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22. Live Audio Effect Reference

Live comes with a selection of custom-designed, built-in audio effects. The Working with Instruments and Effects chapter (see Chapter 17) explains the basics of using effects in Live.

22.1 Amp

Amp.png
The Amp Effect.

Amp is an effect that emulates the sound and character of seven classic guitar amplifiers. Developed in collaboration with Softube, Amp uses physical modelling technology to provide a range of authentic and usable amplifier tones, with a simple and consistent set of controls.

The full version of Amp is not included with the standard version of Live, but is bundled with the Cabinet effect (see 22.5) as a special feature available for purchase separately.

There are seven amp models to choose from:

  • Clean is based on the ”Brilliant” channel of a classic amp from the ’60s. This amp was widely used by guitarists of the British Invasion.
  • Boost is based on the ”Tremolo” channel of the same amp, and is great for edgy rock riffs.
  • Blues is based on a ’70s-era guitar amp with a bright character. This classic amp is popular with country, rock and blues guitarists.
  • Rock is modeled after a classic 45 watt amp from the ’60s. This is perhaps the best known rock amp of all time.
  • Lead is based on the ”Modern” channel of a high-gain amp popular with metal guitarists.
  • Heavy is based on the ”Vintage” channel of the same amp and is also ideal for metal and grunge sounds.
  • Bass is modeled after a rare PA from the ’70s which has become popular with bass players due to its strong low end and ”fuzz” at high volumes.

Although the real-world versions of these amplifiers all have unique parameters, Live’s Amp effect uses the same set of controls for each model. This makes it very easy to quickly change the overall character of your sound without having to make numerous adjustments.

Gain adjusts the level of input to the preamplifier, while Volume adjust the output stage of the power amplifier. Although Gain and Volume work together to determine Amp’s overall level, Gain is the primary control for the distortion amount. Higher Gain settings result in a more distorted sound. When using the Blues, Heavy and Bass models, high Volume levels can also add considerable distortion.

The Bass, Middle and Treble knobs are EQ controls that adjust the timbre of the sound. As on a real-world amplifier, Amp’s EQ parameters interact with each other — and with the rest of Amp’s parameters — in non-linear and sometimes unpredictable ways. For example, increasing EQ levels can, in some cases, also increase the amount of distortion.

Presence is an additional tone control for mid/high frequencies in the power amp stage. Its influence on the sound varies considerably depending on the amp model used but can add (or subtract) ”edge” or ”crispness.”

The Output switch toggles between mono and stereo (Dual) processing. Note that in Dual mode, Amp uses twice as much CPU.

The Dry/Wet control adjusts the balance between the processed and dry signals.

22.1.1 Amp Tips

Because Amp is modeled on real-world analog devices, its behavior can sometimes be difficult to predict. Here are some tips on getting the most out of Amp:

Amps and Cabinets

Guitar amps are designed to be used with accompanying speaker cabinets. For this reason, Amp comes with a companion effect called Cabinet (see 22.5) which is designed to be used after Amp in a device chain. If you’re looking for authenticity, we recommend this signal flow. But you can also achieve interesting and exotic sounds by using Amp and Cabinet independently.

Electricity

The various circuits in guitar amps work with a continuous and fixed amount of electricity. For this reason, turning up a particular parameter may inadvertently decrease the amount of available energy somewhere else in the amp. This is particularly noticeable in the EQ controls. For example, turning up Treble can reduce the level of bass and midrange frequencies. You may find that you need to carefully adjust a number of seemingly unrelated parameters to get the results you want.

More than guitars

While Amp and Cabinet sound great with guitars, you can get very interesting results by feeding them with drums, synthesizers or other sound sources. For example, try using Amp with Operator (see 24.6) or Analog (see 24.1) to add analog grit to your digital sounds.

22.2 Auto Filter

AutoFilter.png
The Auto Filter Effect.

The Auto Filter effect provides classic analog filter emulation. It can be modulated by an envelope follower and/or an LFO to create moving filter effects. The envelope follower can track either the filtered signal or an external sidechain source.

There are four different filter types: lowpass, highpass, bandpass and notch. For each type, the X-Y controller adjusts frequency (to adjust, click and drag on the X-axis) and Q (also called resonance; to adjust, click and drag on the Y-axis). You can also click on the Freq and Q numeric displays and type in exact values.

Low Q values create a broad filter curve, while higher values introduce a narrow, resonant peak to the sound. With bandpass filtering, Q sets the bandwidth of the passed signal.

The Quantize Beat control applies quantized modulation to the filter frequency. With Quantize Beat off, frequency modulation follows the control source. Turning this feature on updates the filter modulation rhythmically with ”stepped” changes that track the master tempo. The numbered buttons represent 16th notes, so, for example, selecting ”4” as a beat value produces a modulation change once per beat.

The Envelope section controls how the envelope modulation affects the filter frequency. The Amount control defines the extent to which the envelope affects the filter frequency, while the Attack control sets how the envelope responds to rising input signals. Low Attack values cause a fast response to input levels; high values integrate any changes gradually, creating a looser, slower response. Think of it as adding inertia to the response.

Lower Release values cause the envelope to respond more quickly to falling input signals. Higher values extend the envelope’s decay.

Normally, the signal being filtered and the input source that triggers the envelope follower are the same signal. But by using sidechaining , it is possible to filter a signal based on the envelope of another signal. To access the Sidechain parameters, unfold the Auto Filter window by toggling the 25691.png button in its title bar.

Enabling this section with the Sidechain button allows you to select another track from the choosers below. This causes the selected track’s signal to trigger the filter’s envelope follower, instead of the signal that is actually being filtered.

The Gain knob adjusts the level of the external sidechain’s input, while the Dry/Wet knob allows you to use a combination of sidechain and original signal as the envelope follower’s trigger. With Dry/Wet at 100%, the envelope follower tracks the sidechain source exclusively. At 0%, the sidechain is effectively bypassed. Note that increasing the gain does not increase the volume of the source signal in the mix. The sidechain audio is only a trigger for the envelope follower and is never actually heard.

The Auto Filter also contains a Low Frequency Oscillator to modulate filter frequency in a periodic fashion. The respective Amount control sets how much the LFO affects the filter. This can be used in conjunction with or instead of the envelope follower.

The Rate control specifies the LFO speed. It can be set in terms of hertz, or synced to the song tempo, allowing for controlled rhythmic filtering.

Available LFO waveform shapes are sine (creates smooth modulations with rounded peaks and valleys), square, triangle, sawtooth up, sawtooth down, and sample and hold (generates random positive and negative modulation values) in mono and stereo.

There are two LFOs, one for each stereo channel. The Phase and Offset controls define the relationship between these two LFOs.

Phase keeps both LFOs at the same frequency, but can set the two LFO waveforms ”out of phase” with each other, creating stereo movement. Set to ”180”, the LFO outputs are 180 degrees apart, so that when one LFO reaches its peak, the other is at its minimum.

Spin detunes the two LFO speeds relative to each other. Each stereo channel is modulated at a different frequency, as determined by the Spin amount.

For sample and hold, the Phase and Spin controls are not relevant and do not affect the sound. Instead, the Auto Filter offers two kinds of sample and hold: The upper sample and hold type available in the chooser provides independent random modulation generators for the left and right channels (stereo), while the lower one modulates both channels with the same signal (mono).

22.3 Auto Pan

AutoPan.png
The Auto Pan Effect.

(Note: the Auto Pan effect is not available in the Lite Edition.)

Auto Pan offers LFO-driven manipulation of amplitude and panning for creating automatic panning, tremolo and amplitude modulation, and beat-synchronized chopping effects.

Auto Pan’s LFOs modulate the amplitude of the left and right stereo channels with sine, triangle, sawtooth down or random waveforms.

The Shape control pushes the waveform to its upper and lower limits, ”hardening” its shape. The waveform can be set to ”Normal” or ”Invert” (use ”Invert” to, for example, create the saw up waveform from the saw down waveform).

LFO speed is controlled with the Rate control, which can be set in terms of hertz. Rate can also be synced to the song tempo.

Though both LFOs run at the same frequency, the Phase control lends the sound stereo movement by offsetting their waveforms relative to each other. Set this to ”180”, and the LFOs will be perfectly out of phase (180 degrees apart), so that when one reaches its peak, the other is at its minimum. Phase is particularly effective for creating vibrato effects.

The Offset control shifts the start point of each LFO along its waveform.

The device’s influence on incoming signals is set with the Amount control.

22.4 Beat Repeat

BeatRepeat.png
The Beat Repeat Effect.

Beat Repeat allows for the creation of controlled or randomized repetitions of an incoming signal.

The Interval control defines how often Beat Repeat captures new material and begins repeating it. Interval is synced to and set in terms of the song tempo, with values ranging from ”1/32” to ”4 Bars.” The Offset control shifts the point defined by Interval forward in time. If Interval is set to ”1 Bar,” for example, and Offset to ”8/16”, material will be captured for repetition once per bar on the third beat (i.e., halfway, or eight-sixteenths of the way, through the bar).

You can add randomness to the process using the Chance control, which defines the likelihood of repetitions actually taking place when Interval and Offset ”ask” for them. If Chance is set to 100 percent, repetitions will always take place at the given Interval/Offset time; if set to zero, there will be no repetitions.

Gate defines the total length of all repetitions in sixteenth notes. If Gate is set to ”4/16”, the repetitions will occur over the period of one beat, starting at the position defined by Interval and Offset.

Activating the Repeat button bypasses all of the above controls, immediately capturing material and repeating it until deactivated.

The Grid control defines the grid size — the size of each repeated slice. If set to ”1/16”, a slice the size of one sixteenth note will be captured and repeated for the given Gate length (or until Repeat is deactivated). Large grid values create rhythmic loops, while small values create sonic artifacts. The No Triplets button sets grid division as binary.

Grid size can be changed randomly using the Variation control. If Variation is set to ”0”, grid size is fixed. But when Variation is set to higher values, the grid fluctuates considerably around the set Grid value. Variation has several different modes, available in the chooser below: Trigger creates variations of the grid when repetitions are triggered; 1/4, 1/8 and 1/16 trigger variations in regular intervals; and Auto forces Beat Repeat to apply a new random variation after each repetition — the most complex form of grid variation in Beat Repeat (especially if triplets are also allowed).

Beat Repeat’s repetitions can be pitched down for special sonic effects. Pitch is adjusted through resampling in Beat Repeat, lengthening segments to pitch them down without again compressing them to adjust for the length change. This means that the rhythmical structure can become quite ambiguous with higher Pitch values. The Pitch Decay control tapers the pitch curve, making each repeated slice play lower than the previous one. Warning: This is the most obscure parameter of Beat Repeat.

Beat Repeat includes a combined lowpass and highpass filter for defining the passed frequency range of the device. You can turn the filter on and off, and set the center frequency and width of the passed frequency band, using the respective controls.

The original signal (which was received at Beat Repeat’s input) is mixed with Beat Repeat’s repetitions according to one of three mix modes: Mix allows the original signal to pass through the device and have repetitions added to it; Insert mutes the original signal when repetitions are playing but passes it otherwise; and Gate passes only the repetitions, never passing the original signal. Gate mode is especially useful when the effect is housed in a return track.

You can set the output level of the device using the Volume control, and apply Decay to create gradually fading repetitions.

22.5 Cabinet

Cabinet.png
The Cabinet Effect.

Cabinet is an effect that emulates the sound of five classic guitar cabinets. Developed in collaboration with Softube, Cabinet uses physical modelling technology to provide a range of authentic sounds, with optimized mics and mic positioning.

The full version of Cabinet is not included with the standard version of Live, but is bundled with the Amp effect (see 22.1) as a special feature available for purchase separately.

The Speaker chooser allows you to select from a variety of speaker sizes and combinations. The chooser’s entries indicate the number of speakers and the speaker size in inches. For example, ”4x12” means four 12-inch speakers. In the real world, more and larger speakers generally means higher volumes.

The Microphone chooser changes the position of the virtual microphone in relation to the speaker cabinet. Near On-Axis micing results in a bright, focused sound, while Near Off-Axis is more resonant and a bit less bright. Choose the Far position for a balanced sound that also has some characteristics of the virtual ”room.”

25756.png
Cabinet Mic Positions.

The switch below the Microphone chooser toggles between a Dynamic and Condenser mic. Dynamic mics are a bit grittier and commonly used when close-micing guitar cabinets because they are capable of handling much higher volumes. Condenser mics are more accurate, and are commonly used for micing from a distance. Of course, Cabinet’s virtual condenser mic won’t be damaged by high volume levels, so feel free to experiment.

The Output switch toggles between mono and stereo (Dual) processing. Note that in Dual mode, Cabinet uses twice as much CPU.

The Dry/Wet control adjusts the balance between the processed and dry signals.

22.5.1 Cabinet Tips

Here are some tips for using Cabinet:

Amps and Cabinets

Guitar cabinets are normally fed by guitar amps. For this reason, Cabinet is paired with Amp (see 22.1), and the two are normally used together. But you can also achieve interesting and exotic sounds by using Amp and Cabinet separately.

