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Cause An Effect

Effects processing is now an integral part of a recording studio - large or small. Aaron Hallas goes back to basics in this introduction to sound treatment.

If you're new to recording and the "outboard" equipment that makes much of it possible, this introduction to audio effects will break down the mystique surrounding effects processing.

IF YOU WORK in a studio environment for any length of time, you'll discover that equalisers, compressors, limiters, and reverb devices are all essential tools for making good recordings. However, these units are seldom used for purely creative purposes - that isn't to say that they're never used creatively, but more often than not they're used to "fix" a technical problem rather than to enhance the music.

The signal processors used to add delay, chorus, flanging and phasing effects as well as pitch shifters and aural exciters, on the other hand, do take a more creative role in the studio. In fact, most of these effects were created specifically to produce special effects with music.


DELAY DEVICES COME in several forms: tape, analogue and digital - digital delay being the most commonly used these days. Essentially, delay devices repeat the input signal at regular intervals, and are often used to add depth to sounds. Most delay devices provide control over Delay Time, Delay Level or Mix, Feedback Level, Modulation Level and Modulation Speed. Figure 1 illustrates the signal path in a typical delay device.

Figure 1: The signal path in a typical delay device.

Looking at the various areas of control we have over the process, we find that Delay Time is the time interval between the input and output signals and is specified in seconds or milliseconds (1/1000 of a second). Various effects can be achieved by using different delay times. If the delay time is between about 20ms and 40ms, an effect called doubling or Automatic Double Tracking (ADT) is produced. As the delay time is increased to between 50ms and 150ms, an effect called slapback echo is produced. Delay times of 200ms or longer are simply called delay or echo.

The Delay Level determines the relative balance between the direct and delayed signals. In most cases the delayed signal is set at a much lower level than the direct signal. The Feedback Level determines the amount of output signal that is returned to the input. High feedback levels will cause the signal to repeat many times; very high feedback levels cause infinite repeats. Modulation is used to automatically vary the pitch of the delayed signal for chorus effects. It's also used to vary the delay time for flanging effects and the centre frequency for phasing effects - more on these shortly.

The first delay devices were based on tape recorders. In a tape delay system, the input signal is recorded onto magnetic tape at the record head. The delayed signal is derived from the playback head. The delay time in this type of system is determined by the distance between the record and playback heads divided by the tape speed - for example, a tape delay unit with the record and playback heads spaced two inches apart and a tape speed of 7½ inches per second (ips) would have a delay time of 266ms.

Dedicated tape delay units (like the infamous Watkins Copycat) usually have several playback heads or a moveable playback head so the delay time can be varied over a wide range. A standard tape recorder can be used to create delay effects if it has separate record and playback heads. However, it stands to reason that a tape recorder with fixed record and playback heads will have a fixed delay time. A recorder with multiple tape speeds and a variable pitch control can be more useful.

There are several drawbacks to tape delay systems. When using the feedback circuit to create multiple repeats, each successive cycle adds tape noise. The sound eventually degenerates into a sea of tape hiss. The mechanical components of the unit must be properly maintained - heads must be cleaned and demagnetized regularly, the tape transport needs to be lubricated and kept in proper alignment, and the record and playback heads may need to be replaced eventually. The magnetic tape itself also needs to be replaced on a regular basis.

Analogue delay systems use electronic circuits to replace the tape record/playback components found in tape delay systems. Although analogue delay devices require less maintenance than their tape delay counterparts, they also have several shortcomings, one of which is that most analogue devices can't reproduce the full bandwidth of music. A typical analogue delay device will have a bandwidth of around 6kHz-10kHz. To accommodate the full bandwidth of a recording, a device is currently expected to offer anything between 15kHz-25Khz, although some very expensive mixing desks use EQ circuitry with 50kHz. Also, maximum delay time is usually limited to around 500ms (½ a second), which can be pretty restricting (500ms is the duration of a crotchet at 120bpm). Since analogue delays are relatively inexpensive to manufacture, they're often used for guitar footpedal effects and the Boss DE200 became something of a classic analogue delay a few years back. Though cheap, these delays aren't the best bet for studio use due to their inherent limitations and high noise levels.

Digital delay units have become increasingly popular and are now very affordable. This is primarily due to advancements in microchip technology and low-cost memory chips. In a digital delay (often called a DDL, or Digital Delay Line), the input signal is sampled and stored in a RAM (Random Access Memory) memory buffer. The sampled sound can be replayed an infinite number of times without loss of signal quality. The maximum delay time is determined by the amount of memory available and can range from about one second to as high as eight seconds.

There are two design characteristics that will affect the sound quality of a DDL: the sampling rate or frequency and the quantisation level. The sample frequency is the rate at which the input signal is digitised into discrete samples. Sample rates are specified in 'samples per second' and typically range from 24kHz to 48kHz. To give you a point of reference, CDs are recorded at 44.1kHz. The highest recordable frequency in a digital device is a little less than one half of the sampling frequency.

The quantisation determines the dynamic range in a digital device and is measured in bits. Early DDLs used 8-bit and 12-bit quantisation, but most currently available units use 16 bits. The higher the quantisation, the greater the dynamic range. Each bit of quantisation will yield 6dB of dynamic range. This means that a 16-bit DDL with a 32kHz sampling rate will have a dynamic range of 96dB and a 15kHz bandwidth.

There is a useful method for calculating tempo-related delay times: multiplying the musical interval you require (quavers, or 8th notes, for example) by 60 and then dividing the tempo (in bpm) by the result, gives you the delay time (in seconds) you need to set on your delay line. To convert to milliseconds (the units used by most delays), simply multiply this answer by 1000.


