Logic Pro 7 Plug-In Reference
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1 Contents Preface 9 10 Introducing Logic’s Plug-ins About This Manual Chapter 1 13 13 16 19 20 21 Basics Using Plug-ins The Plug-in Window Plug-in Settings Plug-in Automation Plug-ins From Other Manufacturers Chapter 2 23 23 24 Instruments and Effects Effect Plug-ins Instrument Plug-ins Chapter 3 29 29 32 33 37 38 38 38 Equalizer Channel EQ Linear Phase EQ Match EQ Fat EQ Silver EQ DJ EQ Individual EQs Chapter 4 39 39 42 42 43 45 45 47 49 50 52 Dynamic Compressor Silver Compressor Expander
Chapter 5 57 57 62 62 63 64 65 66 Distortion Guitar Amp Pro Distortion Overdrive Bitcrusher Clip Distortion Phase Distortion Distortion II Chapter 6 67 67 70 72 79 92 92 Filter AutoFilter Fuzz-Wah EVOC 20 Filterbank EVOC 20 TO High Cut/Low Cut High Pass/Low Pass Filter Chapter 7 93 93 94 96 Delay Sample Delay Tape Delay Stereo Delay Chapter 8 97 97 98 99 99 101 104 105 106 106 107 Modulation Modulation Delay Chorus Flanger Phaser RingShifter—Ring Modulator/Frequency Shifter Tremolo Ensemble Ro
137 138 Creating Impulse Responses About Convolution Chapter 11 143 143 144 145 147 150 152 153 154 Special Spectral Gate Pitch Shifter II Vocal Transformer Pitch Correction SubBass Denoiser Exciter Stereo Spread Chapter 12 157 157 158 159 160 161 163 166 166 Helper Test Oscillator Tuner Gain I/O Direction Mixer Multimeter Correlation Meter Levelmeter Chapter 13 167 167 168 168 169 169 Vocoder—Basics What Is a Vocoder? How Does a Vocoder Work? How Does a Filter Bank Work? Analyzing Speech Signals
202 202 204 205 Global Parameters FM Parameters Modulator and Carrier The Output Section Chapter 18 207 207 ES M Parameters of the ES M Chapter 19 209 209 ES P Parameters of the ES P Chapter 20 213 213 ES E Parameters of the ES E Chapter 21 215 215 ES1 Parameters of the ES1 Chapter 22 223 223 225 283 ES2 Concept and Function The ES2 Parameters Tutorials Chapter 23 301 301 302 309 325 332 340 341 Ultrabeat The Structure of Ultrabeat Overview of Ultrabeat The Synthesizer Parameters Modul
Chapter 25 453 KlopfGeist Chapter 26 455 455 456 460 477 481 482 EVB3 Concepts and Function MIDI Setup The EVB3 Parameters MIDI Controller Assignments Additive Synthesis With Drawbars A Short Hammond Organ Story Chapter 27 485 485 486 502 503 EVD6 The EVD6—Concept and Functions Parameters of the EVD6 Controlling the EVD6 via MIDI A Brief History of the Clavinet Chapter 28 505 505 506 513 516 EVP88 The EVP88—Concept and Functions Parameters of the EVP88 The E-Piano Models Emulated EVP88 and MIDI
Preface Introducing Logic’s Plug-ins The professional Logic music and audio production software features a comprehensive collection of powerful plug-ins. These include; innovative synthesizers, high quality effect plug-ins and authentic recreations of vintage instruments. Logic also supports the use of Audio Unit plug-ins in Mac OS X and also supports TDM plug-ins for users of TDM systems.
Logic’s plug-ins include the following features: • Real-time processing of audio. • Support for sample rates up to 192 kHz. • Altivec optimizations for the Power Macintosh G4 and G5 processors which increase the number of software effects and instruments that can be run simultaneously. • A sophisticated, intuitive, real-time graphical editing interface for most Logic plugins. • A consistent window interface for Logic, Audio Unit and TDM plug-ins.
Conventions of this Guide… Before moving on to the Basics section, we’d like to cover the following conventions used in this manual. Menu Functions For functions that can be reached via hierarchical menus, the different menu levels are described as follows: Menu > Menu entry > Function. Important Entries Some text will be shown as follows: Important: Information on function or parameter. These entries discuss a key concept or technical information that should, or must, be followed or taken into account.
1 Basics 1 This chapter covers all important steps required for plugin use in Logic. The steps include: • Inserting, deleting, and bypassing plug-ins. • Operating plug-ins in the Plug-in window. • Managing plug-in settings. • Automating plug-ins. Using Plug-ins Inserting and Deleting Plug-ins Plug-ins can be either; software instruments, which respond to MIDI note messages, or audio effects, which do not respond to MIDI note messages. • All plug-ins can be added via the plug-in menu of an Audio Object.
To add a plug-in: 1 Click-hold on an Audio Object’s Insert/Instrument slot. 2 The plug-in-menu appears, showing all available plug-ins. Move the mouse through the different levels of the hierarchical menu and choose a plug-in name, then release the mouse button. The Plug-in window is launched automatically. If you do not want the Plug-in window to open automatically after insertion, uncheck the Preferences > Audio > Display > Open Plug-in window on insertion preference.
To remove a plug-in: 1 Click-hold the corresponding Insert/Instrument slot. 2 The plug-in menu is opened. Select the No Plug-In menu option. Inserting Mono/Stereo Plug-ins You can insert mono and stereo effects into Logic’s mono objects. If you use a stereo effect in a mono object, the plug-in menu is limited to stereo effects from this insert point onwards. Note: In general, stereo effects require twice as much processing power as their mono counterparts.
The Plug-in Window Hands-on operation of plug-ins is performed in the Plug-in window. This window allows access to all plug-in parameters. The Plug-in window can be opened by doubleclicking on the blue plug-in label on an Audio Object. Each instance of a plug-in has its own Plug-in window, allowing each to have discrete settings. Operation of Built-in Plug-ins Adjusting Parameters m m m m To toggle a Plug-in window’s buttons: Click on the button.
When changing the Arrange track, an open Plug-in window will update to display the corresponding slot’s plug-in on the newly-selected track. As an example, if the ES1 was loaded on Audio Instrument channel 1, and an EXS24 instance was loaded on Audio Instrument channel 1, switching between these tracks would automatically update the Plug-in window to show the ES1/EXS24, respectively. Bypass The Bypass button allows a plug-in to be deactivated, but not removed from the insert/ instrument slot.
Some Logic plug-ins may have additional parameters that don’t show up on the Editor control panel. This is indicated by an additional 001/011 button next to the Link button. Activate this button to reveal sliders for the extra parameters at the bottom of the Plugin window. Plug-ins With Side Chain Input All plug-ins that support side chain inputs, feature an additional Side Chain pull-down menu in the gray area at the top of the Plug-in window.
Plug-in Settings Logic’s plug-ins ship with a library of ready-to-play preset sounds, known as Settings. These Settings can be found in the Logic > Plug-In Settings subfolder, following the installation procedure. Note: It is strongly recommended that you do not attempt to change the Logic > Plugin Settings folder structure. Within the Plug-in Settings folder you are, however, free to sort your settings into sub folders.
Load Setting This function can be used to load a setting. The file selector box only shows settings for compatible plug-in types. Each plug-in has its own set of parameters, and therefore its own file format. Note: Proprietary plug-in-settings created in Logic for Windows can be read by Logic for Mac OS, and vice versa. Plug-in settings files created on the Mac must be saved with a .pst file extension in order for them to work in Logic for Windows.
Plug-ins From Other Manufacturers Audio Unit Support Correctly installed third-party Audio Unit plug-ins (Effects and Instruments) can be used in Logic. Clicking on an Audio object insert/instrument slot will launch the hierarchical Plug-In menu. A separate Audio Units submenu displays all installed Audio Unit plug-ins. TDM Plug-ins Users of a Digidesign TDM system can utilize TDM plug-ins in Logic.
2 Instruments and Effects 2 This chapter explains the difference between effect and instrument plug-ins. Instrument plug-ins respond to MIDI note messages, effect plug-ins do not. Therefore instrument plug-ins can only be inserted into special Audio Objects, called Audio Instruments. Effect Plug-ins Logic’s effects can be installed into all insert slots of all Audio Object types (See “Inserting and Deleting Plug-ins” on page 13.). This allows processing of all audio and instrument signals.
Within Logic, the effect is placed in an insert slot of a bus object. The signals of the individual tracks can each be sent to the bus, controlled by a Send knob. The audio signal is then processed with the effect, and mixed with the stereo output. The advantage of this “bussed” approach, over inserting effects on tracks, is efficiency.
3 Double click the newly-created Audio Object icon, so that the (grayed out) channel strip appears. 4 Now, go to the Object Parameter box, and set the Channel parameter to an Instrument. The generic Audio Object will now operate as an Audio Instrument, allowing you to insert any Instrument plug-in into the instrument slot.
Logic’s Bounce function allows the entire Audio Instrument track to be recorded as an audio file. This “Bounced” audio file can then be assigned (as an audio region) to a standard Audio track, allowing you to reassign the available processing (CPU) power for further synthesizer tracks. For details, please refer to the Bounce chapter in the Logic Reference manual. You can also make use of the Freeze function to capture the output of an Audio Instrument track, again saving processing power.
Software Instrument Pitch The Song Settings > Tuning > Software Instrument Pitch > Tune parameter remotely controls the main tuning parameter for all software instruments (plug-in synthesizers, such as the ES1 or EXS24 sampler and others) by ±100 cents. Note: Some instruments do not recognize this remote command.