Multiple mics

A common studio technique is to use multiple mics on a single cabinet, and then adjust the balance during mixing. This is easy to do by using Live’s Audio Effect Racks (see Chapter 18). Try this:

  • configure one instance of Cabinet as you like
  • put the Cabinet into an Audio Effect Rack
  • duplicate the Rack chain that contains the original Cabinet as many times as you like
  • in the additional chains, choose a different Microphone setting and/or mic type
  • adjust the relative volumes of the Rack’s chains in the Rack’s mixer

22.6 Chorus

Chorus.png
The Chorus Effect.

The Chorus effect uses two parallel time-modulated delays to create chorus (thickening) and flanging effects.

Each delay has its own delay time control, calibrated in milliseconds. Delay 1 has a highpass filter that can remove low frequencies from the delayed signal. Greater highpass values let only very high frequencies pass through to Delay 1.

Delay 2 can switch among three different modes. When off, only Delay 1 is audible. In Fix Mode, only Delay 1’s delay time will be modulated. When Mod is activated, Delay 2 will receive the same modulation as Delay 1.

To set both delay lines to Delay 1’s delay time, turn on the link button (”=”). This is especially useful if you want to change both delays with a single gesture.

The Modulation X-Y controller can impart ”motion” to the sounds. To change the modulation rate for the delay times, click and drag along the horizontal axis. To change the amount of modulation, click and drag along the vertical axis.

You can also make changes by entering parameter values in the Amount and Rate fields below the X-Y controller. The Amount value is in milliseconds, while the modulation frequency rate is in Hertz.

Clicking the *20 switch multiplies the modulation frequency by 20 to achieve more extreme sounds.

The Feedback control determines how much of the output signal feeds back into the input, while the Polarity switch sets (surprise!) the polarity. Polarity changes have the most effect with high amounts of feedback and short delay times.

The Dry/Wet control adjusts the balance between the processed and dry signals. Set it to 100 percent when using Chorus in a return track.

Enabling the Crisp option via the right-click (PC) / CTRL - click (Mac) context menu can improve the sound quality, particularly at higher frequencies. This is enabled by default, except when loading Sets that use Chorus and were made in earlier versions of Live.

22.7 Compressor

Compressor.png
The Compressor Effect.

A compressor reduces gain for signals above a user-settable threshold. Compression reduces the levels of peaks, opening up more headroom and allowing the overall signal level to be turned up. This gives the signal a higher average level, resulting in a sound that is subjectively louder and ”punchier” than an uncompressed signal.

A compressor’s two most important parameters are the Threshold and the compression Ratio.

The Threshold slider sets where compression begins. Signals above the threshold are attenuated by an amount specified by the Ratio parameter, which sets the ratio between the input and output signal. For example, with a compression ratio of 3, if a signal above the threshold increases by 3 dB, the compressor output will increase by only 1 dB. If a signal above the threshold increases by 6 dB, then the output will increase by only 2 dB. A ratio of 1 means no compression, regardless of the threshold.

The orange Gain Reduction meter shows how much the gain is being reduced at any given moment. The more reduction, the more audible the effect; a gain reduction above 6 dB or so might produce the desired loudness, but significantly alters the sound and is easily capable of destroying its dynamic structure. This is something that cannot be undone in later production steps. Keep this in mind especially when using a compressor, limiter or sound loudness-maximizing tool in the master channel. Less is often more here.

Because compression reduces the volume of loud signals and opens up headroom, you can use the Output (Out) control so that the peaks once again hit the maximum available headroom. The Output meter shows the output signal’s level. Enabling the Makeup button automatically compensates the output level if the threshold and ratio settings change.

Dry/Wet adjusts the balance between the compressed and uncompressed signals. At 100%, only the compressed signal is heard, while at 0%, the device is effectively bypassed.

The Knee control adjusts how gradually or abruptly compression occurs as the threshold is approached. With a setting of 0 dB, no compression is applied to signals below the threshold, and full compression is applied to any signal at or above the threshold. With very high ratios, this so-called ”hard knee” behavior can sound harsh. With higher (or ”soft”) knee values, the compressor begins compressing gradually as the threshold is approached. For example, with a 10 dB knee and a -20 dB threshold, subtle compression will begin at -30 dB and increase so that signals at -10 dB will be fully compressed.

Compressor’s display can be switched between several modes via switches in the bottom corners of the display:

  • The Collapsed view shows only the essential controls.
CompressorCollapsed.png
Compressor’s Collapsed View
  • The Transfer Curve shows the input level on the horizontal axis and output level vertically. This view is useful for setting the Knee parameter, which is visible as a pair of dotted lines around the threshold.
CompressorTransferCurve.png
Compressor’s Transfer Curve.
  • The Activity view shows the level of the input signal in light grey. In this mode, the GR and Output switches toggle between showing the amount of gain reduction in orange or the output level in a darker grey. These views are useful for visualizing what’s happening to the signal over time.
Compressor.png
Compression Activity Display, Showing Input and Output.

The Attack and Release controls are essential parameters for controlling the response time of Compressor by defining how fast it reacts to input-level changes.

Attack defines how long it takes to reach maximum compression once a signal exceeds the threshold, while Release sets how long it takes for the compressor to return to normal operation after the signal falls below the threshold. With Auto Release enabled, the release time will adjust automatically based on the incoming audio.

A slight amount of attack time (10-50 ms) allows peaks to come through unprocessed, which helps preserve dynamics by accentuating the initial portion of the signal. If these peaks cause overloads, you can try shortening the attack time, but extremely short times take the ”life” out of the signal, and may lead to a slight “buzziness” caused by distortion. Short release times can cause ”pumping” as the compressor tries to figure out whether to compress or not; while generally considered an undesirable effect, some engineers use it on full drum kits to give unusual ”sucking” effects. Careful adjustment of attack and release times is essential when it comes to compression of rhythmical sources. If you are not used to working with compressors, play a drum loop and spend some time adjusting Attack, Release, Threshold and Gain. It can be very exciting!

A compressor can only react to an input signal once it occurs. Since it also needs to apply an attack/release envelope, the compression is always a bit too late. A digital compressor can solve this problem by simply delaying the input signal a little bit. Compressor offers three different Lookahead times: zero ms, one ms and ten ms. The results may sound pretty different depending on this setting.

Compressor can be switched between three basic modes of operation. With Peak selected, Compressor reacts to short peaks within a signal. This mode is more aggressive and precise, and so works well for limiting tasks where you need to ensure that there are absolutely no signals over the set threshold. RMS mode causes Compressor to be less sensitive to very short peaks and compress only when the incoming level has exceeded the threshold for a slightly longer time. RMS is closer to how people actually perceive loudness and is usually considered more ”musical.”

In Expand mode, the Ratio can also be set to values below 1. In this state, Compressor acts as an upward expander, and will increase the gain when signals exceed the threshold. (For more information about the various types of dynamics processing, see the Multiband Dynamics chapter (see 22.22).)

In addition to these modes, Compressor can be switched between two envelope follower shapes that offer further options for how the device measures and responds to signal levels. In linear (Lin) mode, the speed of the compression response is determined entirely by the Attack and Release values. In logarithmic (Log) mode, sharply compressed peaks will have a faster release time than less compressed material. This can result in smoother and less noticeable compression than Lin mode. Note that the Lin/Log switch is not visible in Compressor’s collapsed view.

22.7.1 Sidechain Parameters

CompressorSidechain.png
The Compressor Device With Sidechain Section.

Normally, the signal being compressed and the input source that triggers the compressor are the same signal. But by using sidechaining , it is possible to compress a signal based on the level of another signal or a specific frequency component. To access the Sidechain parameters, unfold the Compressor window by toggling the 25877.png button in its title bar.

The sidechain parameters are divided into two sections. On the left are the external controls. Enabling this section with the Sidechain button allows you to select any of Live’s internal routing points (see 14.6.1) from the choosers below. This causes the selected source to act as the compressor’s trigger, instead of the signal that is actually being compressed.

The Gain knob adjusts the level of the external sidechain’s input, while the Dry/Wet knob allows you to use a combination of sidechain and original signal as the compressor’s trigger. With Dry/Wet at 100%, the compressor is triggered entirely by the sidechain source. At 0%, the sidechain is effectively bypassed. Note that increasing the gain does not increase the volume of the source signal in the mix. The sidechain audio is only a trigger for the compressor and is never actually heard.

Note that automatic Makeup is not available when using external sidechain.

On the right of the external section are the controls for the sidechain EQ. Enabling this section causes the compressor to be triggered by a specific band of frequencies, instead of a complete signal. These can either be frequencies in the compressed signal or, by using the EQ in conjunction with an external sidechain, frequencies in another track’s audio.

The headphones button between the external and EQ sections allows you to listen to only the sidechain input, bypassing the compressor’s output. Since the sidechain audio isn’t fed to the output, and is only a trigger for the compressor, this temporary listening option can make it much easier to set sidechain parameters and hear what’s actually making the compressor work.

22.7.2 Compression Tips

This section presents some tips for using Compressor effectively, particularly with the sidechain options.

Mixing a Voiceover

Sidechaining is commonly used for so-called ”ducking” effects. For example, imagine that you have one track containing a voiceover and another track containing background music. Since you want the voiceover to always be the loudest source in the mix, the background music must get out of the way every time the narrator is speaking. To do this automatically, insert a Compressor on the music track, but select the narration track’s output as the external sidechain source.

Sidechaining in Dance Music

Sidechaining/ducking is a dance music producer’s secret weapon because it can help to ensure that basslines (or even whole mixes) always make room for the kick drum. By inserting a compressor on the bass (or master) track and using the kick drum’s track as the sidechain input, you can help to control problematic low frequencies that might interfere with the kick drum’s attack.

Using the sidechain EQ in conjunction with this technique can create ducking effects even if you only have a mixed drum track to work with (as opposed to an isolated kick drum). In this case, insert the Compressor on the track you want to duck. Then choose the drum track as the external sidechain source. Then enable the sidechain EQ and select the lowpass filter. By carefully adjusting the Frequency and Q settings, you should be able to isolate the kick drum from the rest of the drum mix. Using the sidechain listen mode can help you tune the EQ until you find settings you’re happy with.

Note that mastery of these techniques may result in a noticeable increase in drink tickets, remix offers and dates.

22.7.3 Upgrading from Legacy Mode

Compressor’s internal algorithms were updated in Live 9, in collaboration with Dr. Joshua D. Reiss of the Centre for Digital Music, Queen Mary University of London.

Live Sets that use Compressor and which were made in earlier versions of Live will show an Upgrade button in the title bar of each instance of Compressor when loading the Set in Live 9. Press the Upgrade button in order to upgrade that Compressor instance to the latest, improved algorithms. Note that this may cause your Set to sound different.

22.8 Corpus

Corpus.png
The Corpus Effect.

Corpus is an effect that simulates the acoustic characteristics of seven types of resonant objects. Developed in collaboration with Applied Acoustics Systems, Corpus uses physical modelling technology to provide a wide range of parameters and modulation options.

The full version of Corpus is not included with the standard version of Live, but is bundled with the Collision instrument (see 24.2) as a special feature available for purchase separately.

The frequency and/or decay rate of the resonance can be MIDI modulated, by enabling the Frequency and/or Off Decay switches in the Sidechain section. Toggle the 25913.png button in Corpus’s title bar to access Sidechain parameters. The MIDI From choosers allow you to select the MIDI track and tapping point from which to receive MIDI note information.

With Frequency enabled, the tuning of the resonance is determined by the incoming MIDI note. If multiple notes are held simultaneously, the Last/Low switch determines whether the last or the lowest note will have priority. The Transpose and Fine knobs allow for coarse and fine offset of the MIDI-modulated tuning. PB Range sets the range in semitones of pitch bend modulation.

With Frequency disabled, the Tune control adjusts the base frequency of the resonance in Hertz. The corresponding MIDI note number and fine tuning offset in cents is displayed below.

Enabling Off Decay causes MIDI note off messages to mute the resonance. The slider below the switch determines the extent to which MIDI note off messages mute the resonance. At 0%, note offs are ignored, and the decay time is based only on the value of the Decay parameter, which is located under the Resonance Type selector. This is similar to how real-world mallet instruments such as a marimbas and glockenspiels behave. At 100%, the resonance is muted immediately at note off, regardless of the Decay time.

You can hide or show the Sidechain parameters by toggling the 25925.png button in Corpus’s title bar. This button will light up if the sidechain is active.

Corpus contains a Low Frequency Oscillator (LFO) to modulate the resonant frequency. The Amount control sets how much the LFO affects the frequency.

The Rate control specifies the LFO speed. It can be set in terms of Hertz, or synced to the song tempo, allowing for controlled rhythmic modulation.

Available LFO waveform shapes are sine (creates smooth modulations with rounded peaks and valleys), square, triangle, sawtooth up, sawtooth down and two types of noise (stepped and smooth).

Although only one set of LFO controls is visible, there are actually two LFOs, one for each stereo channel. The Phase and Spin controls define the relationship between these two LFOs.

