ratio. For example, an exex-pansion ratio of 1 to 4 means that, for each 1 dB of input, the output expands by 4 dB. An expander is sometimes used aft er a compressor in the signal chain to make the music sound more vibrant. It is in-teresting to compare the sound of the next excerpt with and without expansion.
Noise Gate
A noise gate is another type of expander, but it is used to remove unwanted sounds. A noise gate can also be used to tighten a drum track or for special ef-fects. Th e primary controls on a noise gate are the threshold and reduction knobs. Th e threshold control determines the level below which a signal will be reduced, whereas the reduction control determines the amount of reduction that will occur. In the next screenshot, note the inclusion of a three-stage enve-lope consisting of attack, hold, and release parameters. Th e attack and release parameters function in a similar way to their counterparts in a compressor or limiter. Th e hold parameter determines the amount of time the gate will be held open aft er a signal falls below the threshold.
In the next sample, a gate is used to remove extraneous noise from a gui-tar amp. Gates can also be used to tighten the sound of a percussion instrument by limiting the sustain of high-level transients or instrument ringing between attacks.3
Look Ahead
Some dynamic processing plug-ins provide the capability to analyze prerecorded information prior to the point at which the signal actually enters the processor.
Th e benefi t of engaging look-ahead is that the processor can respond in a more
EXAMPLE 10.4
EXAMPLE 10.5
EXAMPLE 10.6
EXAMPLE 10.7
171 musical way by anticipating events. Th e disadvantage is that the process
intro-duces latency—the diff erence in time between when a signal arrives at an input and when it reaches the output stage. Latency is not a problem during the mix-ing process, but look-ahead processors should be disabled when recordmix-ing be-cause any signal delay will be-cause problems when musicians attempt to play to a click track or overdub a part.
Equalization
An equalizer is a circuit that “allows the frequency-selective manipulation of a signal’s amplitude.”4 EQ is utilized for a number of reasons, including to change the timbre of an instrument so it will better fi t within a mix or to fi x a problem such as sibilance or low-frequency rumble. EQ might be used to make an entire mix sound brighter or darker (similar to the way treble or bass controls are used in a home stereo). Yet another application is to tone down certain frequencies that leak into a microphone. For example, an engineer might attenuate higher fre-quencies on a kick-drum track in order to tone down bleed from the cymbals.
It is important to remember that EQ can be used to boost a range of fre-quencies, but it can also be used to attenuate frequencies. I mention this be-cause, for most of us, it is natural to want to turn up a range of frequencies when, in many instances, this can result in a mix that sounds muddy or can be fatiguing to listen to. You will oft en have better results by toning down a range of frequencies. Th e next Web site example is a case in point. Th e fi rst excerpt demonstrates a muddy and fatiguing mix that resulted from the overapplication of equalization. In the second example, more clarity and warmth is achieved by attenuating frequencies.
Shelving EQ
Th ere are a number of types of fi lters that allow an engineer to adjust the level of specifi c frequencies. You are already familiar with a shelving EQ, since this is the fi ltering method used in the treble and bass controls on most home sound systems. Th e curve of a shelving EQ slopes to (or away) from a preset value and then fl attens out to form a shelf. Notice in fi gure 10.3 how a shelving fi lter is used to attenuate low frequencies.
EXAMPLE 10.8
FIGURE 10.3 Shelving EQ
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High- and Low-Pass Filter
High-pass and low-pass fi lters function in a similar way to a shelving EQ, with the exception that the fi lter curve does not fl atten out. In this instance, fi ltering continues at a rate that is determined by the fi lter slope. As the names imply, a high-pass fi lter lets high frequencies pass and a cutoff is established to fi lter low frequencies. Similarly, a low-pass fi lter lets low frequencies pass and a cutoff is set for the removal of higher frequencies. Compare the curve of a high-pass fi lter in fi gure 10.4 with the low-shelving EQ from the previous fi gure.
FIGURE 10.4 High-pass fi lter
Peak Filter
A peak fi lter is used to amplify or attenuate a range of frequencies around a particular center frequency. Th e term Q, sometimes referred to as bandwidth, denotes the range of frequencies that are aff ected around the center frequency.