3 Equalizer 3 This chapter covers all Logic equalization effects. Equalizers allow you to increase or decrease the level of selected components in the overall audio spectrum. Logic’s built-in equalizers include the Channel EQ, Linear Phase EQ, Match EQ, Fat EQ, Silver EQ, DJ EQ, High/Low Pass Filters, High/Low Cut EQ, Parametric EQ and High/Low Shelving EQ plug-ins. Channel EQ The extremely high-quality Channel EQ offers eight frequency bands and an integrated FFT analyzer.
Bands 2 and 7 are defined as shelving equalizers. Note: When the Q parameter of band 2 and 7 is set to an extremely high value (to 100, for example), the equalizers only apply to a very narrow band, and can work in a fashion that is similar to notch filters. Bands 3to 6 are bandpass filters. You can set the band parameters either in the parameter area or directly in the central EQ display. Move the mouse horizontally over the display.
Use the Scales to the left and right of the EQ display, to change the vertical scale of the EQ and analyzer curves. To increase the resolution of the EQ Gain parameter (dB Warp) in the most interesting area around the zero line, click-hold in the green dB Scale on the left side of the graphic display, and move the mouse up. Moving the mouse down, will decrease the parameter value. The overall range is always ±30, but small values will be easier to recognize.
Linear Phase EQ The extremely high-quality Linear Phase EQ plug-in is almost identical to the Channel EQ. With the exception of the different name and a few different colors, it uses the same familiar eight-band layout, and method of operation, as the Channel EQ.
Match EQ The Match EQ plug-in allows you to “match”, and transfer the frequency spectrum from one signal to another, or to store it as a spectral template file. In this way, you can acoustically match the sound of various songs for an album, or impart the “sound” of any reference source onto your own recordings. The alignment of signals is automatic, but you can also manually draw or modify the filter curve to alter the sound as required. Note: Match EQ acoustically matches two audio signals.
Parameters The View pull-down menu allows you to select the type of information shown on the analyzer display in the center. The following options are available: • Automatic: Depending on the selected function, the analyzer view is automatically toggled between the three following options. • Template: The analyzer display shows the average frequency curve, which is generated by analyzing the input signal or loading a template.
Note: Audio files can also be dragged onto the Template or Current Material Learn buttons to generate template or current spectra. A progress bar displays the progress of the analysis process. If you right-click (or Control click) on either of the Learn buttons, a context menu opens. This menu allows the spectrum of the template or the track signal (Current Material) to be: • cleared • copied to the Match EQ clipboard, which is common to all Match EQ instances in the current song.
The filter curve can be edited via the Smoothing slider. At a value of 0.0, the filter curve is applied to the track signal without any changes. At all other Smoothing settings, the filters are smoothed at a constant bandwidth. A value of 1.0, for example, means that all filters have a constant bandwidth of one semitone that is used to smooth the notchlike filters in the curve. A bandwidth of: four semitones (a value of 4.0—or a major third), an octave (a value of 12.
If the resulting filter curve is displayed, the left scale—and the right, if the analyzer is inactive—shows the dB values for the filter curve in an appropriate color. By clickdragging on one of the scales, the overall gain of the filter curve is adjusted in the range from −30 to +30 dB. Fat EQ The high-quality Fat EQ offers up to 5 fully parametric bands—buttons 1 through 5 activate these individually; inactive bands do not drain your computer’s resources.
At low Q values, the EQ influences a wider frequency range, and at high Q values, the effect of the EQ band is limited to a very narrow frequency range. Please bear in mind that your perception of an attenuated or boosted frequency depends heavily on the Q parameter: If you’re working with a narrow frequency band, you’ll generally need to cut or boost it more drastically to notice a difference.
4 Dynamic 4 This chapter introduces Logic’s Dynamic plug-ins. This includes the Compressor, Silver Compressor, Expander, Noise Gate, Silver Gate, Enveloper, DeEsser, Limiter, Multipressor, and Adaptive Limiter plug-ins. Compressor A compressor tightens up the dynamics of a signal. This means that the difference in levels between loud and soft passages is reduced.
Logic’s Compressor was designed to emulate the response of the finest analog compressors. It follows the following principle: When a signal exceeds the defined Threshold level, the compressor actually alters the response, so that it is no longer linear. What happens is that all levels that exceed the Threshold are attenuated by the value set with the Ratio slider.
When you have configured a compressor so that it dampens the signal at or above the Threshold value by the predetermined Ratio, while the level just below the Threshold is routed through at a 1:1 Ratio, an audio engineer would term the compression as hard knee. In many cases, however, you’ll come up with a better sounding track by using a more gradual transition from the 1:1 Ratio below the Threshold, to the Ratio that you entered for levels above the Threshold.
Silver Compressor The Silver Compressor is a simplified version of the Compressor. It is limited to Threshold, Attack, Release, and Ratio controls. Expander The Expander is similar to the Compressor, with one fundamental difference—it increases, rather than reduces, the dynamic range above the Threshold. The Ratio slider features a value range of 1:1 to 0.5:1.
Noise Gate Ordinarily, a noise gate suppresses unwanted noise that may become audible during a lull in the signal. You can, however, also use it as a creative sound-sculpting tool. Here’s the basic principle behind a noise gate: Signals that lie above the Threshold are allowed to pass unimpeded (open gate). Anything below the defined Threshold (background noise, crosstalk from other signal sources and so on) is fully muted (a closed gate).
Let’s back up a bit for a brief explanation: Noise gates often begin chattering when the level of a signal fluctuates slightly, but very rapidly, during the attack or release phase. Instead of clearly exceeding or falling short of the Threshold value, the signal level hovers around the Threshold. The Noise Gate then rapidly switches on and off to compensate, producing the undesirable chattering effect.
Silver Gate The Silver Gate is a cut-down version of the Noise Gate. It is limited to Threshold, Lookahead, Attack, Hold, and Release controls. Enveloper The Enveloper is an unusual tool that lets you shape transients—the attack and release phases of signals. No other type of dynamic effect (such as a compressor or expander) can achieve similar results—and these results can be quite impressive indeed.
Emphasizing the release also boosts the amount of any reverb on the affected track. Conversely, when you tone down the release phase, tracks originally drenched in reverb end up sounding drier. This effect is particularly useful when you’re working with drumloops, but there are, of course, many other applications. Let your imagination be your guide. When using the Enveloper, you should set the Threshold to the minimum value and leave it there.
DeEsser A DeEsser is a signal processor used for the rejection of hissing, or sibilant noises. This is why it is called a “DeEsser”, and occasionally an “S”-Suppressor. You can, of course, reject sizzling frequencies with an equalizer, but a DeEsser only rejects this high frequency band for as long as a threshold level is being exceeded in a specific frequency band. This “dynamic” ability is why the sound doesn’t get darker when no “sizzling” consonants are present in the signal.
Parameters of the DeEsser Detector Frequency This parameter defines the frequency band the DeEsser acts upon. It’s not necessarily the same band that will be reduced. Detector Sensitivity This parameter defines the threshold level that needs to be exceeded (around the Detector Frequency), in order to reduce the level around the Suppressor Frequency. Monitor Activation of this switch allows you to monitor the Side Chain signal used by the DeEsser.
Limiter The Limiter is also a standard effect for processing a summed stereo signal. It is normally used for mastering. You could say that a limiter is a compressor with an infinite compression ratio. The Limiter restricts dynamics to an absolute level. Any input level that exceeds the Limiter’s threshold (Gain) will be output at this “limited” level, no matter how much higher the original signal level may have been.
Release Here, you can set the time required by the Limiter (after limiting) to release the effect. Output Level This simple volume control sets the desired maximum level of the Limiter’s output signal. Softknee Activate the Softknee button to produce a softer transition from no limiting to full limiting. If switched off, the signal will be limited (following a linear curve) absolutely and exactly when a level of 0 dB is reached. If switched on, the transition to full limiting is non-linear, meaning softer.
Following the Adaptive Limiter process, tracks normalized to 0 dB appear to sound about 2 dB louder, depending on the source signal. As with other Logic dynamic processes, this plug-in also features a lookahead facility (the Lookahead parameter can be set in Controls view, allowing the Adaptive Limiter to look into the future. The Adaptive Limiter reacts to level peaks in signals streaming from the hard disk before they are played back, and delaying the monitored signal.
Multipressor The Multipressor (an abbreviation for multiband compressor) is the epitome of an audio mastering tool. It’s a pretty complex tool; good sounding settings require quite a lot of listening experience. Functional Principle of Multi-band Compressors The multi-band compressor splits the incoming signal into two to four different frequency bands before applying compression. These frequency bands are then compressed independently. After compression, the frequency bands are mixed back together.
Multipressor Parameters Bands This parameter (on the right side) determines the number of independently compressible frequency bands, and has a crucial impact on the amount of computing power needed for the effect. Classic multi-band compressors use three Bands. Lookahead Lookahead (just below the Bands parameter) determines how far the processor looks into the future, in order to react earlier (thus better) to peak volumes.
• Threshold Display The horizontal lines (up to three) in the lower area of the window represent the Threshold values for Compression (upper line), Expansion (middle line), and Reduction (bottom line). You can set these values by using the controls of the same name (see below). Note: You can select the frequency band that you wish to edit by clicking in the lower section of each band. Comp. Ratio This, in conjunction with Compression Threshold, is the central parameter for compression. The Comp.
Level Meter In the Level Meter to the right, you may monitor either; the change of level caused by compression, or the output volume of each band, depending on whether you have selected Gain change or Output (see below). You can individually switch the bands on and off, in order to listen to single bands, by using the switches below the meters. If the switch below a band’s meter is lit (light green), the band will be audible. If a switch is unlit, the band is muted.