Phase (available only when the LFOs are synced to song tempo) keeps both LFOs at the same frequency, but can set the two LFO waveforms ”out of phase” with each other, creating stereo movement. Set to ”180”, the LFO outputs are 180 degrees apart, so that when one LFO reaches its peak, the other is at its minimum. With Phase set to ”360” or ”0” the two LFOs run in sync.

Spin (only available when the LFOs are in Hertz mode) detunes the two LFO speeds relative to each other. Each stereo channel is modulated at a different frequency, as determined by the Spin amount.

For the noise waveforms, the Phase and Spin controls are not relevant and do not affect the sound.

Spread detunes the two resonators in relation to each other. Positive values raise the pitch of the left resonator while lowering the pitch of the right one, while negative values do the opposite. At 0%, the resonators are tuned the same.

The resonance type chooser allows you to select from seven types of physically modeled resonant objects:

  • Beam simulates the resonance properties of beams of different materials and sizes.
  • Marimba , a specialized variant of the Beam model, reproduces the characteristic tuning of marimba bar overtones which are produced as a result of the deep arch-cut of the bars.
  • String simulates the sound produced by strings of different materials and sizes.
  • Membrane is a model of a rectangular membrane (such as a drum head) with a variable size and construction.
  • Plate simulates sound production by a rectangular plate (a flat surface) of different materials and sizes.
  • Pipe simulates a cylindrical tube that is fully open at one end and has a variable opening at the other (adjusted with the Opening parameter.)
  • Tube simulates a cylindrical tube that is fully open at both ends.

The resonator quality chooser controls the tradeoff between the sound quality of the resonators and performance by reducing the number of overtones that are calculated. ”Basic” uses minimal CPU resources, while ”Full” creates more sophisticated resonances. This parameter is not used with the Pipe or Tube resonators.

The Decay knob adjusts the amount of internal damping in the resonator, and thus the decay time.

The Material knob adjusts the variation of the damping at different frequencies. At lower values, low frequency components decay slower than high frequency components (which simulates objects made of wood, nylon or rubber). At higher values, high frequency components decay slower (which simulates objects made of glass or metal). This parameter is not used with the Pipe or Tube resonators.

The Radius parameter is only available for the Pipe and Tube resonators. Radius adjusts the radius of the pipe or tube. As the radius increases, the decay time and high frequency sustain both increase. At very large sizes, the fundamental pitch of the resonator also changes.

The Decay and Material/Radius parameters can also be controlled with the X-Y controller.

Ratio is only available for the Membrane and Plate resonators, and adjusts the ratio of the object’s size along its x and y axes.

The Brightness control adjusts the amplitude of various frequency components. At higher values, higher frequencies are louder. This parameter is not used with the Pipe or Tube resonators.

Inharm. (Inharmonics) adjusts the pitch of the resonator’s harmonics. At negative values, frequencies are compressed, increasing the amount of lower partials. At positive values, frequencies are stretched, increasing the amount of upper partials. This parameter is not used with the Pipe or Tube resonators.

Opening, which is only available for the Pipe resonator, scales between an open and closed pipe. At 0%, the pipe is fully closed on one side, while at 100% the pipe is open at both ends.

The Listening L and R controls adjust the location on the left and right resonator where the vibrations are measured. At 0%, the resonance is monitored at the object’s center. Higher values move the listening point closer to the edge. These parameters are not used with the Pipe or Tube resonators, which are always measured in the middle of their permanently open end.

The Hit knob adjusts the location on the resonator at which the object is struck or otherwise activated. At 0%, the object is hit at its center. Higher values move the activation point closer to the edge. This parameter is not used with the Pipe or Tube resonators.

The processed signal is fed through a lowpass and highpass filter that can be controlled with an X-Y controller. To define the filter bandwidth, click and drag on the vertical axis. To set the position of the frequency band, click and drag on the horizontal axis. The filter can be toggled on or off with the Filter switch.

Width adjusts the stereo mix between the left and right resonators. At 0%, both resonators are fed equally to each side, resulting in mono output. At 100%, each resonator is sent exclusively to one channel.

Bleed mixes a portion of the unprocessed signal with the resonated signal. At higher values, more of the original signal is applied. This is useful for restoring high frequencies, which can often be damped when the tuning or quality are set to low values. This parameter is unavailable with the Pipe or Tube resonators.

Gain boosts or attenuates the level of the processed signal, while the Dry/Wet control adjusts the balance between the dry input signal and the signal sent to Corpus’s processing. Turning Dry/Wet down will not cut resonances that are currently sounding, but rather stop new input signals from being processed.

Corpus contains a built-in limiter that automatically activates when the audio level is too high. This is indicated by the LED in the upper-right corner of Corpus’s display.

22.9 Dynamic Tube

DynamicTube.png
The Dynamic Tube Effect.

(Note: the Dynamic Tube effect is not available in the Intro and Lite Editions.)

The Dynamic Tube effect infuses sounds with the peculiarities of tube saturation. An integrated envelope follower generates dynamic tonal variations related to the level of the input signal.

Three tube models, A, B and C, provide a range of distortion characteristics known from real amplifier tubes. Tube A does not produce distortions if Bias is set low, but will kick in whenever the input signal exceeds a certain threshold, creating bright harmonics. Tube C is a very poor tube amp that produces distortions all the time. The qualities of Tube B lie somewhere between these two extremes.

The Tone control sets the spectral distribution of the distortions, directing them into the higher registers, or through the midrange and deeper.

The Drive control determines how much signal reaches the tube; greater Drive yields a dirtier output. The intensity of the tube is controlled by the Bias dial, which pushes the signal into the celebrated realms of nonlinear distortion. With very high amounts of Bias, the signal will really start to break apart.

The Bias parameter can be positively or negatively modulated by an envelope follower, which is controlled with the Envelope knob. The more deeply the envelope is applied, the more the Bias point will be influenced by the level of the input signal. Negative Envelope values create expansion effects by reducing distortion on loud signals, while positive values will make loud sounds dirtier.

Attack and Release are envelope characteristics that define how quickly the envelope reacts to volume changes in the input signal. Together, they shape the dynamic nature of the distortions. Note that if Envelope is set to zero, they will have no effect.

Cut or boost the device’s final signal level with the Output dial.

Aliasing can be reduced by enabling Hi-Quality mode, which can be accessed via the right-click (PC) / CTRL - click (Mac) context menu. This improves the sound quality, particularly with high frequency signals, but there is a slight increase in CPU usage.

22.10 EQ Eight

EQEight.png
The EQ Eight Effect.

(Note: the EQ Eight effect is not available in the Intro and Lite Editions.)

The EQ Eight effect is an equalizer featuring up to eight parametric filters per input channel, useful for changing a sound’s timbre.

The input signal can be processed using one of three modes: Stereo, L/R and M/S. Stereo mode uses a single curve to filter both channels of a stereo input equally. L/R mode provides an independently adjustable filter curve for the left and right channels of a stereo input; M/S mode (Mid/Side) provides the same functionality for signals that have been recorded using M/S encoding. In all modes, the frequency spectrum of the output is displayed behind the filter curves when the Analyze switch is on.

When using the L/R and M/S modes, both curves are displayed simultaneously for reference, although only the active channel is editable. The Edit switch indicates the active channel, and is used to toggle between the two curves.

Each filter has a chooser that allows you to switch between eight responses. From top to bottom in the choosers, these are:

  • 48 or 12 dB/octave Low cut (cuts frequencies below the specified frequency);
  • Low shelf (boosts or cuts frequencies lower than the specified frequency);
  • Bell curve (boosts or cuts over a range of frequencies);
  • Notch (sharply cuts frequencies within a narrow range);
  • High shelf (boosts or cuts frequencies higher than the specified frequency);
  • 12 or 48 dB/octave High cut (cuts frequencies above the specified frequency).

Each filter band can be turned on or off independently with an activator switch below the chooser. Turn off bands that are not in use to save CPU power. To achieve really drastic filtering effects, assign the same parameters to two or more filters.

To edit the filter curve, click and drag on the filter dots in the display. Drag-enclose multiple filter dots to adjust them simultaneously, either with the mouse or with your computer keyboard’s arrow keys. Horizontal movement changes the filter frequency, while vertical movement adjusts the filter band’s gain. To adjust the filter Q (also called resonance or bandwidth), hold down the ALT (PC) / ALT (Mac) modifier while dragging the mouse. Note that the gain cannot be adjusted for the low cut, notch and high cut filters. In these modes, vertical dragging adjusts the filter Q.

To get an even better view, you can toggle the location of the display between the device chain and Live’s main window by clicking the 25973.png button in EQ Eight’s title bar. When using this expanded view, all eight filters can be edited simultaneously in the Device View.

EQEightExpandedView.png
EQ Eight’s Controls With the Display Expanded.

By default, EQ Eight’s output spectrum is shown in the display. If you would prefer to work entirely “by ear,” you can turn off the Analyze button to disable the spectrum view.

With Adaptive Q enabled, the Q amount increases as the amount of boost or cut increases. This results in a more consistent output volume and is based on the behavior of classic analog EQs.

To temporarily solo a single filter, enable Audition mode via the headphone icon. In Audition mode, clicking and holding on a filter dot allows you to hear only that filter’s effect on the output.

You can also select a band for editing by clicking near its number and then edit parameter values with the Freq, Gain and Q dials (and/or type values into the number fields below each dial).

As boosting will increase levels and cutting will decrease levels, use the global Gain slider to optimize the output level for maximum level consistent with minimum distortion.

The Scale field will adjust the gain of all filters that support gain (all except low cut, notch and high cut).

Context Menu Options

Several of EQ Eight’s controls are only available via the right-click (PC) / CTRL - click (Mac) context menu. These include:

  • Oversampling - enabling this option causes EQ Eight to internally process two times the current sample rate, which allows for smoother filter behavior when adjusting high frequencies. There is a slight increase in CPU usage with Oversampling enabled.
  • Shelf Scaling Legacy Mode - As of Live 9, the shape of EQ Eight’s shelf filters has been improved. Live Sets that use EQ Eight and which were made prior to Live 9 may sound slightly different. To ensure that older Sets sound exactly the same, the Shelf Scaling Legacy Mode option will be enabled by default when loading an old Set that uses EQ Eight. You can disable this via a right-click (PC) / CTRL - click (Mac) context menu entry in EQ Eight’s title bar.

Note: as of Live 9, the “Hi-Quality” right-click (PC) / CTRL - click (Mac) context menu option has been removed. EQ Eight now always operates in this mode.

22.11 EQ Three

EQThree.png
The EQ Three Effect.

If you have ever used a good DJ mixer you will know what this is: An EQ that allows you to adjust the level of low, mid and high frequencies independently.

Each band can be adjusted from -infinite dB to +6 dB using the gain controls. This means that you can completely remove, for example, the bass drum or bassline of a track, while leaving the other frequencies untouched.

You can also turn on or off each band using the On/Off buttons located under the gain controls. These buttons are especially handy if assigned to computer keys.

EQ Three gives you visual confirmation of the presence of a signal in each frequency band using three LEDs. Even if a band is turned off, you can tell if there is something going on in it. The internal threshold for the LEDs is set to -24 dB.

The frequency range of each band is defined via two crossover controls: FreqLo and FreqHi. If FreqLo is set to 500 Hz and FreqHi to 2000 Hz, then the low band goes from 0 Hz to 500 Hz, the mid band from 500 Hz to 2000 Hz and the high band from 2000 Hz up to whatever your soundcard or sample rate supports.

A very important control is the 24 dB/48 dB switch. It defines how sharp the filters are cutting the signal at the crossover frequency. The higher setting results in more drastic filtering, but needs more CPU.

Note: The filters in this device are optimized to sound more like a good, powerful analog filter cascade than a clean digital filter. The 48 dB Mode especially does not provide a perfect linear transfer quality, resulting in a slight coloration of the input signal even if all controls are set to 0.00 dB. This is typical behavior for this kind of filter, and is part of EQ Three’s unique sound. If you need a more linear behavior choose 24 dB Mode or use the EQ Eight.

22.12 Erosion

Erosion.png
The Erosion Effect.

The Erosion effect degrades the input signal by modulating a short delay with filtered noise or a sine wave. This adds noisy artifacts or aliasing/downsampling-like distortions that sound very ”digital.”

To change the sine wave frequency or noise band center frequency, click and drag along the X-axis in the X-Y field. The Y-axis controls the modulation amount. If you hold down the ALT (PC) / ALT (Mac) modifier key while clicking in the X-Y field, the Y-axis controls the bandwidth. Note that bandwidth is not adjustable when Sine is selected.

The Frequency control determines the color, or quality, of the distortion. If the Mode control is set to Noise, this works in conjunction with the Width control, which defines the noise bandwidth. Lower values lead to more selective distortion frequencies, while higher values affect the entire input signal. Width has no effect in Sine Mode.

Noise and Sine use a single modulation generator. However, Wide Noise has independent noise generators for the left and right channels, which creates a subtle stereo enhancement.

22.13 External Audio Effect

ExternalAudioEffect.png
The External Audio Effect.