A high Q value means that a narrow band of frequencies will be aff ected while a low Q value will aff ect a broad range of frequencies (see fi gure 10.5).
FIGURE 10.5 Peak (bell) fi lter with high and low Q values
Multiband EQ
Multiband EQs are found in most DAW programs and will typically consist of a shelving EQ and/or high- and low-pass fi lters to control the high and low bands of the frequency spectrum. Several parametric EQs are used to control the middle bands of frequencies. Notice how the bands of a multiband EQ work together to alter a broad range of frequencies, as shown in fi gure 10.6.
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Effects
Th ere are a wide variety of eff ects available in modern audio workstations. Ef-fects like reverb and delay can help to give a recording a sense of ambience or spaciousness while processors such as a phase shift er or chorus are typically used for special eff ects.
Reverb
As you learned in the chapter on acoustics, reverberation results from multiple refl ections of a sound source as sound waves interact with the surfaces of walls and objects in a room. While natural ambience or reverberation is part of most classical recordings, popular recordings typically utilize close microphone place-ment to minimize the natural acoustics of a room and artifi cial reverb is added during the mixing and mastering process. As detailed below, digital reverb plug-ins provide a number of parameters that can help you to tailor reverberation for any application.
Room Size/Reverb Time
Early refl ections are the primary aural cue that help us determine the dimensions of a room.5 One of the reasons we can aurally diff erentiate between a small room and an auditorium is the delay in time between the start of a sound (e.g., clap-ping) and the early refl ections. As such, room size is a primary consideration in establishing the dimensionality of a recording. Some reverb plug-ins provide the ability to adjust the dimensions, refl ectivity, and even the shape of a virtual room.
Density/Diffusion
Density refers to the number of reverberations. A diff use sound is created by multiple refl ections while echo eff ects can be created when individual refl ections are discernable.
FIGURE 10.6 Multiband EQ
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Pre-delay
Th e term pre-delay is used to defi ne the amount of time between the start of a signal and the return of early refl ections. Although pre-delay is conceptually similar to room size (e.g., it will take longer for early refl ections to return in a large room than in a small room), the two parameters work together. One way to visualize this concept is to think of a band that is performing in a large room.
If you stand in the middle of the room, early refl ections will return to your ear at a rate that is determined by the dimensions of the room. Now, imagine that you are standing at the end of a long hallway that is connected to the room via a large door. Th ere will be a delay before the reverberant sound of the band reaches the end of the hallway. Th is delay of reverberant sound is conceptually similar to the pre-delay setting found on most reverb units.
Mix
Th e mix parameter refers to the mixture of original (dry) signal and the pro-cessed (wet) signal. As with all of the other reverb parameters, fi nding the right balance of wet and dry signal is a highly subjective process, but it is interesting to note that recordings of popular music seem to be dryer today than in previ-ous eras.
Convolution Reverb
Many digital audio workstations now feature what is known as a convolution reverb. In this type of plug-in, a variety of performance spaces are sampled and the acoustic properties of the room are stored in an audio fi le known as an im-pulse response fi le. A convolution reverb combines or convolutes a source signal with the impulse response fi le in order to apply the characteristics of the given performance space to the sound source. Convolution reverbs are typically ex-tensible so that the user can create and import his or her own impulse response fi les.
Delay
Where reverberation occurs as the result of the combination of multiple refl ec-tions, delay involves fewer discrete refl ections. Both reverb and delay can be used to simulate a sense of three-dimensional space, but delay can also be ben-efi cial in establishing the position of an instrument or voice in the soundfi eld.
Delay can also be used for doubling or echo eff ects.
One of the ways that a human can identify the position of the sound is by diff erentiating a slight delay in the arrival of the signal in one ear. You can test this by playing a mono signal through the left and right channels and adding a small amount of delay to either the left or right side. Delay on the left side will help to move the instrument to the right and vice versa. Th is phenomenon can be heard in example 10.9 at the companion Web site.
EXAMPLE 10.9
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