5 Distortion 5 This chapter introduces you to Logic’s distortion effects. This includes the Distortion, Overdrive, Bitcrusher, Clip Distortion, Phase Distortion, Distortion II, and Guitar Amp effect plug-ins. Guitar Amp Pro The Guitar Amp Pro plug-in simulates the sound of several famous guitar amplifiers and a number of cabinets/speakers. You can process guitar signals directly within Logic, allowing you to reproduce the sound of high-quality guitar amplification systems.
The Guitar Amp Pro panel is divided into three areas. The upper part of the raised Yshaped user interface contains the Amplifier parameters section. The bottom of this panel houses the Effect and Output section of the plug-in. The Speaker sections to the left and right provide access to the miking parameters of the virtual speaker. Amp Section In the upper area of the Amp section, you will find three pull-down menus.
• US 1×10 open back—Not much resonance in the low frequency range. Suitable for use with (blues) harmonicas. • US 1×12 open back 1—Open enclosure of an American lead combo with a single 12" speaker. • US 1×12 open back 2—Open enclosure of an American clean/crunch combo with a single 12" speaker. • US 1×12 open back 3—Open enclosure of another American clean/crunch combo with a single 12" speaker. • US broad range—Cabinet simulation of a classic electric piano speaker.
To the extreme right of the Guitar Amp Pro GUI, you will find the Master knob, which controls the output volume of the amplifier (to the speaker). Typically, in tube amplifiers, an increase in the Master control level produces a self-compressed and saturated sound, along with increased level, resulting in a more distorted and powerful amp signal. In the analog domain, this results in an extreme increase in loudness.
Effect Section The Effect section of Guitar Amp Pro contains the Tremolo and Vibrato effects— essentials in any guitar rig, and the Reverb. Note: The Effect section is placed before the Master control in the signal flow, and therefore receives the preamplified signal (pre-Master). In order to configure the Effect section, you must activate it via the On/Off buttons, found to the lower left of the FX and Reverb panels.
Distortion This distortion effect simulates the lo-fi dirt generated by a bipolar transistor. Move the Drive slider up to increasingly saturate the transistor. Generally, the distortion created by the plug-in tends to increase the signal level, an effect that you can compensate for with the Output slider. The Tone knob filters the harmonics-laden distortion signal, delivering a somewhat less grating, softer tone. The Distortion Eye is watching—it visually represents the Drive and Tone parameter settings.
Bitcrusher Bitcrusher is the ultimate digital distortion box. You can do all kinds of wild stuff with it, such as recreate the 8-bit sound of the pioneering days of digital audio, create artificial aliasing by dividing the sample rate, or distort signals so radically that they are rendered unrecognizable. Warning: The Bitcrusher can damage your hearing (and speakers) when operated at high volumes. The Drive slider boosts the level at the input of the Bitcrusher.
Clip Distortion The Clip Distortion plug-in is a non-linear distortion effect that produces unpredictable spectra. Beyond drastic distortions, it’s well suited for the simulation of warm tube overdrive sounds. The best way to learn what effect the various parameters have is to experiment with them on different signal sources. As a starting point, the following describes what each control basically does: The signal is first amplified by the Drive value, which is a simple gain control.
Phase Distortion The Phase Distortion plug-in is based on a modulated delay line, much like the wellknown chorus or flanger effects. As opposed to these, the delay time is not modulated by a low frequency oscillator (LFO), but rather by a lowpass-filtered version of the audio input signal itself. This is how the signal modulates its own phase position. In the signal flow of this effect, the parameters do the following: The input signal only passes the delay line and is not affected by any other process.
Distortion II The Distortion II plug-in is based on the EVB3’s distortion effect. More information about its parameters can be found in the EVB3’s “Distortion” section on page 473. The Distortion II plug-in offers one additional parameter: the PreGain knob. This allows you to raise the input signal by up to 20 dB or lower it by as much as 10 dB, in order to provide a broader range of distortion colors.
6 Filter 6 This chapter covers Logic’s filter effects. The filter effects include the AutoFilter, Fuzz-Wah, EVOC 20 FB, EVOC 20 TO, Low/High Pass Filter and Low/High Cut plug-ins. The EVOC 20 TO is based on a vocoder. Further information about vocoders can be found in the chapter “Vocoder—Basics” on page 167. AutoFilter The AutoFilter is an extremely versatile, resonance-capable lowpass filter, that offers a couple of truly unique features.
The Resonance knob emphasizes the frequency range surrounding the cutoff frequency. When you turn the Resonance up sufficiently, the filter itself begins oscillating (at the cutoff frequency). Self-oscillation is initiated before you max out the Resonance parameter, just like the filters on the legendary Minimoog.
LFO: the wave shape used for LFO oscillation is determined by the Waveform buttons. The choices are: descending sawtooth (saw down), ascending sawtooth (saw up), triangle, pulse wave, or random (random values, Sample & Hold). Once you’ve selected a waveform, you can shape the curve with the Pulsewidth knob. Use the Frequency knobs to define the desired LFO frequency: Coarse sets a value between 0.1 and 10,000 Hz, Fine lets you adjust it in smaller increments. The Speed Mod.
Fuzz-Wah The Fuzz-Wah effect is the standalone plug-in version of the EVD6’s Wah effect. It incorporates additional compressor and distortion (Fuzz) facilities, and features some additional parameters over the integrated EVD6 Wah. These are outlined below. Parameters of the Fuzz-Wah FX Order This parameter allows to you select the order in which the Fuzz/Wah effects are placed. Choices are: Fuzz –Wah or Wah–Fuzz.
Warning: Please take care while doing this, or your ears and speaker system may be damaged. Relative Q The quality of the main filter peak can be increased/decreased, relative to the model setting, thereby obtaining a sharper/softer wah sweep. When set to a value of 0, the original setting of the model is active. Range: −1.00 to +1.00 (0.00 is the default) Pedal Range Common MIDI foot pedals have a much larger mechanical range than most classic Wah pedals.
AutoWah Attack/Release These parameters allow you to define how much time it takes for the Wah filter to open and close. Range (in milliseconds): 10 to 10,000 Comp Ratio The Comp Ratio of the integrated compressor can be adjusted between 1:1 (no compression) and 30:1. The Compressor is tied to the Fuzz effect, and always precedes it. As such, the FX Order parameter is very important for placement of the Compressor in the effects chain. Fuzz Gain Controls the level of Fuzz (distortion).
Parameters of the EVOC 20 FB The EVOC 20 FB interface is divided into three main sections. These are the Formant Filter, Modulation, and Output areas. The Formant Filter Area The Formant Filter Window The Formant Filter window is divided into two sections by a horizontal line. The upper half applies to the Filter Bank A, and the lower half to the Filter Bank B. The individual vertical bars in each bank of settings are faders which represent the level of a particular frequency band/formant.
Note: Increasing the number of bands also increases the processor overhead. High/Low Frequency The blue bar shown just beneath the EVOC label is a multi-part control which is used to determine the lowest and highest frequencies allowed to pass by the filter. Frequencies which fall outside these boundaries will be cut. All filter bands are distributed evenly across the range defined by the High/Low Frequency values.
Note: Boost is also quite handy to adjust the levels of both filter banks to each other, so that using Fade A/B (see below) leads only to a sound color change, but not to a level change. Fade AB Control The Fade AB crossfades between the A and B filter bank. At its extreme top or bottom position, you will only hear one of the filter banks. Formant Shift Moves the position of all bands in both filter banks up and down the frequency range. Note: You can jump directly to the values −0.5, −1, 0, +0.5 and +1.
Waveform The Waveform switches allow the selection of the waveform type used by LFO Shift and LFO Fade. A selection of Triangle, falling and rising Sawtooth, Square up and down around zero (bipolar), Square up from zero (unipolar), a random stepped waveform (S&H), and a smoothed random waveform is available for each LFO. Intensity The Intensity sliders control the amount of Formant Shift and Fade A/B modulation by the respective LFOs. Rate These knobs determine the speed of the modulation.
Output Section Overdrive This switch enables/disables the Overdrive circuit of the EVOC 20 FB. Note: To actually hear the Overdrive effect, you may need to boost the level of one or both filter banks. Level The Level slider controls the level of the EVOC 20 FB’s output signal. Stereo Mode This pull-down menu determines the input/output mode of the EVOC 20 FB. Choices are: m/s—mono input to stereo output and s/s—stereo input to stereo output.
MIDI Controllers Received The following tables show the CC numbers used when the following MIDI preference is active: Options > Settings > MIDI Options > (Version 4.x behavior).
Filter Bands Band Level Bank A LFO 1 (Shift) LFO 1 Rate CC #93 LFO 1 Waveform Select CC #94 LFO 1 Intensity CC #95 LFO 2 Rate CC #121 LFO 2 Waveform Select CC #122 LFO 2 Intensity CC #123 Level CC #124 LFO 2 (Fade) Output Mono/Stereo Select CC #87 Stereo Width CC #86 Bank B EVOC 20 TO The EVOC 20 TO is a vocoder equipped with a monophonic pitch tracking oscillator, hence the TO in its name (more information about vocoders can be found in the chapter “Vocoder—Basics” on page 167).
The EVOC 20 TO can be used in the insert slots of Audio, Audio Input, Bus, Master, and Audio Instrument channels. The signal path of the EVOC 20 TO is shown in the block diagram on page 190. Parameters of the EVOC 20 TO The EVOC 20 TO interface is divided into five main sections. From left to right, these are the Analysis/Synthesis, Formant/Filter, Modulation, Unvoiced/Voiced (U/V) Detection and Output areas.