Figure 2: The pitch of the delayed signal is modulated by an LFO in a chorus device.

WHERE REVERBS AND delays can be said to add depth to a sound, chorus adds breadth or fullness. Chorusing is produced by modulating the pitch of the delayed signal in a DDL using a low-frequency oscillator, or LFO (see Figure 2). Very short delay times of 20ms or less are used. The dry signal and the modulated signal are mixed to add fullness or to thicken a sound. If the two signals are panned to either side of the stereo field, a broadening of the sound occurs. Most units allow you to route the output signal back to the input ('regeneration') to get a deeper chorus effect. Chorus effects work very well on most instruments including guitar, bass and synthesiser sounds - however, using chorus in an attempt to make a solo singer sound like a chorus of singers is seldom convincing.


FLANGING IS SIMILAR to chorusing, but it has a more profound effect on the timbre of a sound. According to who you believe, the effect either got its name from two tape machines playing the same recording while one machine was speeded up and slowed down against the other by applying pressure to the "flange" of the reel. This would cause the phase relationship between the two recorded sounds to vary, producing the flanging effect. The other story describes the same effect but credits John Lennon with having invented the term as part of a longer, pseudo-technical term. That's rock 'n' roll.

Figure 3: The delay time in a flanger is modulated by an LFO.

In a modern flanger, the delay time is automatically varied (or "swept") from 0-20ms, again using an LFO (see Figure 3). The delayed and direct signals are mixed at the output. As the delay time changes, so does the phase relationship of the two signals. You may recall that when combined signals are in phase, they reinforce each other, causing an emphasis or resonant peak at that frequency. When they are out of phase, they will cancel each other out, causing dips in the frequency response. This is called the Comb Filter effect. When the delay time is modulated, the comb filter moves up and down the audio spectrum and the flanging effect occurs.


Figure 4: The centre frequency of several filters is modulated by an LFO in a phaser.

PHASING EFFECTS ARE similar to flanging, but more subtle, and the phaser achieves its effect in a different way. The resonant peaks and cancellations at certain frequencies are caused by combining the outputs from a series of phase shift networks with the direct signal. (Phase shift networks are filters with a variable centre frequency). As the centre frequency is varied (again, with an LFO), the phase cancellations move through the audio spectrum (see Figure 4). To get a deeper effect, more phase shift networks are needed. Studio engineers often use phasers to simulate Leslie (rotating speaker) effects.


THE THEORY BEHIND pitch shifting is quite simple. When a sound is recorded or sampled at one sample rate and played back at another rate, its pitch will be shifted up or down. Increasing the playback rate will raise the pitch; slower playback rates will lower the pitch. Pitch shifters can be used to add harmonies to a melody line or to transpose an instrument to another key, and if a small amount of pitch shift is used, the resulting sound will be very similar to chorusing.

It's worth remembering that unless your device employs specialised pitch-shifting algorithms, the length of the sample also will change. For example, if a note is pitch-shifted up one octave by playing the sample back at twice the recorded rate, the sound will be only half as long. Some dedicated pitch shifters also have intelligent algorithms that maintain the correct diatonic intervals for a given scale. For example, playing a C would yield a major third, while playing a D would result in a minor third (in the key of C major).


THE APHEX AURAL Exciter and the BBE Sonic Maximizer are two examples of devices that employ psychoacoustic processing to add clarity to a sound. As an alternative to using high-frequency boost, the psychoacoustic processor alters the harmonic content of the sound or changes the phase relationship of high and low frequencies within the sound.

The Exciter and similar devices add harmonics to the sound. When mixed at very low levels with the direct signal, the added harmonics serve to enhance the original sound. On the other hand, the Sonic Maximizer divides the audio spectrum into high- and low-frequency bands. The low frequencies are then delayed by a few milliseconds so that the high frequencies reach the listener first. This gives an apparent boost in high-frequency content to the sound.


IT'S NOT UNCOMMON to find most or all of the effects described here, plus reverb, in a single unit. These are called Multi-fx devices or Digital Signal Processors (DSP), and include units such as the Alesis Quadraverb, ART Multiverb and the old Yamaha SPX90. The creative possibilities of these units are vast; however, the advantages gained by having numerous effects in a single device are offset by inflexible internal signal routing and the lack of individual inputs and outputs for the different effects.

Most multi-fx units have a single input, but offer stereo outputs. A stereo signal is usually derived from the mono input source, although some do have true stereo signal routing capabilities. If you need true stereo processing (for use on a complete mix, for example), be sure to check the device's specifications carefully, because several currently-available devices have two inputs that are summed together inside the unit before the signal is processed. This means that you will lose the original stereo separation of your input signal. Another important factor is the nominal input and output levels and impedance. Most effects devices allow you to select between -10dB and +4dB levels. Make sure that these are set properly for your system.

Of course, the technicalities (at least, those you have to deal with) of effects processors are straightforward. What really counts is the musical uses they can be put to. Echoes - on a snare drum for example - can be synchronised to match some multiple of the tempo of a track or de-sync'd to give syncopation effects (a trick used extensively by Stuart Copeland with the Police). A pitch shifter, alternatively, can be used to correct the pitch of a drum loop that has had its playback speed altered. And that's where you come in...

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Music Technology - Copyright: Music Maker Publications (UK), Future Publishing.


Music Technology - May 1991

Donated by: Mike Gorman, Ian Sanderson

Scanned by: Mike Gorman

Feature by Aaron Hallas

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