(Note: the External Audio Effect is not available in the Intro and Lite Editions.)

The External Audio Effect is a bit different than Live’s other effects devices. Instead of processing audio itself, it allows you to use external (hardware) effects processors within a track’s device chain.

The Audio To chooser selects the outputs on your computer’s audio hardware that will go to your external device, while the Audio From chooser selects the inputs that will bring the processed signal back into Live. As with the track inputs and outputs, the list of available inputs and outputs depends on the Audio Preferences, which can be reached via the ”Configure...” option at the bottom of each chooser.

Below each chooser is a Peak level indicator that shows the highest audio level attained. Click on the indicators to reset them.

The Gain knobs next to the choosers adjust the levels going out of and back into Live. These levels should be set carefully to avoid clipping, both in your external hardware and when returning the audio to your computer.

The Dry/Wet control adjusts the balance between the processed and dry signals. Set it to 100 percent if using the External Audio Effect in a return track.

The Invert button inverts the phase of the processed signal coming back into Live.

Since hardware effects introduce latency that Live cannot automatically detect, you can manually compensate for any delays by adjusting the Hardware Latency slider. The button next to this slider allows you to set your latency compensation amount in either milliseconds or samples. If your external device connects to Live via a digital connection, you will want to adjust your latency settings in samples, which ensures that the number of samples you specify will be retained even when changing the sample rate. If your external device connects to Live via an analog connection, you will want to adjust your latency settings in milliseconds, which ensures that the amount of time you specify will be retained when changing the sample rate. Note that adjusting in samples gives you finer control, so even in cases when you’re working with analog devices, you may want to ”fine tune” your latency in samples in order to achieve the lowest possible latency. In this case, be sure to switch back to milliseconds before changing your sample rate.

Note: If the Delay Compensation option (see 17.5) is unchecked in the Options menu, the Hardware Latency slider is disabled.

For instructions on how to accurately set up latency compensation for your hardware, please see the ”Driver Error Compensation” lesson.

22.14 Filter Delay

FilterDelay.png
The Filter Delay Effect.

(Note: the Filter Delay effect is not available in the Intro and Lite Editions.)

The Filter Delay provides three independent delay lines, each preceded by linked lowpass and highpass filters. This allows applying delay to only certain input signal frequencies, as determined by the filter settings. The feedback from each of the three delays is also routed back through the filters.

Each of the three delays can be switched on and off independently. The Filter Delay device assigns delay 1 to the input signal’s left channel, delay 2 to the left and right channels and delay 3 to the right channel. The Pan controls at the right can override the delay channels’ outputs; otherwise each delay outputs on the channel from which it derives its input.

Each delay channel’s filter has an associated On switch, located to the left of each X-Y controller. The X-Y controllers adjust the lowpass and highpass filters simultaneously for each delay. To edit filter bandwidth, click and drag on the vertical axis; click and drag on the horizontal axis to set the filter band’s frequency.

To refer delay time to the song tempo, activate the Sync switch, which allows using the Delay Time beat division chooser. The numbered switches represent time delay in 16th notes. For example, selecting ”4” delays the signal by four 16th notes, which equals one beat (a quarter note) of delay. With Sync Mode active, changing the Delay Time field percentage value shortens and extends delay times by fractional amounts, thus producing the ”swing” type of timing effect found in drum machines.

If the Sync switch is off, the delay time reverts to milliseconds. In this case, to edit the delay time, click and drag up or down in the Delay Time field, or click in the field and type in a value.

The Feedback parameter sets how much of the output signal returns to the delay line input. Very high values can lead to runaway feedback and produce a loud oscillation — watch your ears and speakers if you decide to check out extreme feedback settings!

Each delay channel has its own volume control, which can be turned up to +12 dB to compensate for drastic filtering at the input.

The Dry control adjusts the unprocessed signal level. Set it to minimum if using Delay in a return track.

22.15 Flanger

Flanger.png
The Flanger Effect.

(Note: the Flanger effect is not available in the Lite Edition.)

Flanger uses two parallel time-modulated delays to create flanging effects.

Flanger’s delays can be adjusted with the Delay Time control. The Feedback control sends part of the output signal back through the device input, while the Polarity switch (”+” or ”-”) sets the polarity. Delay Time and Feedback can be changed simultaneously using the effect’s X-Y controller.

Periodic control of delay time is possible using the envelope section. You can increase or decrease the envelope amount (or invert its shape with negative values), and then use the Attack and Release controls to define envelope shape.

Flanger contains two LFOs to modulate delay time for the left and right stereo channels. The LFOs have six possible waveform shapes: sine, square, triangle, sawtooth up, sawtooth down and random. The extent of LFO influence on the delays is set with the Amount control.

LFO speed is controlled with the Rate control, which can be set in terms of hertz. Rate can also be synced to the song tempo and set in meter subdivisions (e.g., sixteenth notes).

The Phase control lends the sound stereo movement by setting the LFOs to run at the same frequency, but offsetting their waveforms relative to each other. Set this to ”180”, and the LFOs will be perfectly out of phase (180 degrees apart), so that when one reaches its peak, the other is at its minimum.

Spin detunes the two LFO speeds relative to each other. Each delay is modulated at a different frequency, as determined by the Spin amount.

Adjusting the HiPass control will cut low frequencies from the delayed signal.

The Dry/Wet control adjusts the balance between the processed and dry signals. Set it to 100 percent if using Flanger in a return track.

Hi-Quality mode can be toggled on or off via a right-click (PC) / CTRL - click (Mac) context menu entry. Enabling Hi-Quality results in a brighter sound, but there is a very slight increase in CPU usage.

22.16 Frequency Shifter

FrequencyShifter.png
The Frequency Shifter Effect.

(Note: the Frequency Shifter effect is not available in the Intro and Lite Editions.)

The Frequency Shifter moves the frequencies of incoming audio up or down by a user-specified amount in Hertz. Small amounts of shift can result in subtle tremolo or phasing effects, while large shifts can create dissonant, metallic sounds.

The Coarse and Fine knobs set the amount of shift that will be applied to the input. For example, if the input is a sine wave at 440 Hz and the frequency is set to 100 Hz, the output will be a sine wave at 540 Hz.

By changing the Mode from Shift to Ring, Frequency Shifter switches from classical frequency shifting to ring modulation . In Ring mode, the selected frequency amount is added to and subtracted from the input. For example, if the incoming audio signal (A) is a sine wave at 440 Hz and the Frequency is set to 100 Hz (B), the output will contain partials at 340 Hz (A-B) and 540 Hz (A+B).

The Drive button enables a distortion effect, while the slider below it controls the level of the distortion. Drive is only available in Ring mode.

Enabling the Wide button creates a stereo effect by inverting the polarity of the Spread value for the right channel. This means that increasing the Spread value will shift the frequency down in the right channel while shifting it up in the left. Note that Wide has no effect if the Spread value is set to 0.

Frequency Shifter contains two LFOs to modulate the frequency for the left and right stereo channels. The LFOs have six possible waveform shapes: sine, square, triangle, sawtooth up, sawtooth down and random. The extent of LFO influence on the frequency is set with the Amount control.

LFO speed is controlled with the Rate control, which can be set in terms of Hertz. Rate can also be synced to the song tempo and set in meter subdivisions (e.g., sixteenth notes).

The Phase control lends the sound stereo movement by setting the LFOs to run at the same frequency, but offsetting their waveforms relative to each other. Set this to ”180”, and the LFOs will be perfectly out of phase (180 degrees apart), so that when one reaches its peak, the other is at its minimum.

Spin detunes the two LFO speeds relative to each other. Each stereo channel is modulated at a different frequency, as determined by the Spin amount.

When using the random waveform, the Phase and Spin controls are not relevant and do not affect the sound.

The Dry/Wet control adjusts the balance between the processed and dry signals. This knob is called Mix when Drive is enabled. Note that the Drive effect is post-Mix, which means that you can use Frequency Shifter as a pure distortion effect by enabling Drive and setting Mix to 0%.

Frequency shifting is accomplished by simply adding or subtracting a value in Hertz to the incoming audio. This is distinct from pitch shifting , in which the ratios of the incoming frequencies (and thus their harmonic relationships) are preserved. For example, imagine you have an incoming audio signal consisting of sine waves an octave apart at 440 Hz and 880 Hz. To pitch shift this up an octave, we multiply these frequencies by two, resulting in new frequencies at 880 Hz and 1760 Hz.

22.16.1 Frequency Shifter Tips

Frequency shifting and ring modulation can produce some very interesting sounds. Here are some tips for using the Frequency Shifter device.

Drum tuning

Tuning sampled acoustic drums can be tricky. Using a sampler’s transposition controls often changes the character of the sounds in unrealistic ways, resulting in ”pinched” or ”tubby” samples. Frequency shifting can be a useful alternative.

Try using the device in Shift mode with the Dry/Wet amount at 100%. Then adjust the Fine frequency no more than about 100 Hz up or down. This should change the apparent size and tuning of the drum while retaining the quality of the original sample.

Phasing

To create lush phasing effects, try using extremely small amounts of shift (no more than about 2 Hz). Note that the phasing is caused by the interaction of the processed and dry signals, so you won’t hear an effect until you adjust the Dry/Wet balance so that both are audible; the strongest phasing will be heard when Dry/Wet is at 50%.

Tremolo

In Ring mode, frequencies below the audible range (about 20 Hz) create a tremolo effect. You can also impart a sense of stereo motion to the tremolo by turning on Wide and using small Spread values.

Learning more...

Try putting a Spectrum device (see 22.31) after the Frequency Shifter to watch how the signal changes as you change parameters. To get a good overview of what’s happening, try using a simple, continuous sine wave as your input.

22.17 Gate

Gate.png
The Gate Effect.

(Note: the Gate effect is not available in the Lite Edition.)

The Gate effect passes only signals whose level exceeds a user-specified threshold. A gate can eliminate low-level noise that occurs between sounds (e.g., hiss or hum), or shape a sound by turning up the threshold so that it cuts off reverb or delay tails or truncates an instrument’s natural decay.

As of Live 9, Gate’s internal behavior has been subtly improved. Although it works more “correctly” than it did in previous Live versions, Live Sets that use Gate and which were made prior to Live 9 may sound slightly different. To ensure that older Sets sound exactly the same, the Gate Legacy Mode option will be enabled by default when loading an old Set that uses Gate. You can disable this via a right-click (PC) / CTRL - click (Mac) context menu entry in Gate’s title bar.

Gate’s display area shows the level of the input signal in light grey and the level of the output signal in a darker grey with a white outline. This allows you to see the amount of gating that is occurring at any moment, and helps you to set the appropriate parameters.

The Threshold knob sets the gate’s sensitivity. The Threshold value is represented in the display as a horizontal orange line, which can also be dragged.

Return (also known as “hysteresis”) sets the difference between the level that opens the gate and the level that closes it. Higher hysteresis values reduce “chatter” caused by the gate rapidly opening and closing when the input signal is near the threshold level. The Return value is represented in the display as an additional horizontal orange line.

With the Flip button enabled, the gate works in reverse; the signal will only pass if its level is below the threshold.

A gate can only react to an input signal once it occurs. Since it also needs to apply an attack/release envelope, the gating is always a bit too late. A digital gate can solve this problem by simply delaying the input signal a little bit. Gate offers three different Lookahead times: zero ms, one ms and ten ms. The results may sound pretty different depending on this setting.

The Attack time determines how long it takes for the gate to switch from closed to open when a signal goes from below to above the threshold. Very short attack times can produce sharp clicking sounds, while long times soften the sound’s attack.

When the signal goes from above to below the threshold, the Hold time kicks in. After the hold time expires, the gate closes over a period of time set by the Release parameter.

The Floor knob sets the amount of attenuation that will be applied when the gate is closed. If set to -inf dB, a closed gate will mute the input signal. A setting of 0.00 dB means that even if the gate is closed, there is no effect on the signal. Settings in between these two extremes attenuate the input to a greater or lesser degree when the gate is closed.

Normally, the signal being gated and the input source that triggers the gate are the same signal. But by using sidechaining , it is possible to gate a signal based on the level of another signal. To access the Sidechain parameters, unfold the Gate window by toggling the 26132.png button in its title bar.

Enabling this section with the Sidechain button allows you to select another track from the choosers below. This causes the selected track’s signal to act as the gate’s trigger, instead of the signal that is actually being gated.

The Gain knob adjusts the level of the external sidechain’s input, while the Dry/Wet knob allows you to use a combination of sidechain and original signal as the gate’s trigger. With Dry/Wet at 100%, the gate is triggered entirely by the sidechain source. At 0%, the sidechain is effectively bypassed. Note that increasing the gain does not increase the volume of the source signal in the mix. The sidechain audio is only a trigger for the gate and is never actually heard.

Sidechain gating can be used to superimpose rhythmic patterns from one source onto another. For example, a held pad sound can be triggered with the rhythm of a drum loop by inserting a Gate on the pad’s track and choosing the drum loop’s track as the sidechain input.