Freeze The Freeze button holds the current analysis sound spectrum indefinitely. The “frozen” Analysis signal can capture a particular characteristic of the source signal, which is then imposed as a complex sustained filter shape on the Synthesis section. Using a spoken word pattern as a source, for example, the Freeze parameter could capture the attack or tail phase of an individual word within the pattern—the vowel a, for example.
• Track—allows you to use the audio track, into which the EVOC 20 TO is inserted, as the Synthesis source signal. • Side Chain—allows you to use another audio track as the source material for the Synthesis section. Selection of the Side Chain track is achieved by click-holding on the Side Chain pull-down menu in the gray area at the top of the Plug-in window. Note: If no Side Chain track is assigned in the Side Chain pull-down menu, the track’s signal will be used.
FM Int This knob selects the basic waveform and controls the intensity of FM modulation. • If set to 0, the FM tone generator is disabled, and a sawtooth wave is generated instead. • If set to values higher than 0, the FM tone generator is activated. Higher values result in a more complex and brighter sound. FM Ratio The FM Ratio (value range 0.5 to 3.5) knob defines the ratio between the Carrier and Modulator frequencies—the frequencies of Oscillators 1 and 2.
• Strength—determines how pronounced the automatic pitch correction is. • Glide—determines the amount of time the pitch correction takes, allowing sliding transitions to the quantized pitches. Root/Scale The Root and Scale parameters, in combination with the onscreen keyboard, define the pitch(es) that the tracking oscillator is quantized to. • If you click on the value shown below the word Scale, a pull-down menu opens. Here • • • • you can select a scale or chord.
Max Track This parameter cuts the high frequencies of the analysis signal, making the pitch detection more robust. Should the pitch detection produce unstable results, reduce the Max Track parameter value to the lowest possible setting. Formant Filter The Formant Filter Window The Formant Filter window is divided into two sections by a horizontal line. The upper half applies to the Analysis section and the lower half to the Synthesis section.
Lowest/Highest These parameters can be found in the two small fields on either side of the Formant Filter window. These switches determine whether the lowest and highest filter bands are bandpass filters (just like all the bands between them), or whether they act as lowpass/highpass filters, respectively. Click once on them to switch between the two curve shapes available. • In the bandpass setting, the frequencies below/above the lowest/highest bands are ignored on analysis and synthesis.
Note: Formant Stretch and Formant Shift are especially useful if the frequency spectrum of the Synthesis signal does not complement the frequency spectrum of the Analysis signal. You could create a Synthesis signal in the high frequency range from an Analysis signal which mainly modulates the sound in a lower frequency range, for example.
Rate This knob determines the speed of the modulation. Values to the left of the center positions are synchronized with the sequencer’s tempo and include bar values, triplet values and more. Values to the right of the center positions are non-synchronous and displayed in Hertz (cycles per second). Note: The ability to use synchronous bar values could be used to perform a formant shift every four bars on a one bar percussion part, which is being cycled.
Mode Here, you select the sound source(s) which can be used to replace the unvoiced content of the input signal. Possible settings are Off, Noise, Noise + Synth, or Blend. • Noise—uses noise alone for the unvoiced portions of the sound. • Noise + Synth—uses noise and the synthesizer for the unvoiced portions of the sound, • Blend—uses the Analysis signal after it has passed through a highpass filter, for the unvoiced portions of the sound.
Level Level controls the volume of the EVOC 20 TO’s output signal. Stereo Mode This pull-down menu determines the input/output mode of the Synthesis filter bank. Choices are: m/s—mono input to stereo output and s/s—stereo input to stereo output. Note: The Stereo Mode should be set to m/s if the signal going into the Synthesis filter bank is monophonic or Synthesis In is set to Osc. Note: Stereo/stereo (s/s) is the preferred setting for stereo Synthesis input signals.
MIDI Controllers Received The following tables show the CC numbers used when the following MIDI preference is active: Options > Settings > MIDI Options > (Version 4.x behavior).
Root/Scale kyb. C CC #91 C# CC #92 D CC #93 D# CC #94 E CC #95 F CC #96 F# CC #97 G CC #98 G# CC #99 A CC #100 A# CC #101 B CC #102 High Cut/Low Cut • The Low Cut filter attenuates the frequency range below the selected frequency. • The High Cut filter attenuates the frequency range above the selected frequency. High Pass/Low Pass Filter • The High Pass Filter affects the frequency range below the set frequency. Higher frequencies pass through the filter.
7 Delay 7 This chapter describes Logic’s delay effects. This includes the Sample Delay, Tape Delay and Stereo Delay plug-ins. Sample Delay This plug-in allows the simple delaying of a channel by single sample values. The stereo version of the plug-in provides separate controls for each channel. This plug-in (when used in conjunction with the phase inversion capabilities of the Gain plug-in) is particularly suited to the correction of run-time problems that may occur with multichannel microphones.
Tape Delay The Tape Delay simulates a vintage tape echo device, although with some very useful features that such old devices never offered. The first of these is that it’s delay settings are variable in musical increments. It is equipped with a highpass and lowpass filter in the feedback circuit, as well as a circuit that simulates tape saturation effects. This plugin is ideal for the dub delays invented by Jamaican toast masters, and used in many styles of music today.
You can shape the sound of the echoes, using the on-board highpass and lowpass filters. Although these filters are fairly flat, they’re not located post-output. They are located in the feedback circuit, meaning that the effect achieved by these filters increases in intensity with each repeat. If you’re in the mood for an increasingly muddy tone, move the High Cut filter slider towards the left. For ever thinner echoes, move the Low Cut filter slider towards the right.
Stereo Delay The Stereo Delay works much like the Tape Delay, which is why we’ll skip the general info, and take a closer look at the differences between the two. There is just one Stereo Delay (s/s), hence the stereo input and output. You are free to use the Stereo Delay for monaural tracks or busses, when you want to create independent delays for the two stereo sides. Please bear in mind that if you use this option, the track or bus has two channels from the point of insertion forward.
8 Modulation 8 This chapter introduces Logic’s modulation effects. This includes the Modulation Delay, Chorus, Flanger, Phaser, Ring Modulator, Tremolo, Ensemble, Rotor Cabinet, Scanner Vibrato, and Spreader plug-ins. Modulation Delay As its name implies, the Modulation Delay generates effects such as flanging or chorus, based on modulated short delays. It can also be used—without modulation—to create resonator or doubling effects.
Set the basic delay time with the Flanger-Chorus knob. Set to the far left position, the Modulation Delay puts on its flanger cap. As you move towards the center position, it thinks it’s a chorus. As you move the knob closer to the far right position, you will hear clearly audible delay taps. This latter type of setting is generally used without modulation (Width = 0), for doubling effects. The Stereo Phase knob defines the phase of the modulation between the left and right stereo sides.
Flanger The Flanger works in a similar fashion to the Chorus, but with a shorter delay time, and the output signal being fed back into the input of the delay line. Use the Intensity slider to determine the Flanger’s modulation width. Speed sets the frequency of the modulation. Feedback determines the amount of the delayed signal that is routed back into the input. Negative values invert the phase of the routed signal. The Mix slider determines the balance of dry and wet signals.
The modulation section offers two LFOs, featuring individually variable frequencies, and freely variable mix options (LFO Mix). Additionally, the frequency of LFO 1 can be modulated by the level of the input signal. Use the Envelope Modulation slider to set the desired modulation intensity. By staking out the limits of the modulation with its highest and lowest values, you can determine the modulation width and range.
RingShifter—Ring Modulator/Frequency Shifter Logic’s RingShifter plug-in combines a ring modulator with a frequency shifter effect. These two related effects are based on modulation of the signal amplitude. Both effects were popular during the 1970’s, and are currently experiencing something of a renaissance. The ring modulator, for example, was extensively used on jazz rock and fusion records in the early 70’s. The frequency shifter was, and still is, found as part of many modular synthesizer systems.
Modes The four Mode buttons determine whether the plug-in either operates as a frequency shifter or as a ring modulator. The frequency shifter offers the Single and Dual settings. The ring modulator provides the OSC and Side Chain settings. • Single (Frequency Shifter): The frequency shifter generates a single shifted effect signal. The position of the large Frequency rotary control determines whether the signal is shifted up (positive value) or down (negative value).
Delay The effect signal is routed through a delay, following the oscillator. The Level control sets the level of the delay added to the ring modulated or frequency shifted signal. Note: A Level value of 0 passes the effect signal directly to the output (bypass). The Time control sets the delay value from 0 to 2,000 milliseconds. Activation of the Sync button synchronizes the delay to your Logic song tempo, in musical note values.
The LFO is the second modulation source. It is activated/deactivated via its own Power button. The LFO produces continuously cycled control signals. The LFO waveform can be shaped as required via the Symmetry and Smooth sliders. The LFO waveform display provides visual feedback of the resulting waveform. The Rate rotary control sets the cycle speed of the LFO. Press the Sync button if you want to synchronize the LFO cycles with the Logic song tempo (using musical note values).
Ensemble The Ensemble is like a pitch shifter on steroids—it consists of eight internal, modulatable pitch shifters. Two standard LFOs and one random LFO enable you to come up with fairly complex pitch modulations, which—much like a natural chorus effect—conjure up the impression of an instrumental or vocal ensemble. The Ensemble’s graphic visually represents the number of voices, and their modulations. Use the Voices slider to determine how many voices (1 to 8) are generated, in addition to the original.
Rotor Cabinet The Rotor Cabinet plug-in is based on the EVB3’s Rotor Cabinet effect section. A detailed description of it’s parameters can be found in the EVB3’s “Rotor Cabinet” section on page 474. Please note: There is no Speed Control parameter on the Rotor Cabinet plug-in. You can switch rotor speeds manually.
Spreader The Spreader plug-in widens the stereo spectrum with an effect that is quite similar to the Chorus effect. The frequency range of the original signal is periodically shifted in a non-linear way. In comparison to the Stereo Spread effect, the perceived pitch changes. Use the LFO Intensity parameter to set the modulation width of the Spreader. LFO Speed controls the modulation frequency. Channel Delay determines the delay time in Samples. Mix sets the balance of dry and wet signals.
9 Reverb 9 This chapter describes Logic’s reverb effects. This includes AVerb, SilverVerb, GoldVerb, PlatinumVerb, Enverb, and Space Designer. Space Designer is Logic’s only convolution reverb, and is described separately in the “Convolution Reverb: Space Designer” chapter, from page 117. AVerb Although the AVerb is based on a simple reverb algorithm, it delivers remarkably good results.
Where high Pre Delay settings tend to generate something similar to an echo, low values often muddy the original signal. Ideally, you should go for as high a setting as possible before the plug-in begins generating something that sounds like a tap delay. With appropriate Pre Delay settings, you can apply relatively generous amounts of reverb to percussive parts, while retaining definition on the attack portions of the sounds.
The Modulation Rate, Modulation Int and Modulation Phase parameters control an additional modulation delay. It consists of two LFOs with variable frequencies (set with Modulation Rate). The desired modulation width is set with the Modulation Int slider. When this slider is set to the far right position, delay modulation is switched off completely. The Modulation Phase knob defines the phase of the modulation between the left and right stereo sides.
Predelay Predelay is the amount of time that elapses between the original signal, and the arrival of the early reflections. In any given room size and shape, Predelay determines the distance between the listener and the walls, ceiling, and floor. When used with artificially generated reverb, it has proven advantageous to allow this parameter to be manipulated separately from, and over a greater range than, what is considered natural for Predelay.
Density This parameter controls the density of the diffuse reverb. Ordinarily, you want the signal to be as dense as possible. However, less Density means the plug-in eats up less computing power. Moreover, in rare instances, too great a Density can color the sound, which you can fix simply by reducing the Density knob value. Conversely, if you select a Density value that is too low, the reverb tail will sound grainy. Diffusion (Controls View) Diffusion sets the diffusion of the reverb tail.
PlatinumVerb The difference between the PlatinumVerb and the GoldVerb is the former’s enhanced Reverb section. The Early Reflections sections of the two plug-ins are identical. For more information, please read the “GoldVerb” section, on page 111. We’ll focus on the additional features offered by the PlatinumVerb in this section. The Reverb section of the PlatinumVerb is based on a genuine dual-band concept.
In the vast majority of mixes, your best bet is to set a lower level for the low frequency reverb signal. This enables you to turn up the level of the bass instrument—making it sound punchier. This also helps to counter bottom-end masking effects. The Controls view offers four additional parameters. ER Scale allows you to scale the early reflections along the time axis, enabling the Room Shape, Room Size and Stereo Base parameters to be influenced simultaneously.
Predelay This is the delay between the (undelayed) original signal, and the starting point of the reverb attack phase. Attack This is the amount of time it takes for the reverb to climb to its peak level. Decay This is the amount of time it takes for the level of the reverb to drop from its peak to the sustain level. Sustain This is the level of the reverb that remains constant throughout the sustain phase. Hold This is the duration (time) of the sustain phase.
10 Convolution Reverb: Space Designer 10 This chapter introduces you to Logic’s Space Designer Reverb effect. Space Designer is a convolution reverb plug-in. Reverberation is generated by means of a real-time convolution process, using any loaded impulse response (IR) recording (reverb sample). Put another way, an IR recording of an actual real-world room, a vintage plate or spring reverb unit, for example. The result is an exceptionally realistic reverb/room sound.
Using Space Designer Inserting Space Designer As per all effect plug-ins in Logic, Space Designer can be inserted into any Audio Object’s insert slot. In doing so, you should take the following information into consideration: • Set the Direct Output parameter to a value of 0 (mute), if Space Designer is inserted into a bus channel. This happens by default in Logic, but should you change it (accidentally or by design), and things don’t sound right, this is the first thing to look at.
Note that parameter changes occur after the release of the mouse button, and after an additional grace period has elapsed, which is indicated by a blue bar. The calculation itself requires a certain amount of processing time—depending on the speed of your Macintosh CPU(s). During this calculation time, no other parameters can be adjusted. A (red) progress bar below the Length parameter panel will advise you of the calculation status.
IR Sample When the IR Sample button is initially clicked, an operating system File Selector dialog will be launched, allowing you to select the desired Impulse Response file from a folder on your hard disks or CD. If you have already loaded an IR file, this button is simply used to switch back from Synthesized IR to IR Sample mode. To change the Impulse Response click the downward pointing arrow to the right of the button.
Sample Rate This parameter determines the sample rate of the Impulse Response. By default the current Logic song sample rate is used by Space Designer as well (if the Logic song is running at 96 kHz, Space Designer uses the same rate). When loading an Impulse Response, Space Designer automatically converts the sample rate of the IR to match the current Logic song sample rate—should it be necessary. As an example, this allows you to load a 44.
Note: If running Space Designer in a song at 96 kHz (utilizing an Impulse Response originally recorded at 44.1 kHz), you may want to reduce the IR Sample Rate to 1/2. To do so, use the Sample Rate Slider in Space Designer. Make sure the Preserve Length function is enabled. This cuts CPU power consumption by about 50%, without compromising reverb quality. There is no loss in reverb quality, because the Impulse Response—originally recorded at 44.
Global Parameters Input (Crossfeed) This parameter allows a stereo input signal to be: • processed on both channels, retaining the stereo balance of the original signal—top of slider, • to be processed in mono—middle of slider, • to be inverted, with processing for the right channel occurring on the left and vice versa—bottom of slider. • a mixture of stereo to mono cross feeds (in-between positions) Note: This slider is not available when Space Designer is used as a mono plug-in.
Rev Vol Compensation Rev Vol Compensation (Reverb Volume Compensation) attempts to match the perceived (not actual) volume differences of Impulse Response files. It is set to on by default, and should generally be left in this mode, although you may find that it isn’t successful with all types of Impulse Responses. In such situations, switch it off and adjust the Input and Output levels accordingly.
The IR Start parameter can for example be used to eliminate any peaks at the beginning of the IR sample. It also offers a number of creative options, such as its use when combined with the Reverse function. Note: The IR Start parameter is not available in Synthesized IR mode. In the Synthesized IR mode this parameter is not required as, by design, the Length parameter provides identical functionality.
Envelope Window Node Handling Before we get to the buttons, we’d like to briefly touch on the Envelope window of the Space Designer GUI. When first launched, a default synthesized Impulse Response and set of Envelopes are automatically created. You will see a few nodes placed around and within the Volume Envelope that is displayed. These nodes are indicators of several parameter positions/values.
Envelope Mode Buttons Clicking on these buttons will switch to the selected envelope mode. The Envelope window display will adjust accordingly, with the selected envelope being topped in the superimposed display. The other envelope curves are shown as transparencies in the background. • The Volume Envelope is shown in red. • The Filter Envelope is shown in yellow (the Filter needs to be switched on). • The Density Envelope is shown in light blue (Synthesized IR only).
The A and D buttons alongside the Zoom to Fit parameter are for the Attack and Decay portions of the (currently selected) envelope. The A and D buttons are only available to the Volume and Filter Envelopes. Simply click on the appropriate button(s) to activate the desired viewing mode. The small Overview display indicates which portion of the IR file is currently visible in the Envelope window. Uncheck all buttons to return to the standard, non-zoomed view.
Adjusting this up or down can be done graphically in the Envelope window (the large node on the left), or by click-holding and dragging the mouse cursor up/down on the numerical entry. Attack Time This parameter alters the level of the Impulse Response attack phase over time. It is expressed in second values, with the maximum possible Attack Time mirroring the value set by the Length parameter (see “Length” on page 122).
Volume Decay Mode You have two options—Exp (Exponential) and Lin (Linear). The output of the envelope generator is shaped by an exponential function during the decay phase. If the decay phase is set as a straight line, the result is an exponential function that describes a natural decay. If set Lin, no shaping is performed. An Exp curve will result in a much more natural sounding reverb tail. Click on the desired button to activate the required mode.
The curve shape can also be changed by clicking and dragging on the envelope curve directly. Filter Envelope The entire Filter Envelope graphic, including nodes plus the Filter Envelope parameters below. The central large node indicates/controls the Attack endpoint (and Decay startpoint) and Break Level parameters simultaneously. The large node on the righthand edge controls the Decay endpoint and End Level parameters simultaneously.
Attack Time This parameter determines the time required to reach the Break Level (see below) value. You may use the Filter Attack Curve Form node parameters to alter the shape of the Filter Attack curve. The combined total of the Filter Attack and Decay Time parameters is equal to the total length of the (synthesized or sampled) Impulse Response (determined by the Length parameter, see “Length” on page 122), unless the Decay time is reduced.
End Level This parameter is used to set the filter end cutoff frequency. It is expressed as a percentage of the overall Filter envelope scale. If set to a value of 0, the filter stays “open” for the remainder of the reverb signal (provided that the filter is configured as “Low Pass” 6 dB or 12 dB). The Filter Attack and Decay Curve Form Nodes On the Attack curve displayed in the Envelope window, you will see two nodes in the attack portion of the overall Filter envelope curve.
Please note that the Density Envelope is only available when in the Synthesized IR mode. Init Level This parameter controls the density (the average number of reflections in a given period of time) of the diffuse reverb. Both the Initial and End Density (Init and End Level) can be controlled over time through use of the Density Envelope. Lowering the density levels will result in audible reflections patterns and discreet echoes.