On the right of the external section are the controls for the sidechain EQ. Enabling this section causes the gate to be triggered by a specific band of frequencies, instead of a complete signal. These can either be frequencies in the signal to be gated or, by using the EQ in conjunction with an external sidechain, frequencies in another track’s audio.

The headphones button between the external and EQ sections allows you to listen to only the sidechain input, bypassing the gate’s output. Since the sidechain audio isn’t fed to the output, and is only a trigger for the gate, this temporary listening option can make it much easier to set sidechain parameters and hear what’s actually making the gate work. When this button is on, the display area shows the level of the sidechain input signal in green.

22.18 Glue Compressor

Glue.png
The Glue Compressor Effect.

(Note: the Glue Compressor effect is not available in the Intro and Lite Editions.)

The Glue Compressor is an analog-modeled compressor created in collaboration with Cytomic, and is based on the classic bus compressor from a famous 80’s mixing console. Like Live’s original Compressor (see 22.7), the Glue Compressor can be used for basic dynamics control of individual tracks, but is mainly designed for use on the Master track or a Group Track to “glue” multiple sources together into a cohesive sounding mix.

The Threshold knob sets where compression begins. Signals above the threshold are attenuated by an amount specified by the Ratio parameter, which sets the ratio between the input and output signal. Unlike the Compressor, the Glue Compressor does not have a user-adjustable knee. Instead, the knee becomes more sharp as the ratio increases.

Attack defines how long it takes to reach maximum compression once a signal exceeds the threshold. The Attack knob’s values are in milliseconds. Release sets how long it takes for the compressor to return to normal operation after the signal falls below the threshold. The Release knob’s values are in seconds. When Release is set to A (Auto), the release time will adjust automatically based on the incoming audio. The Glue Compressor’s Auto Release actually uses two times - a slow one as a base compression value, and a fast one to react to transients in the signal. Auto Release may be too slow to react to sudden changes in level, but generally is a useful way to tame a wide range of material in a gentle way.

Dry/Wet adjusts the balance between the compressed and uncompressed signals. At 100%, only the compressed signal is heard, while at 0%, the device is effectively bypassed. Another way of controlling the amount of compression is with the Range slider, which sets the how much compression can occur. Values between about -60 and -70 dB emulate the original hardware, while values between -40 and -15 dB can be useful as an alternative to the Dry/Wet control. At 0 dB, no compression occurs.

Makeup applies gain to the signal, allowing you to compensate for the reduction in level caused by compression. A Makeup value that roughly corresponds to the position of the needle in the display should result in a level close to what you had before compressing.

The Soft clip switch toggles a fixed waveshaper, useful for taming very loud transients. When enabled, the Glue Compressor’s maximum output level is -.5 dB. (Note that with Oversampling enabled, very loud peaks may still exceed 0 dB.) The Soft clipper is not a transparent limiter, and will distort your signal when active. We recommend leaving it disabled unless this particular type of “colored” distortion is what you’re looking for.

The Glue Compressor’s needle display shows the amount of gain reduction in dB. The Clip LED turns red if the device’s output level is exceeding 0 dB. If Soft clipping is enabled, this LED turns yellow to indicate that peaks are being clipped.

22.18.1 Sidechain Parameters

GlueSidechain.png
The Glue Compressor With Sidechain Section.

Normally, the signal being compressed and the input source that triggers the compressor are the same signal. But by using sidechaining , it is possible to compress a signal based on the level of another signal or a specific frequency component. To access the Sidechain parameters, unfold the Glue Compressor window by toggling the 26177.png button in its title bar.

The sidechain parameters are divided into two sections. On the left are the external controls. Enabling this section with the Sidechain button allows you to select any of Live’s internal routing points from the choosers below. This causes the selected source to act as the Glue Compressor’s trigger, instead of the signal that is actually being compressed.

The Gain knob adjusts the level of the external sidechain’s input, while the Dry/Wet knob allows you to use a combination of sidechain and original signal as the Glue Compressor’s trigger. With Dry/Wet at 100%, the Glue Compressor is triggered entirely by the sidechain source. At 0%, the sidechain is effectively bypassed. Note that increasing the gain does not increase the volume of the source signal in the mix. The sidechain audio is only a trigger for the Glue Compressor and is never actually heard.

On the right of the external section are the controls for the sidechain EQ. Enabling this section causes the Glue Compressor to be triggered by a specific band of frequencies, instead of a complete signal. These can either be frequencies in the compressed signal or, by using the EQ in conjunction with an external sidechain, frequencies in another track’s audio.

The headphones button between the external and EQ sections allows you to listen to only the sidechain input, bypassing the Glue Compressor’s output. Since the sidechain audio isn’t fed to the output, and is only a trigger for the Glue Compressor, this temporary listening option can make it much easier to set sidechain parameters and hear what’s actually making the Glue Compressor work.

Context Menu Options

Oversampling can be toggled on or off via the right-click (PC) / CTRL - click (Mac) context menu. Enabling this option causes the Glue Compressor to internally process at two times the current sampling rate, which may reduce aliasing and transient harshness. There is a slight increase in CPU usage with Oversampling enabled. Note that with Oversampling enabled, the level may exceed 0 dB even with Soft clip enabled.

22.19 Grain Delay

GrainDelay.png
The Grain Delay Effect.

(Note: the Grain Delay effect is not available in the Lite Edition.)

The Grain Delay effect slices the input signal into tiny particles (called ”grains”) that are then individually delayed and can also have different pitches compared to the original signal source. Randomizing pitch and delay time can create complex masses of sound and rhythm that seem to bear little relationship to the source. This can be very useful in creating new sounds and textures, as well as getting rid of unwelcome house guests, or terrifying small pets (just kidding!).

You can route each parameter to the X-Y controller’s horizontal or vertical axis. To assign a parameter to the X-axis, choose it from the parameter row below the controller. To assign a parameter to the Y-axis, use the parameter row on the left side.

To refer delay time to the song tempo, activate the Sync switch, which allows using the Delay Time beat division chooser. The numbered switches represent time delay in 16th notes. For example, selecting ”4” delays the signal by four 16th notes, which equals one beat (a quarter note) of delay. With Sync Mode active, changing the Delay Time field percentage value shortens and extends delay times by fractional amounts, thus producing the ”swing” type of timing effect found in drum machines.

If the Sync switch is off, the delay time reverts to milliseconds. In this case, to edit the delay time, click and drag up or down in the Delay Time field, or click in the field and type in a value.

The Delay Time can also be routed to the horizontal axis of the X-Y controller.

The Spray control adds random delay time changes. Low values smear the signal across time, which adds ”noisiness” to the sound. High Spray values completely break down the structure of the source signal, introducing varying degrees of rhythmic chaos. This is the recommended setting for anarchists.

The size and duration of each grain is a function of the Frequency parameter. The sound of Pitch and Spray depends very much on this parameter.

You can transpose the grain pitch with the Pitch parameter, which acts much like a crude pitch shifter.

The Random Pitch control adds random variations to each grain’s pitch. Low values create a sort of mutant chorusing effect, while high values can make the original source pitch completely unintelligible. This parameter can interact with the main Pitch control, thus allowing degrees of stability and instability in a sound’s pitch structure.

The Feedback parameter sets how much of the output signal returns to the delay line input. Very high values can lead to runaway feedback and produce a loud oscillation — watch your ears and speakers if you decide to check out extreme feedback settings!

Grain Delay also has a dry/wet control, which can be routed to the vertical axis of the X-Y controller.

22.20 Limiter

Limiter.png
The Limiter Effect.

(Note: the Limiter effect is not available in the Lite Edition.)

The Limiter effect is a mastering-quality dynamic range processor that ensures that the output does not exceed a specified level. Limiter is ideal for use in the Master track, to prevent clipping. A limiter is essentially a compressor with an infinite ratio. (For more information about compression theory, see the manual entry for the Compressor device (see 22.5).)

The Gain knob allows you to boost or attenuate the incoming level before applying limiting.

Ceiling sets the absolute maximum level that the Limiter will output. If your incoming signal level has no peaks that are higher than the ceiling, Limiter will have no effect.

The Stereo / L/R switch determines how Limiter treats peaks that occur on only one side of the stereo signal. In L/R mode, Limiter functions as two separate limiters, with independent limiting for each channel. Stereo mode applies limiting to both channels whenever either of them has a peak that requires compression. L/R mode allows Limiter to apply more compression, but with some distortion of the stereo image.

The Lookahead chooser affects how quickly Limiter will respond to peaks that require compression. Shorter Lookahead times allow for more compression, but with an increase in distortion — particularly in the bass.

The Release knob adjusts how long it takes for Limiter to return to normal operation after the signal falls below the ceiling. With Auto enabled, Limiter analyzes the incoming signal and sets an appropriate release time automatically.

The meter gives a visual indication of how much gain reduction is being applied to the signal.

Note that any devices or channel faders that appear after Limiter may add gain. To ensure that your final output will never clip, place Limiter as the last device in the Master track’s device chain and keep your Master fader below 0 dB.

22.21 Looper

Looper.png
The Looper Effect.

(Note: the Looper effect is not available in the Lite Edition.)

Looper is an audio effect based on classic real-time looping devices. It allows you to record and loop audio, creating endless overdubs that are synced to your Set. If the Set is not playing, Looper can analyze incoming audio and set Live’s tempo to match it. You can also predefine a loop length before recording and Live’s tempo will adjust so that your loop fits into the specified number of bars. Furthermore, audio can be imported to Looper to create a background for newly overdubbed material, or exported from Looper as a new clip.

The top half of Looper’s interface is a large display area optimized for easy readability during performance. During recording, the entire display area turns red. After recording, the display shows the current position in the loop and the total loop length in bars and beats.

Looper’s transport buttons work in a similar way to other transport controls in Live. The Record button records incoming audio until another button is pressed. This overwrites any audio currently stored in Looper. Overdub continues to add additional layers of incoming audio that are the length of the originally recorded material. The Play button plays back the current state of Looper’s buffer without recording any new material. The Stop button stops playback.

The behavior of the transport controls changes depending on whether or not Live’s playback is running. With the transport running, Looper behaves like a clip, and is subject to launch quantization as determined by the Quantization chooser (see 4.11). When Live’s playback is stopped, Looper’s transport engages immediately, regardless of the Quantization setting.

The Clear button erases Looper’s buffer. If you press Clear in Overdub mode while the transport is running, the contents of the buffer are cleared but the tempo and length are maintained. Pressing Clear in any other mode resets the tempo and length.

The Undo button erases everything that you’ve overdubbed since the last time Overdub was enabled. Your original recording, and anything that was overdubbed in a previous pass, is preserved. After pressing Undo, the button changes to Redo, which replaces the material removed by the last undo.

The large button below the transport controls is the Multi-Purpose Transport Button. As with the normal transport buttons, this button’s behavior changes depending on Looper’s current playback state, and whether or not material has already been recorded. If the buffer is empty, a single click starts recording. If Looper is recording, overdubbing or stopped, a single click switches to play mode. During playback, a click switches to overdub mode, allowing you to toggle back and forth between overdub and playback via additional single clicks.

Quickly pressing the button twice stops Looper, from either play or overdub mode. Clicking and holding the button for two seconds while in play mode activates Undo or Redo. Pressing and holding for two seconds while stopped clears Looper’s buffer.

26246.png
Diagram of Looper’s Multi-Purpose Transport Button Behavior

Looper’s Multi-Purpose Transport Button is optimized for use with a MIDI footswitch. To assign a footswitch, enter MIDI Map Mode (see 26.2.1), click the button and then press your attached footswitch. Then exit MIDI Map Mode.

The Tempo Control chooser affects how Looper determines the tempo of recorded material:

  • None : Looper’s internal tempo is independent of Live’s global tempo.
  • Follow song tempo : The speed of Looper’s playback will be adjusted so that the recorded material plays back at Live’s global tempo.
  • Set & Follow song tempo : Live’s global tempo will be adjusted to match the tempo of material recorded into Looper. Any subsequent changes to Live’s global tempo will adjust the speed of Looper’s playback so that the recorded material plays back at the new global tempo.

The Record Length chooser is used to set the length of recorded material. Its behavior changes depending on whether or not Live’s global transport is running and, depending on the setting of the Tempo Control chooser, can set Live’s global tempo:

  • Song running : If Looper’s Record Length chooser is set to the default ”x bars,” Looper will record until you press another transport button. If you specify a fixed number of bars to record by selecting another option in the chooser, Looper will record for the specified time and then switch to Play or Overdub, as determined by the button next to this chooser.
  • Song not running : If Looper’s Record Length chooser is set to the default ”x bars,” Looper will make a guess about the tempo of the material you’ve recorded as soon as you press Overdub, Play or Stop. But this might result in a tempo that’s twice or half as fast as you’d like. If you first specify a fixed number of bars, Looper’s tempo will adjust so that your recording fits into this time.

The Song Control chooser determines how Looper’s transport controls will affect Live’s global transport:

  • None means that Looper’s transport controls have no effect on Live’s global transport.
  • Start Song will start Live’s global transport whenever Looper enters Play or Overdub mode. Looper’s Stop button has no effect on the global transport.
  • Start & Stop Song locks Live’s global transport to Looper’s transport controls. Entering Play or Overdub mode will start Live’s transport, while pressing Looper’s Stop button will stop Live’s transport.