This is handy when recreating rooms constructed of different materials. Use of this parameter, in conjunction with suitable settings for the Envelopes, Density, and Early Reflection will assist you in creating rooms of almost any shape and material. Click-hold on the triangle, and slide left/right to adjust. Filter Parameters The filter section of the Space Designer provides control over the timbre of the reverb.
Effect Parameters Low Shelving EQ As the name suggests, this EQ has a specific frequency that, once set, only allows frequencies that fall below it to be affected. The Gain control determines the amount of cut or boost of frequencies below the value set with the Freq parameter (expressed in Hertz). To adjust these values, click-hold on the applicable parameter field or knob and drag the mouse up and down. Stereo Spread This facility applies additional stereo information to a Synthesized IR.
Creating Impulse Responses This section discusses briefly the different methods for creating your own Impulse Response files for use with Space Designer. About Impulse Responses Impulse Responses are recordings (stored as AIFF, SDII or WAV files) made in acoustic spaces. To create an impulse response the sound of a starter pistol, digital spike or sine sweep is recorded inside the desired room together with the resulting reflections.
Using the Deconvolution Facility Now that you know about sine sweep responses you’d probably like to know how to use it to roll your own. It couldn’t be easier. Accessing the Deconvolution Facility The Deconvolution facility is necessary only in combination with a sweep response recording. To access the Deconvolution facility: 1 Bypass the plug-in by clicking on the Bypass button at the top of the of the Plug-in window. This conserves CPU resources.
So, Just What Is Reverberation? Put simply, reverberation is basically a delay of the source signal x number of times (by a very small time value), which is then fed back onto itself, simulating the way sound bounces around a room. Prior to the invention of digital systems, engineers used a variety of techniques to create reverb-type effects.
Typical reverb algorithms have parameters for: • room size (church, club, closet, bathroom, and so on), • brightness (hard walls, soft walls or curtains and so on) and • feedback or absorption coefficient (are there people, carpets, and so on in the room?—how quickly does the sound die?). • Upmarket reverbs may also contain several filter parameters.
The reason why we can actually perform the necessary calculations in real-time these days is due to a mathematical operation known as the FFT (Fast Fourier Transform). For filters (and we use the term filters here as this is what is effectively happening—the input sound is being filtered by the impulse response) with lots of nonzero values, it is easier to compute the convolution in the spectral domain.
11 Special 11 This chapter introduces Logic’s special plug-ins. This includes the Spectral Gate, Pitch Shifter II, SubBass, Denoiser, Exciter, and Stereo Spread plug-ins, amongst others. Spectral Gate The Spectral Gate separates the signals above and below the Threshold level into frequency ranges that can be independently modulated. It does this via a Fast Fourier Transformation (FFT) of the entire signal.
The actual frequency band can be modulated by three parameters: Speed determines the modulation frequency, CF Mod. (Center Frequency Modulation) defines the intensity of the center frequency modulation, and BW Mod. (Band Width Modulation) controls the bandwidth modulation. The Gain slider lets you adjust the level of the generated effects signal. We suggest you use a drumloop when you begin experimenting with this plug-in. Set Center Freq.
Note: When in doubt, Speech is a good place to start. A/B the options to compare them, and use the one that suits a given recording best. When auditioning and judging settings for quality, it’s a good idea to temporarily turn the Mix knob up to 100%. Keep in mind that Pitch Shifter II artefacts are a lot harder to hear when you mix a smaller percentage of a transposed audio to the overall signal.
Pitch, Formant, and Mix Mix defines the level ratio between the original (dry) and effect signals. Note: In order to get an idea of what the plug-in does, it may be helpful to listen to the original dry signal in the background. A Mix setting of about 75% should suffice. The Pitch parameter transposes the pitch of the signal (up to) two octaves upwards or downwards. Adjustments are made in semitone steps. Incoming pitches are indicated by a vertical line.
• At a setting of −100% (switch −1), all intervals are mirrored. The Pitch Base parameter is used to transpose the note that the Tracking parameter is following. As an example: the note which is spoken, if Tracking is set to 0%. Pitch Correction The Pitch Correction plug-in enables you to correct the pitch of audio tracks. Improper intonation is a common problem with vocal tracks, and this can be easily fixed with the Pitch Correction plug-in.
Range The Normal/Low parameter determines the pitch range that you wish to search for notes that need correction. Simply select Normal or Low for the track. Normal is the default range, and works for most audio material. Low should only be used for audio material that contains extremely low frequencies (below 100 Hz) which may cause the pitch dectection to work incorrectly. The parameter does not affect the sound. It is designed to optimize tracking in the target pitch range.
Bypass Use of the small bypass buttons (byp) above the green (black) and below the blue (white) keys excludes notes from correction. This is useful for blue notes. Blue notes are notes that slide between pitches, making the major and minor status of the keys difficult to identify. As you may know, one of the major differences between C minor and C major is the e flat and the b flat, instead of the e and the b. Blues singers glide between these notes, creating an uncertainty or tension between the scales.
Correction Amount The amount of pitch change is indicated in the horizontal bar display. If you keep a close eye on this display, you can use it for two important tasks: To better understand the inner workings of the algorithm works, and adjust the Response accordingly. You can also use the display when discussing (and optimizing) the vocal intonation with a singer during a recording session. Automation As with almost all Logic plug-ins, Pitch Correction can be fully automated.
The simplest application of the SubBass plug-in is as an octave divider, similar to “Octaver” effects pedals for electric bass guitar. A simple frequency division circuit in such pedals requires a monophonic input sound source, with a clearly defined pitch. This type of device is only capable of producing an output signal which is one or two octaves lower than the input signal. An octave is a frequency division by two. A ratio of four means two octaves, and a ratio of eight equals three octaves.
Denoiser The Denoiser eliminates or reduces almost any kind of noise floor. Denoiser Parameters Threshold The value of this parameter determines how high you think the noise floor of the material is. Tip: Find a passage where only noise can be heard in isolation, and set the Threshold value so that only signals of this volume will be filtered out. Reduce Reduce determines the level that the noise floor should be reduced to. A CD theoretically has a maximum signal to noise ratio of 96 dB.
If you use the Denoiser too aggressively, the algorithm will produce artefacts, such as “glass noise” which—in most cases—are less desirable than the existing noise. Therefore, there are three parameters for reducing this effect in all three dimensions of sound: • Time Smoothing This is the simplest form of smoothing. This parameter sets the time required by the Denoiser to reach (or release) maximum reduction.
The cutoff frequency parameter of the high pass filter is called Frequency. The graphic displays the frequency range that is used as the source signal for the process. Harmonics controls the level of the effect signal mixed to the original signal. If you disable Input, the original signal won’t be fed through. This should be your approach when using the Exciter plug-in on a Bus channel, being fed by Sends from several channel strips simultaneously, or should you wish to listen to the soloed effect signal.
Parameters of the Stereo Spread Order The Order knob determines the number of frequency bands that the signal is to be divided into. Upper Int. This parameter controls the intensity of the base extension of the upper frequency bands. Lower Int. This parameter controls the intensity of the base extension of the lower frequency bands.
12 Helper 12 This chapter introduces you to Logic’s Helper plug-ins. This includes Test Oscillator, Tuner, Gain, I/O, Direction Mixer, MultiMeter, Levelmeter, and Correlation Meter. Test Oscillator The Test Oscillator generates a static frequency or a sine sweep. The latter is a userdefined frequency spectrum tone sweep that can be used for the creation of Impulse Responses, for use with Logic’s Space Designer (see “Convolution Reverb: Space Designer”, on page 117).
The Sine Sweep section generates a user-defined frequency spectrum sine wave sweep. The Time field determines the duration of the sweep. The Start and End Freq(uency) parameters define the oscillator frequency at the beginning and end of the sine sweep. The Trigger button behavior can be switched via the edit field below: • single: pressing the x1 symbol triggers the sweep once. • continuous: pressing the Infinity symbol triggers the sweep indefinitely.
Adjust the pitch of your instrument—using the tuning nuts on your guitar, for example—until the center segment (at the very top of the ET1) is illuminated. This indicates that the incoming note/string pitch is perfectly tuned. Gain This plug-in allows a constant amplification or reduction, by a specific decibel amount, of an Audio Track or Bus Object. It is ideal for use in situations where you’re working with automated tracks during post-processing, and you want to quickly adjust master levels.
Phase Invert These buttons invert the phase of the left and right channels. This allows you to combat time alignment problems, particularly those caused by running multiple microphones at the same time. When you invert the phase of a signal, it sounds identical to the original. Only when the signal is heard in conjunction with other signals does phase inversion have an audible effect.
Direction Mixer The Direction Mixer plug-in offers the following features: • MS Decoder • The option of influencing the stereo base • Variable pan positioning of a stereo recording Parameters of the Direction Mixer Plug-in Input The LR and MS radio buttons determine whether the input signal is a standard left/right signal, or if you’re dealing with an MS encoded (middle and side) signal, for example when the two sides of an MS stereo mic setup were recorded directly.
If you chose to use the Direction Mixer simply to spread the stereo base, please keep in mind that as the Basis values increase beyond 1, monaural compatibility decreases accordingly. Once you process a stereo signal with an extreme setting of 2, when you play back the track in mono, the signal will be cancelled out completely—after all, L-R plus R-L doesn’t leave you with much.
Multimeter This plug-in consists of a collection of professional gauge and analysis tools, namely; • a 1/3 octave Analyzer • a Goniometer for judging the phase coherency in the stereo sound field • a Correlation Meter to spot mono phase compatibility • an integrated Level Meter The control panel to the left of the display allows you to switch between the Analyzer and Goniometer, and contains parameter controls for the Multimeter. The Stereo level and Phase Correlation meter is always displayed.
These features are useful when analyzing highly compressed material, as you can identify smaller level differences more easily by moving and/or reducing the display range. Mode There are three display response modes: RMS slow, fast, and Peak. The two RMS modes with slow and fast response settings show the effective signal average (“Root Mean Square”), and provide a good representative overview of the perceived volume levels. The Peak mode shows level peaks accurately.
Note: Auto Gain is a display parameter only, and increases display levels in order to enhance readability. The actual audio levels are not affected by this parameter. Decay The Decay parameter determines the time that it takes for the Goniometer trace to fade to black. Correlation Meter This is another phase measurement tool that gauges the phase relationship of a stereo signal. This meter is also available as a separate plug-in. The Correlation Meter’s scale ranges from −1 to +1.
Peak Hold This section controls the peak hold behavior of all metering tools. Hold The Hold button activates the Multimeter’s peak hold function. The duration of the Hold time is set in the parameter field alongside the Hold button. • In the Analyzer and Level Meter a small yellow segment above each 1/3 octave level bar, and stereo level bar, labels the most recent peak level. • In the Goniometer, all illuminated pixels of the display are held in place during peak hold.
13 Vocoder—Basics 13 If you are new to vocoders you should read this chapter. It provides you with basic knowledge about vocoders and their functionality. You will also find tips on using vocoders, and achieving good speech intelligibility. What Is a Vocoder? The word Vocoder is an abbreviation for VOice enCODER. As with many technologies in this otherwise beautiful world, it is a child of war.
How Does a Vocoder Work? The speech analyzer and synthesizer referred to above are actually two filter banks of bandpass filters. Bandpass filters allow a frequency band (a slice) in the overall frequency spectrum to pass through unchanged, and cut the frequencies which fall outside of the band’s range. In Logic’S EVOC 20 plug-ins, these filter banks are named the Analysis and Synthesis sections.
Analyzing Speech Signals The principles you’ve been introduced to thus far are insufficient for the transmission of speech signals. The reason is that human speech consists of a series of voiced sounds (tonal sounds) and unvoiced sounds (noisy sounds). The main distinction between voiced and unvoiced sounds is that voiced sounds are produced by an oscillation of the vocal cords, while unvoiced sounds are produced by blocking and restricting the air flow with lips, tongue, palate, throat, and larynx.
Due to the way human beings hear, the intelligibility of speech is highly dependent on the presence of high frequency content. To aid in keeping speech clear, it may be worthwhile using equalization to boost or cut particular frequencies in analysis signals before processing them with the vocoders. If the Side Chain (analysis) signal consists of vocals or speech, a simple shelving filter should be sufficient.
Gating Background Noises in the Side Chain If the Side Chain signal is compressed, as recommended, the level of breath, rumble, and background noises will rise. These background noises can cause the Vocoder bands to open, but this is normally not intended. In order to eliminate these noises, it’s therefore a good idea to employ a noise gate before compression and treble boosting. If the Side Chain signal is gated appropriately, you may find that you want to reduce the analysis Release value.
Treatment of the Side Chain signal (speech) with Compressor, Shelving Filter and Gate. The Silver Compressor, Silver Gate or another EQ are well-suited for these purposes. Achieving the Best Analysis and Synthesis Signals For good speech intelligibility, please keep these points in mind: • The spectra of the analysis and synthesis signals should overlap almost completely. Low male voices with synthesis signals in the treble range do not work well.
Note: A nice example is the rolled R of “We are the Robots”, by Kraftwerk, a classic vocoder track. This pronunciation was specifically made to cater to the demands of the vocoder. Feel free to do what you like when setting the Formant parameters. The intelligibility of speech is surprisingly little affected by shifting, stretching or compressing the formants. Even the number of frequency bands used has a minimal impact on the quality of intelligibility.
14 The EVOC 20 PS 14 The EVOC 20 PS combines a vocoder with a polyphonic synthesizer, and can be played in real-time. This chapter explains the use of the EVOC 20 TO and its parameters. The EVOC 20 PS is a sophisticated vocoder, equipped with a polyphonic synthesis engine, and capable of receiving MIDI note input. This allows the EVOC 20 PS to be played, resulting in classic vocoder choir sounds, for example.
6 Ensure that the corresponding Audio Instrument track is selected in the Arrange window. The EVOC 20 PS is now ready to accept incoming MIDI data, and has been assigned to see the output from the selected audio track via a Side Chain. 7 In the Track Mixer or Environment Audio layer (not the Arrange!), mute the audio track (the vocal track) serving as the Side Chain input. 8 Press the Play button on the Transport Bar, or use the Play key command (0 on the numeric keypad).
• When Poly is selected, the maximum number of voices is set via the numeric field alongside the Poly button. To change the value, click and hold with your mouse, and drag up or down to increase/decrease polyphony. Note: Increasing the number of voices also increases processor overhead. • When Mono or Legato is selected, the EVOC 20 PS is monophonic, and uses only one voice. • In Legato mode, Glide (see page 180) is only active on tied notes.
As you can see, there are some subtle differences between the two modes. We will look at the common parameters first, and will then look at the mode-specific options. Wave 1 Parameters The footages below the Wave 1 label in both modes harks back to the days of pipe organs. The longer the pipe, the deeper the tone. This also applies to Wave 1. Simply click on the 16, 8 or 4 foot value to select the range in which Wave (oscillator) 1 functions. Your selection will be illuminated.
Note: Turn Color full-right and Level a tiny bit up to achieve a more lively and fresh synthesis signal. Dual Mode Parameters The parameters specific to Dual mode are found in the Wave 2 section, and the Balance slider to the right. • The Semi parameter adjusts the tuning of the second oscillator (Wave 2) in semitone steps. Adjustment is made by using the mouse as a slider directly on the numerical field. Its range: ±24, or up/down two octaves. • The Detune parameter fine-tunes Wave 1 and Wave 2 in cents.
Analog Tuning The Analog tuning parameter simulates the instability of analog circuitry found in vintage vocoders. Analog alters the pitch of each note randomly. This behavior is much like that of polyphonic analog synthesizers. The Analog knob controls the intensity of this random detuning. Tuning The range of detuning is defined in the Tune window. Adjustments are made by using the mouse as a slider. The range is from 425 to 455 Hz.
Envelope The EVOC 20 PS features an Attack/Release envelope generator used for level control of the Oscillator section. • The Attack parameter determines the amount of time that it takes for the Oscillators of the Synthesis section to reach their maximum level. • The Release parameter determines the amount of time that it takes for the Oscillators of the Synthesis section to reach their minimum level.
Freeze The Freeze button holds the current Analysis sound spectrum infinitely. The frozen Analysis signal can capture a particular characteristic of the source signal which is then imposed as a complex sustained filter shape on the Synthesis section. Using a spoken word pattern as a source, for example, the Freeze parameter could capture the attack or tail phase of an individual word within the pattern—the vowel a, for example.
Formant Filter Parameters The Formant Filter Window The Formant Filter window is divided into two sections by a horizontal line. The upper half applies to the Analysis section and the lower half to the Synthesis section. Changes made to the High/Low frequency parameters, the Bands parameter or the Formant Stretch and Shift parameters will result in visual changes to the Formant Filter window.
Formant Stretch This parameter alters the width and distribution of all bands in the Synthesis filter bank, extending or narrowing the frequency range defined by the blue bar (Low/High Frequency parameters) for the Synthesis filter bank. With Formant Stretch set to 0, the width and distribution of the bands in the Synthesis filter bank is equal to the width of the bands in the Analysis filter bank. Low values narrow the width of each band, while high values widen them. The control range is from 0.
Resonance is responsible for the basic sonic character of the vocoder: low settings give it a soft character, high settings will lead to a more snarling, sharp character. Increasing the Resonance value emphasizes the middle frequency of each frequency band. The use of either, or both, of the Formant Stretch and Formant Shift parameters can result in the generation of unusual resonant frequencies—when high Resonance settings are used.
Rate Knobs These knobs determine the speed of modulation. Values to the left of the center positions are synchronized with the sequencer’s tempo and include bar values, triplet values and more. Values to the right of the center positions are non-synchronous, and are displayed in Hertz (cycles per second). Note: The ability to use synchronous bar values could be used to perform a formant shift every four bars on a cycled one bar percussion part.
When high settings are used, the increased sensitivity to unvoiced signals can lead to the U/V source—determined by the Mode parameter—being used on the majority of the input signal, including voiced signals. Sonically, this results in a sound that resembles a radio signal which is breaking up, and contains a lot of static or noise. Mode Mode selects the sound source(s) that can be used to replace the unvoiced content of the input signal. Possible settings are Off, Noise, Noise + Synth, or Blend.
This pull-down menu offers the choice of Voc(oder), Syn(thesis) and Ana(lysis). These options allow you to determine the signal that you wish to send to the EVOC 20 PS main outputs. To hear the vocoder effect, the Signal parameter should be set to Voc. The other two settings are useful for monitoring purposes. Ensemble The three Ensemble switches switch the ensemble effect(s) on or off. Ensemble I is a special chorus effect.
Glide Oscillator Filter Osc.
Block Diagram This block diagram illustrates the signal path in the EVOC 20 TO and EVOC 20 PS. L Side Track Chain stereo to mono R Analysis Source Frequency range between Highest/Lowest 1 2 3 4 5 Filter bank with 5 bands (example) Sensitivity A EF EF EF EF EF Freeze U/V Detection B TO: Pitch Analysis ANALYSIS Section SYNTHESIS Section TO: Max/ Quant.