The ”\*2” button doubles the length of Looper’s recording buffer. Any material that you’ve already recorded will simply be duplicated. This allows you to, for example, record a series of one-bar ideas, and then overlay a series of two-bar ideas. The length and tempo of Looper’s buffer is shown in the display area.

Similarly, the ”:2” button cuts the length of the current buffer in half. The material in the currently playing half is kept, while the other half is discarded.

The ”Drag me!” area in the display allows you to export Looper’s buffer as a new audio file. You can drag and drop to the browser or directly to a track, creating a new clip. The newly created clip’s Warp mode will be set to Re-Pitch by default (see 9.3.4). You can also drag audio files to the ”Drag me!” area, which will replace the contents of Looper’s buffer. You can then use this material as a bed for further overdubs, for example.

The Speed knob adjusts Looper’s playback speed (and thus pitch). The up and down arrow buttons to the left are shortcuts to raise or lower the pitch by octaves (thus doubling or halving the playback speed). These buttons are subject to the Quantization chooser setting.

Enabling the Reverse button plays the previously recorded material backwards. Any material that you overdub after enabling Reverse will be played forward. Note that disabling Reverse will then swap this behavior; the original material will play forward again, while the material that was overdubbed while Reverse was enabled will play backwards. Engaging the Reverse button is subject to the Quantization chooser setting.

Feedback sets the amount of previously recorded signal that is fed back into Looper when overdubbing. When set to 100%, the previously recorded material will never decrease in volume. When set to 50%, it will be half as loud with each repetition. Any changes to the Feedback amount won’t take effect until the next repetition. Note that Feedback has no effect in Play mode; each repetition will be at the same volume.

The Input -> Output chooser provides four options for monitoring Looper’s input:

  • Always allows the input signal to be heard regardless of Looper’s playing or recording state. You’ll typically want to choose Always when using Looper as an effect in a single track.
  • Never means that the input signal will never be heard. You’ll typically want to choose Never when using Looper as an effect in a return track, where it may be fed by send levels from a variety of other tracks.
  • Rec/OVR means that the input is only audible when recording or overdubbing, but not when Looper is in Play mode or stopped. This is useful for situations in which you are feeding audio to multiple tracks, each containing its own Looper. If each of these Loopers is controlled with its own foot pedal, you can switch the recording and playback state while playing an instrument, without having to worry about monitor settings.
  • Rec/OVR/Stop allows the input signal to be heard except when Looper is in play mode. This is similar to Beat Repeat’s (see 22.4) Insert mode, and can be used to record material that can suddenly interrupt your live playing.

22.21.1 Feedback Routing

Looper can be used as both a source and a target for internal routing (see 14.6) to other tracks. This allows you to, for example, create Looper overdubs that continually feed back through another track’s devices. To set this up:

  1. Insert Looper on a track.
  2. Record at least one pass of material into Looper.
  3. Create another audio track.
  4. In the new track’s top Audio From and Audio To choosers, select the track containing the Looper.
  5. In the new track’s bottom Audio From and Audio To choosers, select ”Insert-Looper.”
  6. Switch this track’s Monitoring to ”In.”
  7. Add additional effects devices to the device chain of the new track.
  8. Put Looper into Overdub mode.

Looper’s output will now be routed through the other track’s device chain and then back into itself, creating increasingly processed overdub layers with each pass.

22.22 Multiband Dynamics

MultibandDynamics.png
The Multiband Dynamics Effect.

(Note: the Multiband Dynamics effect is not available in the Intro and Lite Editions.)

The Multiband Dynamics device is a flexible tool for modifying the dynamic range of audio material. Designed primarily as a mastering processor, Multiband Dynamics allows for upward and downward compression and expansion of up to three independent frequency bands, with adjustable crossover points and envelope controls for each band. Each frequency range has both an upper and lower threshold, allowing for two types of dynamics processing to be used simultaneously per band.

22.22.1 Dynamics Processing Theory

To understand how to use the Multiband Dynamics device, it helps to understand the four different methods of manipulating dynamics.

When we use the term “compression,” we’re typically talking about lowering the level of signals that exceed a threshold. This is how Live’s Compressor (see 22.7) works, and is more accurately called downward compression because it pushes loud signals down, thus reducing the dynamic range. But it is also possible to reduce a signal’s dynamic range by raising the levels of signals that are below a threshold. This much-less-common form of compression is called upward compression . As you can see from this diagram, employing either type of compression results in a signal with a smaller dynamic range than the original.

26297.png
Downward and Upward Compression.

The opposite of compression is expansion. A typical expander lowers the levels of signals that are below a threshold. This is how Live’s Gate (see 22.17) works, and is more accurately called downward expansion because it pushes quiet signals down, thus increasing the dynamic range. It is also possible to increase a signal’s dynamic range by raising the levels of signals that are above a threshold. Like upward compression, this technique is known as upward expansion and is much less common. This diagram shows that either type of expansion results in a signal with a larger dynamic range.

26315.png
Downward and Upward Expansion.

To summarize:

  • Downward compression (common): make loud signals quieter
  • Upward compression (uncommon): make quiet signals louder
  • Downward expansion (common): make quiet signals quieter
  • Upward expansion (uncommon): make loud signals louder

The Multiband Dynamics device allows for all of these types of processing. In fact, because the device allows for incoming audio to be divided into three frequency bands, and each band has both an upper and lower threshold, a single instance of Multiband Dynamics can provide six types of dynamics processing simultaneously.

22.22.2 Interface and Controls

The High and Low buttons toggle the high and low bands on or off. With both bands off, the device functions as a single-band effect. In this case, only the Mid controls affect the incoming signal. The frequency sliders below the High and Low buttons adjust the crossovers that define the frequency ranges for each band. If the low frequency is set to 500 Hz and the high frequency is set to 2000 Hz, then the low band goes from 0 Hz to 500 Hz, the mid band from 500 Hz to 2000 Hz and the high band from 2000 Hz up to whatever your soundcard or sample rate supports.

Each band has activator and solo buttons. With the activator button disabled for a given band, its compression/expansion and gain controls are bypassed. Soloing a band mutes the others. The Input knobs boost or attenuate the level of each band before it undergoes dynamics processing, while the Output knobs to the right of the display adjust the levels of the bands after processing.

The display area provides a way of both visualizing your dynamics processing and adjusting the relevant compression and expansion behavior. For each band, the output level is represented by large bars, while the input level before processing is represented by small bars. With no processing applied, the input meters will be aligned with the top of the output meters. The scaling along the bottom of the display shows dB. As you adjust the gain or dynamics processing for a band, you can see how its output changes in comparison to its input.

As you move your mouse over the display, the cursor will change to a bracket as it passes over the edges of the blocks on the left or right side. These blocks represent the signal levels under the Below and over the Above thresholds, respectively. Dragging left or right on the edges of these blocks adjusts the threshold level. Holding down Shift while dragging a threshold will adjust the same threshold for all bands. Hold down ALT (PC) / ALT (Mac) to simultaneously adjust the Above and Below thresholds for a single band.

As you mouse over the middle of the block, the cursor will change to an up-down arrow. Click and drag up or down to make the signal within the selected volume range louder or quieter. Holding down Shift while dragging up or down will adjust the volume of the same block for all bands. Hold down ALT (PC) / ALT (Mac) to simultaneously adjust the Above and Below volumes for a single band. Double-clicking within the region resets the volume to its default.

In technical terms, lowering the volume in the block above the Above threshold applies downward compression, while raising it applies upward expansion. Likewise, lowering the volume in the block below the Below threshold applies downward expansion, while raising it applies upward compression. In all cases, you are adjusting the ratio of the compressor or expander.

The thresholds and ratios of all bands can also be adjusted via the column to the right of the display. The ”T,” ”B” and ”A” buttons at the bottom right of the display area toggle between displaying the Time (attack and release), Below (threshold and ratio) and Above (threshold and ratio) for each band.

For the Above thresholds, Attack defines how long it takes to reach maximum compression or expansion once a signal exceeds the threshold, while Release sets how long it takes for the device to return to normal operation after the signal falls below the threshold.

For the Below thresholds, Attack defines how long it takes to reach maximum compression or expansion once a signal drops below the threshold, while Release sets how long it takes for the device to return to normal operation after the signal goes above the threshold.

With Soft Knee enabled, compression or expansion begins gradually as the threshold is approached.

The RMS/Peak switch also affects how quickly Multiband Dynamics responds to level changes. With Peak selected, the device reacts to short peaks within a signal. RMS mode causes it to be less sensitive to very short peaks and to begin processing only when the incoming level has crossed the threshold for a slightly longer time.

The global Output knob adjusts the overall output gain of the device.

The Time control scales the durations of all of the Attack and Release controls. This allows you to maintain the same relative envelope times, but make them all faster or slower by the same amount.

The Amount knob adjusts the intensity of the compression or expansion applied to all bands. At 0%, each compressor/expander has an effective ratio of 1, meaning that it has no effect on the signal.

22.22.3 Sidechain Parameters

MultibandDynamicsSidechain.png
The Multiband Dynamics Device With Sidechain Section.

Normally, the signal being processed and the input source that triggers the device are the same signal. But by using sidechaining , it is possible to apply dynamics processing to a signal based on the level of another signal or a specific frequency component. To access the Sidechain parameters, unfold the Multiband Dynamics window by toggling the 26352.png button in its title bar.

Enabling the Sidechain button allows you to select any of Live’s internal routing points from the choosers below. This causes the selected source to act as the device’s trigger, instead of the signal that is actually being processed.

The Gain knob adjusts the level of the external sidechain’s input, while the Dry/Wet knob allows you to use a combination of sidechain and original signal as the trigger. With Dry/Wet at 100%, the device is triggered entirely by the sidechain source. At 0%, the sidechain is effectively bypassed. Note that increasing the gain does not increase the volume of the source signal in the mix. The sidechain audio is only a trigger for the device and is never actually heard.

The headphones button allows you to listen to only the sidechain input, bypassing the device’s output. Since the sidechain audio isn’t fed to the output, and is only a trigger for the device, this temporary listening option can make it much easier to set sidechain parameters and hear what’s actually making the device work.

22.22.4 Multiband Dynamics Tips

Multiband Dynamics is a feature-rich and powerful device, capable of up to six independent types of simultaneous processing. Because of this, getting started can be a bit intimidating. Here are some real-world applications to give you some ideas.

Basic Multiband Compression

By using only the upper thresholds, Multiband Dynamics can be used as a traditional ”downward” compressor. Adjust the crossover points to suit your audio material, then apply downward compression (by dragging down in the upper blocks in the display or by setting the numerical ratios to values greater than 1.)

De-essing

To remove ”harshness” caused by overly loud high frequency content, try enabling only the upper band and setting its crossover frequency to around 5 kHz. Then gradually adjust the threshold and ratio to apply subtle downward compression. It may help to solo the band to more easily hear the results of your adjustments. Generally, de-essing works best with fairly fast attack and release times.

Uncompression

Mastering engineers are often asked to perform miracles, like adding punch and energy to a mix that has already been heavily compressed, and thus has almost no remaining transients. Most of the time, these mixes have also been heavily maximized, meaning that they also have no remaining headroom. Luckily, upward expansion can sometimes help add life back to such overly squashed material. To do this:

  1. Turn down the Input knob to provide some additional headroom.
  2. Adjust the Above thresholds for the bands so that they’re below the highest peaks.
  3. Add a small amount of upward expansion to each band. Be careful — excessive upward expansion can make transients very loud.
  4. Carefully adjust the attack and release times for each band. Note that, unlike in typical downward compression, very fast attack times will increase the impact of transients, while slower times result in a more muffled sound.

Note : Adding a maximizer or limiter to boost gain after you’ve returned some peaks to your material may simply destroy them again.

22.23 Overdrive

Overdrive.png
The Overdrive Effect.

(Note: the Overdrive effect is not available in the Intro and Lite Editions.)

Overdrive is a distortion effect that pays homage to some classic pedal devices commonly used by guitarists. Unlike many distortion units, it can be driven extremely hard without sacrificing dynamic range.

The distortion stage is preceded by a bandpass filter that can be controlled with an X-Y controller. To define the filter bandwidth, click and drag on the vertical axis. To set the position of the frequency band, click and drag on the horizontal axis. These parameters can also be set via the slider boxes below the X-Y display.

The Drive control sets the amount of distortion. Note that 0% does not mean zero distortion!

Tone acts as a post-distortion EQ control. At higher values, the signal has more high-frequency content.

The Dynamics slider allows you to adjust how much compression is applied as the distortion is increased. At low settings, higher distortion amounts result in an increase in internal compression and make-up gain. At higher settings, less compression is applied.

The Dry/Wet control adjusts the balance between the processed and dry signals. Set it to 100 percent if using Overdrive in a return track.

22.24 Phaser

Phaser.png
The Phaser Effect.

(Note: the Phaser effect is not available in the Lite Edition.)

Phaser uses a series of all-pass filters to create a phase shift in the frequency spectrum of a sound.

The Poles control creates notches in the frequency spectrum. The Feedback control can then be used to invert the waveform and convert these notches into peaks (or poles). Filter cutoff frequency is changed with the Frequency control, which can be adjusted in tandem with Feedback using the effect’s X-Y controller.

The device has two modes, Space and Earth, for changing the spacing of notches along the spectrum, and hence the ”color” of the sound. This effect can be further adjusted with the Color control.

Periodic control of the filter frequency is possible using the envelope section. You can increase or decrease the envelope amount (or invert its shape with negative values), and then use the Attack and Release controls to define envelope shape.

Phaser contains two LFOs to modulate filter frequency for the left and right stereo channels. The LFOs have six possible waveform shapes: sine, square, triangle, sawtooth up, sawtooth down and random. The extent of LFO influence on the filter frequency is set with the Amount control.

LFO speed is controlled with the Rate control, which can be set in terms of hertz. Rate can also be synced to the song tempo and set in meter subdivisions (e.g., sixteenth notes).

The Phase control lends the sound stereo movement by setting the LFOs to run at the same frequency, but offsetting their waveforms relative to each other. Set this to ”180”, and the LFOs will be perfectly out of phase (180 degrees apart), so that when one reaches its peak, the other is at its minimum.

Spin detunes the two LFO speeds relative to each other. Each filter frequency is then modulated using a different LFO frequency, as determined by the Spin amount.

The Dry/Wet control adjusts the balance between the processed and dry signals. Set it to 100 percent if using Phaser in a return track.

22.25 Ping Pong Delay

PingPongDelay.png
The Ping Pong Delay Effect.

(Note: the Ping Pong Delay effect is not available in the Lite Edition.)

The Ping Pong Delay effect uses a single tapped delay line to create a delay that jumps from the left to the right output.

The delay is preceded by a lowpass and highpass filter that can be controlled with an X-Y controller. To define the filter bandwidth, click and drag on the vertical axis. To set the position of the frequency band, click and drag on the horizontal axis.

To refer delay time to the song tempo, activate the Sync switch, which allows using the Delay Time beat division chooser. The numbered switches represent time delay in 16th notes. For example, selecting ”4” delays the signal by four 16th notes, which equals one beat (a quarter note) of delay. This delay time represents the time it takes for the input signal to appear at the left channel. The delay time between the input and the right channel is twice as long.

If the Sync switch is off, the delay time reverts to milliseconds. In this case, to edit the delay time, click and drag up or down in the Delay Time field, or click in the field and type in a value.

The Feedback parameter controls how much of the right channel output signal returns to the delay line input. The feedback loop also includes a filter that can color the feedback sound, thus producing different timbres with successive echoes.

The Freeze button, labeled ”F”, will cause the delay to endlessly cycle the audio which is in its buffer at the moment that the button is pressed, ignoring any new input until Freeze is disabled.

The Dry/Wet control adjusts the balance between the processed and dry signals. Set it to 100 percent if using Ping Pong Delay in a return track.

Changing the delay time while Ping Pong Delay is processing audio can cause abrupt changes in the sound of the delayed signal. You can choose between three delay transition modes via the right-click (PC) / CTRL - click (Mac) context menu of the device’s title bar:

  • Repitch causes a pitch variation when changing the delay time, similar to the behavior of old hardware delay units.
  • Fade creates a crossfade between the old and new delay times. This sounds similar to time stretching if the delay time is gradually changed. Fade mode is the default option.
  • Jump immediately jumps to the new delay time. Note that this will cause an audible click if the delay time is changed while delays are sounding. Jump mode corresponds to the default behavior prior to Live 8. When loading Sets that were made in earlier versions, Jump will be selected automatically.

22.26 Redux

Redux.png
The Redux Effect.

Nostalgic for the famed low-resolution sound quality of the Ensoniq Mirage, Fairlight CMI or Commodore-64 computer? Redux returns us to the Dark Ages of digital by reducing a signal’s sample rate and bit resolution.

The Downsample section has two parameters: ”Downsample” and a downsample Mode switch.

If the downsample dial is set to ”1”, every input sample passes to the output and the signal does not change. If set to ”2”, only every second sample will be processed, so the result sounds a bit more ”digital.” The higher the number, the lower the resulting sample rate, and the more ”deconstructed” the sound. Downsampling is like applying a mosaic effect to an image: There’s a loss of information and sharp edges occur between the blocks.

The Downsample Mode switch defines if the downsampling either interpolates over a smaller range (”soft,” down to 20.0 samples) or does not interpolate over a larger range (”hard,” down to 200 samples).

Bit Reduction is similar, but while downsampling superimposes a grid in time, bit reduction does the same for amplitude.

If the Bit Reduction amplitude dial is set to 8, amplitude levels are quantized to 256 steps (8 bit resolution). If set to 1, the result is pretty brutal: Each sample contains either a full positive or full negative signal, with nothing in between.

Bit Reduction defines an input signal of 0dB as 16 bit. Signals above 0dB are clipped, and the red overload LED will light up.

Turning off Bit Reduction results in modest CPU savings.

22.27 Resonators

Resonators.png
The Resonators Effect.

(Note: the Resonators effect is not available in the Intro and Lite Editions.)

This device consists of five parallel resonators that superimpose a tonal character on the input source. It can produce sounds resembling anything from plucked strings to vocoder-like effects. The resonators are tuned in semitones, providing a musical way of adjusting them. The first resonator defines the root pitch and the four others are tuned relative to this pitch in musical intervals.

The input signal passes first through a filter, and then into the resonators. There are four input filter types to select from: lowpass, bandpass, highpass and notch. The input filter frequency can be adjusted with the Frequency parameter.

The first resonator is fed with both the left and right input channels, while the second and fourth resonators are dedicated to the left channel, and the third and fifth to the right channel.

The Note parameter defines the root pitch of all the resonators ranging from C-1 to C5. It can also be detuned in cents using the Fine parameter. The Decay parameter lets you adjust the amount of time it takes for the resonators to be silent after getting an input signal. The longer the decay time, the more tonal the result will be, similar to the behavior of an undamped piano string. As with a real string, the decay time depends on the pitch, so low notes will last longer than higher ones. The Const switch holds the decay time constant regardless of the actual pitch.

Resonators provides two different resonation modes. Mode A provides a more realistic sounding resonation, while Mode B offers an effect that is especially interesting when Resonator I’s Note parameter is set to lower pitches.

The brightness of the resulting sound can be adjusted using the Color control.

All of the resonators have an On/Off switch and a Gain control. A resonator that is turned off does not need CPU. Turning off the first resonator does not affect the other ones.

Resonators II through V follow the Note parameter defined in Resonator I, but they can each be individually transposed +/- 24 semitones using the Pitch controls and detuned in cents using the Detune controls.

The output section features the obligatory Dry/Wet control and a Width parameter that affects only the wet signal and blends the left and right outputs of Resonators II-V into a mono signal if set to zero.

22.28 Reverb

Reverb.png
The Reverb Effect.

22.28.1 Input Processing

The input signal passes first through high and low cut filters, whose X-Y controller allows changing the band’s center frequency (X-axis) and bandwidth (Y-axis). Either filter may be switched off when not needed to save CPU power.

Predelay controls the delay time, in milliseconds, before the onset of the first early reflection. This delays the reverberation relative to the input signal. One’s impression of the size of a real room depends partly on this delay. Typical values for ”natural” sounds range from 1ms to 25ms.

22.28.2 Early Reflections

These are the earliest echoes that you hear after they bounce off a room’s walls, before the onset of the diffused reverberation ”tail.” Their amplitude and distribution give an impression of the room’s character.

The Shape control ”sculpts” the prominence of the early reflections, as well as their overlap with the diffused sound. With small values, the reflections decay more gradually and the diffused sound occurs sooner, leading to a larger overlap between these components. With large values, the reflections decay more rapidly and the diffused onset occurs later. A higher value can sometimes improve the source’s intelligibility, while a lower value may give a smoother decay.

Spin applies modulation to the early reflections. The X-Y control accesses the depth and frequency of these modulations. A larger depth tends to provide a less-colored (more spectrally neutral) late diffusion response. If the modulation frequency is too high, doppler frequency shifting of the source sound will occur, along with surreal panning effects. Spin may be turned off, using the associated switch, for modest CPU savings.

22.28.3 Global Settings

The Quality chooser controls the tradeoff between reverb quality and performance. ”Eco” uses minimal CPU resources, while ”High” delivers the richest reverberation.

The Size parameter controls the ”room’s” volume. At one extreme, a very large size will lend a shifting, diffused delay effect to the reverb. The other extreme — a very small value — will give it a highly colored, metallic feel.

The Stereo Image control determines the width of the output’s stereo image. At the highest setting of 120 degrees, each ear receives a reverberant channel that is independent of the other (this is also a property of the diffusion in real rooms). The lowest setting mixes the output signal to mono.

22.28.4 Diffusion Network

The Diffusion network creates the reverberant tail that follows the early reflections. The decay time control adjusts the time required for this tail to drop to 1/1000th (-60 dB) of its initial amplitude.

High and low shelving filters provide frequency-dependent reverberation decay. The high-frequency decay models the absorption of sound energy due to air, walls and other materials in the room (people, carpeting and so forth). The low shelf provides a thinner decay. Each filter may be turned off to save CPU consumption.

The Freeze control freezes the diffuse response of the input sound. When on, the reverberation will sustain almost endlessly. Cut modifies Freeze by preventing the input signal from adding to the frozen reverberation; when off, the input signal will contribute to the diffused amplitude. Flat bypasses the high and low shelf filters when freeze is on. If Flat is off, the frozen reverberation will lose energy in the attenuated frequency bands, depending on the state of the high and low shelving filters.

The Echo Density and Scale parameters provide additional control over the diffusion’s density and coarseness, and, when the room size is extremely small, have a large impact on the coloration contributed by the diffusion.

The Chorus section adds a little modulation and motion to the diffusion. Like the Spin section, you can control the modulation frequency and amplitude, or turn it off.

22.28.5 Output

At the reverb output, you can adjust the effect’s overall Dry/Wet mix, and vary the amplitude of reflections and diffusion with the Reflect Level and Diffuse Level controls.

22.29 Saturator

Saturator.png
The Saturator Effect.

(Note: the Saturator effect is not available in the Lite Edition.)

Saturator is a waveshaping effect that can add that missing dirt, punch or warmth to your sound. It can coat input signals with a soft saturation or drive them into many different flavors of distortion.

An X-Y grid helps to visualize Saturator’s shaping curve. The shaper’s input and output values are mapped to the X and Y axes, respectively. The curve defines the transfer function, which is the extent to which output values fluctuate in relation to input values. Because this is usually a nonlinear process, the incoming signal is reshaped to a greater or lesser degree depending upon its level at each moment in time.

Incoming signals are first clipped to the dB level set by the Drive control. The meter on the right side of the display shows how much Saturator is influencing the signal.

Signal shaping has six fixed modes: Analog Clip, Soft Sine, Medium Curve, Hard Curve, Sinoid Fold and Digital Clip. There is also the flexible Waveshaper mode, featuring six adjustable waveshaping parameters.

In the Analog Clip and Digital Clip modes, the signal is clipped completely and immediately. Soft Sine, Medium Curve and Hard Curve modes soften signal clipping to varying degrees. Sinoid Fold mode can be good for special effects.

The most dramatic effects can be created by selecting the Waveshaper curve, which has its own dedicated set of controls. To access these six parameter fields, unfold the Saturator window by toggling the 26491.png button in its title bar.

The Waveshaper mode’s six additional parameters are: Drive, Lin, Curve, Damp, Depth and Period.

  • Drive determines how much the input signal will be influenced by the Waveshaper parameters. Setting Drive to zero will negate the effect entirely.
  • Lin works together with the Curve and Depth parameters to alter the linear portion of the shaping curve.
  • Curve adds mostly third-order harmonics to the input signal.
  • Damp flattens any signal near the grid’s origin. It behaves like an ultra-fast noise gate.
  • Depth controls the amplitude of a sine wave that is superimposed onto the distortion curve.
  • Period determines the density of ripples in the superimposed sine wave.

The DC button activates a DC filter at Saturator’s input stage. This is mainly useful for removing DC offsets from audio material that contains them.

Activating the Color button enables two filters. The first of these, controlled with the Base control, dictates how much the effect will be reduced or increased for very low frequencies. The second filter, essentially an equalizer, is used for controlling higher frequencies. It is shaped with the Freq (cutoff frequency), Width and Depth controls.

The Output control attenuates the level at the device output. When the Soft Clip switch is activated, Saturator will also apply an instance of its ”Analog Clip” curve to the output.

The Dry/Wet control adjusts the balance between the processed and dry signals. Set it to 100 percent when using Saturator in a return track.

Aliasing can be reduced by enabling Hi-Quality mode, which can be accessed via the right-click (PC) / CTRL - click (Mac) context menu. This improves the sound quality, particularly with high frequency signals, but there is a slight increase in CPU usage.

22.30 Simple Delay

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The Simple Delay Effect.

The Simple Delay provides two independent delay lines, one for each channel (left and right).

With Link engaged, the right channel’s delay setting is disabled, thus applying the left channel’s settings to both sides.

To refer delay time to the song tempo, activate the Sync switch, which allows using the Delay Time beat division chooser. The numbered switches represent time delay in 16th notes. For example, selecting ”4” delays the signal by four 16th notes, which equals one beat (a quarter note) of delay.

If the Sync switch is off, the delay time reverts to milliseconds. In this case, to edit the delay time, click and drag up or down in the Delay Time field, or click in the field and type in a value.

The Feedback parameter defines how much of each channel’s output signal feeds back into the delay lines’ inputs. Internally, they are two independent feedback loops, so a signal on the left channel does not feed back into the right channel and vice versa.

The Dry/Wet control adjusts the balance between the processed and dry signals. Set it to 100 percent when using Simple Delay in a return track.

Changing the delay time while Simple Delay is processing audio can cause abrupt changes in the sound of the delayed signal. You can choose between three delay transition modes via the right-click (PC) / CTRL - click (Mac) context menu of the device’s title bar:

  • Repitch causes a pitch variation when changing the delay time, similar to the behavior of old hardware delay units.
  • Fade creates a crossfade between the old and new delay times. This sounds similar to time stretching if the delay time is gradually changed. Fade mode is the default option.
  • Jump immediately jumps to the new delay time. Note that this will cause an audible click if the delay time is changed while delays are sounding. Jump mode corresponds to the default behavior prior to Live 8. When loading Sets that were made in earlier versions, Jump will be selected automatically.

22.31 Spectrum

Spectrum.png
The Spectrum Device.

(Note: the Spectrum device is not available in the Intro and Lite Editions.)

Spectrum performs realtime frequency analysis of incoming audio signals. The results are represented in a graph, with dB along the vertical axis and frequency/pitch along the horizontal. The peak levels are retained on the graph until the song is restarted. Note that Spectrum is not an audio effect, but rather a measurement tool — it does not alter the incoming signal in any way.

The Block chooser selects the number of samples that will be analyzed in each measurement. Higher values result in better accuracy, but at the expense of increased CPU load.

Channel determines which channel is analyzed — left, right or both.

The Refresh slider determines how often Spectrum should perform an analysis. As with the Block parameter, this allows for a tradeoff between accuracy and CPU load. A fast response time is more accurate, but also more CPU intensive.

The Avg slider allows you to specify how many blocks of samples will be averaged for each update of the display. With a setting of one, each block is shown. This results in much more activity in the display, which can be useful for finding the spectrum of short peaks. As you increase the Avg value, the display updates more smoothly, providing an average of the spectrum over time. This is more consistent with the way we actually hear.

The Graph button switches between displaying the spectrum as a single interpolated line and discrete frequency bins.

Max toggles the display of the accumulated maximum amplitude. With Max enabled, you can reset the maximum amplitude by clicking in the display.

The Scale X buttons allow you to toggle the scaling of the frequency display between linear, logarithmic, and semitone. Note that logarithmic and semitone are actually the same scaling, but switch the legending at the top of the display between Hertz and note names. Linear scaling is particularly useful for detailed analysis of high frequencies.

As you move your mouse over Spectrum’s display, a box appears that shows the amplitude, frequency and note name at the pointer’s position. The Range/Auto button at the bottom left of Spectrum’s interface toggles between manually and automatically adjusting the display’s dynamic range. With Range selected, you can zoom and scroll the amplitude by moving your mouse over the amplitude legending on the display’s left side. Drag vertically to scroll and horizontally to zoom. You can also use the Range sliders to set the minimum and maximum amplitude values shown. With Auto selected, the display automatically scales itself based on the incoming audio level. Note that in Auto mode, the Range sliders and zooming are disabled.

To get an even better view, you can toggle the location of the display between the device chain and Live’s main window by clicking the 26541.png button in Spectrum’s title bar or by double-clicking in the display.

22.32 Utility

Utility.png
The Utility Effect.

Utility can perform some very useful tasks, especially in combination with other devices.

The Mute button simply silences the incoming signal when enabled. Note: The active/mute controls of a track are always placed at the very end of the signal chain. However, since you can place Utility anywhere in a signal chain, you can use its mute function to cut the input of a delay line or reverb without turning off the output of these devices.

The DC switch filters out DC offsets and extremely low frequencies that are far below the audible range. It will only have a sonic effect if a signal contains these frequencies and is processed after Utility with nonlinear effects such as compressors or waveshapers.

The Gain control adjusts the level of the input signal +/- 35 dB.

The Channel Mode chooser allows selective processing of the left and right channels of a sample. If, for example, Left is selected, the right channel is ignored and the left channel appears on both outputs. This is especially useful if you have a stereo file that contains different information on both channels and you want to use only one.

The Panorama chooser pans the signal anywhere in the stereo field.

The Width control acts as a continuous mono to stereo controller when set from 0 to 100 percent. However, beyond 100 percent the output starts to ”fold in” on itself. At 200 percent the output contains only the difference between the left and right channels. If either Left or Right have been chosen in the Channel Mode chooser, the Width control has no function and is therefore disabled.

At the bottom of the device you will find two Phase controls, one for each channel. As their names imply, they invert the phase of each channel.

22.33 Vinyl Distortion

VinylDistortion.png
The Vinyl Distortion Effect.

(Note: the Vinyl Distortion effect is not available in the Intro and Lite Editions.)

The Vinyl Distortion effect emulates some of the typical distortions that occur on vinyl records during playback. These distortions are caused by the geometric relationships between the needle and the recorded groove. The effect also features a crackle generator for adding noisy artifacts.

The Tracing Model section adds even harmonic distortion to the input signal. Adjust the amount of distortion with the Drive knob, or click and drag vertically in the Tracing Model X-Y display. To adjust the distortion’s frequency or ”color,” drag horizontally in the X-Y display or double-click on the Freq field and type in a value. Holding the ALT (PC) / ALT (Mac) modifier while dragging vertically in the X-Y display changes the frequency band’s Q (bandwidth).

The Pinch Effect section adds odd harmonics to the input signal. These distortions typically occur 180 degrees out of phase, creating a richer stereo image. The Pinch Effect has the same controls as the Tracing Model, but generates a rather different sound.

The Drive control increases or decreases the overall distortion amount created by both the Tracing Model and Pinch.

There are two distortion modes: Soft and Hard. The Soft Mode simulates the sound of a dub plate, while Hard Mode is more like that of a standard vinyl record.

The stereo/mono switch determines whether the Pinch distortion occurs in stereo or mono. Set it to stereo for realistic simulation of vinyl distortions.

The Crackle section adds noise to the signal, with noise density set by the Density control. The Volume control adjusts the amount of gain applied to the noise.

22.34 Vocoder

Vocoder.png
The Vocoder Effect.

(Note: the Vocoder effect is not available in the Intro and Lite Editions.)

A vocoder is an effect that combines the frequency information of one audio signal (called the carrier ) with the amplitude contour of another audio signal (called the modulator ). The modulator source is generally something with a clear rhythmic character such as speech or drums, while the carrier is typically a harmonically-rich synthesizer sound such as a string or pad. The most familiar application of a vocoder is to create ”talking synthesizer” or robotic voice effects.

Vocoders work by running both the carrier and modulator signals through banks of bandpass filters. The output level of each of the modulator’s filters is then analyzed and used to control the volume of the corresponding filter for the carrier signal.

Live’s Vocoder should be inserted on the track that contains the audio material you plan to use as your modulator. The Carrier chooser then provides a variety of options for the carrier signal:

  • Noise uses Vocoder’s internal noise generator as the carrier source. With this selected, an X-Y display is shown which allows you to adjust the character of the noise. The horizontal axis adjusts downsampling. Click and drag to the left to decrease the sample rate of the carrier’s output. The vertical axis adjusts the density of the noise. Click and drag downward to decrease the density.
  • External allows you to select any available internal routing points from the choosers below. This is the option you’ll want for classic ”robot voice” applications.
  • Modulator uses the modulator itself as the carrier. This essentially outputs a resynthesized version of the modulator signal, but allows you to use Vocoder’s sound-shaping controls to adjust the sound.
  • Pitch Tracking enables a monophonic oscillator, which tunes itself to the pitch of the modulator. The High and Low sliders allow you to limit the frequency range that the oscillator will attempt to track. Choose from sawtooth or one of three pulse waveforms and adjust the coarse tuning of the oscillator via the Pitch slider. Pitch tracking is particularly effective with monophonic modulator sources such as melodic instruments or voices. Note that the oscillator only updates its frequency when it detects a clear pitch. It then maintains this pitch until it detects a new one. This means that changing the oscillator’s parameters or causing it to reset (when grouping (see 15.3) Vocoder’s track, for example) can cause unexpected changes in the sound. With polyphonic material or drums, pitch tracking is generally unpredictable (but can be very interesting.)

Particularly when using external carrier sources, a vocoder’s output can sometimes lose a lot of high end. Enabling the Enhance button results in a brighter sound by normalizing the spectrum and dynamics of the carrier.

The Unvoiced knob adjusts the volume of an additional noise generator, which is used to resynthesize portions of the modulator signal that are pitchless, such as ”f” and ”s” sounds.

Sens. sets the sensitivity of the unvoiced detection algorithm. At 100%, the unvoiced noise generator is always on. At 0%, only the main carrier source is used. The Fast/Slow switch adjusts how quickly Vocoder switches between unvoiced and voiced detection.

Vocoder’s large central area shows the levels of the individual bandpass filters. Clicking within this display allows you to attenuate these levels.

The Bands chooser sets the number of filters that will be used. Using more bands results in a more accurate analysis of the modulator’s frequency content, but requires more CPU.

The Range sliders adjust the frequency range over which the bandpass filters will operate. For most sources, a fairly large range works well, but you may want to adjust the outer limits if the sound becomes too piercing or bassy. The BW control sets the bandwidth of the filters. At low percentages, each filter approaches a single frequency. As you increase the bandwidth, you increase the overlap of the filter bands. A bandwidth of 100% is the most accurate, but higher or lower settings can create interesting effects.

The Precise/Retro switch toggles between two types of filter behavior. In Precise mode, all filters have the same gain and bandwidth. In Retro mode, bands become narrower and louder at higher frequencies.

Gate sets a threshold for the filterbank. Any bands whose levels are below the threshold will be silent.

The Level slider boosts or cuts Vocoder’s output.

Depth sets how much of the modulator’s amplitude envelope is applied to the carrier’s signal. At 0%, the modulator’s envelope is discarded. At 200%, only high amplitude peaks will be used. 100% results in ”classic” vocoding.

The Attack and Release controls set how quickly Vocoder responds to amplitude changes in the modulator signal. Very fast times preserve the transients of the modulator, but can cause distortion artifacts.

The Mono/Stereo switches determine how many channels are used for the carrier and modulator. In Mono mode, both the carrier and modulator are treated as mono sources. Stereo uses a mono modulator but processes the carrier in stereo. L/R processes both the carrier and modulator signals in stereo.

The frequencies of the carrier’s filterbank can be shifted up or down via the Formant knob. With voice as the modulator, small Formant changes can alter the apparent gender of the source.

The Dry/Wet control adjusts the balance between the processed and dry signals.

22.34.1 Vocoder Tips

This section explains how to set up the most common Vocoder applications.

Singing Synthesizer

The classic vocoder application is the ”singing synthesizer.” To set this up in Live:

  1. Insert Vocoder in the track that contains your vocal material. You can either use a clip that contains a prerecorded voice clip or, to process a live vocal signal, connect a microphone to a channel on your audio hardware and choose this as the input source (see 14.2) for the track.
  2. Insert a synthesizer such as Analog (see 24.1) in another track. Again, you can either create a MIDI clip to drive this synthesizer or play it live.
  3. Set the vocoder’s Carrier chooser to External.
  4. Select the synthesizer track in the vocoder’s Audio From choosers. (For best results, choose Post FX in the bottom chooser.)
  5. If you’re creating your synthesizer and vocal material in real time, make sure the Arm button is enabled on both tracks.
  6. Play the synthesizer as you speak into the microphone. You’ll hear the rhythm of your speech, but with the timbral character and frequencies of the synthesizer. To hear the vocoded signal alone, solo the voice track so that the ”normal” synthesizer track is muted.

Note: you’ll generally get the best results if your synthesizer sound is bright and rich in harmonics. Try sawtooth-based patches to improve the intelligibility of the voice. For even more brightness and clarity, try adjusting the Unvoiced control and/or enabling Enhance.

Formant Shifter

If the Vocoder is set to use the modulator as its own carrier, it can be used as a powerful formant shifter. To do this:

  1. Set the Carrier chooser to Modulator.
  2. Set the Depth to 100%.
  3. Enable Enhance.

Now experiment with different settings of the Formant knob to alter the character of the source. For even more sound-sculpting possibilities, try adjusting the various filterbank parameters as well.

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