15 Vocoder History 15 The vocoder is over 50 years old. This chapter discusses its history. You may be surprised you to learn that the Voder and Vocoder date back to 1939 and 1940, respectively. Homer Dudley, a research physicist at Bell Laboratories, New Jersey (USA) developed the Voice Operated reCOrDER as a research machine. It was originally designed to test compression schemes for the secure transmission of voice signals over copper phone lines.
The Voder was demonstrated at the 1939 World Fair, where it caused quite a stir: In World War II, the Vocoder (now called VOice enCODER) proved to be of crucial importance, scrambling the transoceanic conversations between Winston Churchill and Franklin Delanore Roosevelt. Werner Meyer-Eppler, the director of Phonetics at Bonn University, recognized the relevance of the machines to electronic music after Dudley visited the University in 1948.
1978 saw the beginning of mainstream vocoder use, riding on the back of popularity created through the music of Herbie Hancock, Kraftwerk, and a handful of other artists. Among the manufacturers who jumped into vocoder production at this time are: Synton/Bode, Electro-Harmonix, and Korg, with the VC-10. In 1979, Roland released the VP 330 ensemble/vocoder keyboard. The late 70’s and early 80’s were the heyday of the vocoder.
16 Synthesizer Basics 16 If you are new to synthesizers, you should read this chapter. It covers important facts about the synthesizer and explains the difference between analog, digital and virtual analog synthesizers. Important synthesizer terms such as cutoff, resonance, envelope, and waveform are also introduced. Analog and Subtractive An analog synthesizer signal is an electrical signal, measured in volts.
Undesirable analog synthesizer phenomena, such as the habit of going completely out of tune, are not simulated by virtual analog synthesizers. You can, however, set the voices of the ES1 to randomly detune, adding “life” to the synthesizer’s sound.
Cutoff and Resonance—illustrated with a sawtooth wave This picture shows an overview of a sawtooth wave (a = 220 Hz); The filter is open, with cutoff set to its maximum, and with no resonance applied. The screenshot shows the output signal of Logic’s ES1, routed to a monophonic Logic Output Object. The recording was performed with the Bounce function of this Audio Object, and is displayed in Logic’s Sample Editor at a high zoom setting.
Fourier Theorem and Harmonics “Every periodic wave can be seen as the sum of sine waves with certain wave lengths and amplitudes, the wave lengths of which have harmonic relations (ratios of small numbers)”. This is known as the Fourier theorem. Roughly translated into more musical terms, this means that any tone with a certain pitch can be regarded as a mix of sine partial tones. This is comprised of the basic fundamental tone and its harmonics (overtones).
Other Oscillator Waveforms Waveforms (waves) are named sawtooth, square, pulse, or triangular because of their shape when displayed as an oscillogram (as in Logic’s Sample Editor). This is the triangular wave: The triangular wave has few harmonics—which is evident by the fact that is shaped more like a sine than a sawtooth wave. This wave contains only odd harmonics—which means no octaves.
When you strike a key, the envelope travels from zero to it’s maximum level in the attack time, falls from this maximum level to the sustain level in the decay time, and maintains the sustain level as long as you hold the key. When the key is released, the envelope falls from its sustain level to zero over the release time. The brass or string-like envelope of the following sound—the envelope itself is not shown in this graphic—has longer attack and release times, and a higher sustain level.
17 EFM 1 17 The 16-voice polyphonic EFM 1 is a powerful synthesizer based on frequency modulation. It produces the typically rich bell and digital sounds that FM synthesis has become synonymous with. Concept and Function At the core of the EFM 1 engine, you’ll find a multi-wave Modulator oscillator and a sine wave Carrier oscillator. The Modulator oscillator modulates the frequency of the Carrier oscillator within the audio range, thus producing new harmonics. These harmonics are known as sidebands.
Global Parameters Transpose The base pitch is set with the Transpose parameter. You can transpose the EFM 1 by ±2 octaves. Tune Tune will fine-tune the EFM 1 ± 50 cents. A cent is 1/100th of a semitone. Randomize The Randomize facility generates new sounds with each mouse click. Click the Randomize button to create a new randomized sound, based on the Intensity value. Higher Intensity values—set in the numeric field by click-dragging up/down—will produce more random sounds.
Modulation Env(elope) To control the FM (Intensity) parameter dynamically, the EFM 1 provides a dedicated ADSR (FM) Modulation Envelope, consisting of four sliders: A (Attack time), D (Decay time), S (Sustain level) and R (Release time). The envelope is triggered every time a MIDI note is received. The Attack slider sets the time needed to reach the maximum envelope level. The Decay slider sets the time needed to reach the Sustain level (determined by the Sustain slider).
Modulator and Carrier Harmonic In FM synthesis, the basic overtone structure is determined by the tuning relationship of the Modulator and Carrier. This is often expressed as a tuning ratio. In the EFM 1, this ratio is achieved with the Modulator and Carrier Harmonic controls. Additional tuning control is provided by the Fine (Tune) parameters. You can tune the Modulator and Carrier to any of the first 32 harmonics.
Modulator Wave In classic FM synthesis, sine waves are use as Modulator and Carrier waveforms. To extend its sonic capabilities, the EFM 1 Modulator provides a number of additional digital waveforms. When turned completely counter clockwise the Modulator produces a sine wave. Turning the Wave parameter clockwise will step/fade through a series of complex digital waveforms. These digital waveforms add a new level of harmonic richness to the resulting FM sounds.
Main Level The Main Level control adjusts the overall output level of the EFM 1. Turning it clockwise makes the EFM 1 output louder. Turning it counter clock-wise will decrease the output level. Pitch Bend, Modulation Wheel, Aftertouch The EFM 1 responds to pitch bend, modulation wheel and aftertouch controller data. Pitch bend is hardwired to pitch. The modulation wheel introduces vibrato while aftertouch offers control over FM intensity.
18 ES M 18 This chapter introduces you to Logic’s ES M synthesizer. The monophonic ES M (ES Mono) is a good starting point if you’re looking for bass sounds that punch through your mix. Parameters of the ES M 8, 16, 32 The 8, 16, and 32 buttons set the ES M’s octave transposition. Glide The ES M permanently works in a fingered portamento mode, with notes played in a legato style resulting in a glide (portamento) from pitch to pitch. The speed of the glide is set with the Glide parameter.
Resonance This parameter sets the resonance of the dynamic lowpass filter. Increasing the Resonance value results in a rejection of bass (low frequency energy) when using low pass filters. The ES M compensates for this side-effect internally, resulting in a more bassy sound. Int The ES M features two very simple envelope generators with a single Decay parameter. Int enables modulation of the cutoff frequency by the filter envelope. Decay (Filter) This parameter sets the decay time of the filter envelope.
19 ES P 19 This chapter introduces you to Logic’s eight-voice polyphonic ES P (ES Poly) synthesizer. Functionally, (despite its velocity sensitivity) this flexible synthesizer is somewhat reminiscent of the affordable polyphonic synthesizers produced by the leading Japanese manufacturers in the 1980s: Its design is easy to understand, it is capable of producing lots of useful musical sounds, and you may be hard-pressed to make sounds with it that can’t be used in at least some musical style.
Vib/Wah The ES P features an LFO which can either modulate the frequency of the oscillators (resulting in a vibrato), or the cutoff frequency of the dynamic low pass filter (resulting in a wah wah effect). Turn the control to the left in order to set a vibrato, or to the right to cyclically modulate the filter. Speed Speed controls the rate of the oscillator frequency or cutoff frequency modulation. Frequency This parameter set the cutoff frequency of the resonance-capable dynamic low pass filter.
S The S slider determines the sustain level of the envelope generator. R The R slider determines the release time of the envelope generator. Chorus This parameter sets the intensity of the integrated chorus effect. Overdrive This parameter sets the overdrive/distortion level of the ES P output. Caution: The overdrive effect significantly increases the output level.
20 ES E 20 This chapter introduces Logic’s eight-voice polyphonic ES E synthesizer. The ES E (ES Ensemble) is designed for pad and ensemble sounds. It is great for adding atmospheric sounds to your music. Parameters of the ES E 4, 8, 16 The 4, 8, and 16 buttons determine the ES E’s octave transposition. Wave The left-most setting of the Wave parameter causes the oscillators to output sawtooth signals, which can be modulated in frequency by the integrated LFO.
Speed Speed controls the frequency of the pitch (sawtooth) or pulse width modulation. Cutoff This parameter sets the cutoff frequency of the resonance-capable dynamic lowpass filter. Resonance This parameter sets the resonance of the ES E’s dynamic lowpass filter. AR Int The ES E features one simple envelope generator per voice. It features an Attack and a Release parameter. AR Int, defines the amount of cutoff frequency modulation applied by the envelope generator.
21 ES1 21 This chapter introduces Logic’s virtual analog ES1 synthesizer. The ES1’s flexible tone generation system and interesting modulation options place an entire palette of analog sounds at your disposal: punchy basses, atmospheric pads, biting leads, and sharp percussion.
2', 4', 8', 16', 32' These footage values allow you to switch the pitch in octaves. 32 feet is the lowest, and 2 feet, the highest setting. The origin of the term feet to measure octaves, comes from the measurements of organ pipe lengths. Wave Wave allows you to select the waveform of the oscillator, which is responsible for the basic tone color. You can freely set any pulse width in-between the square wave and pulse wave symbols.
Cutoff and Resonance The Cutoff parameter controls the cutoff frequency of the ES1’s lowpass filter. Resonance emphasizes the portions of the signal which surround the frequency defined by the Cutoff parameter. This emphasis can be set so intensively, that the filter begins to oscillate by itself. When driven to self-oscillation, the filter outputs a sine oscillation (a sine wave). If key is set to 1, you can play the filter chromatically from a MIDI keyboard.
Level Via Vel The upper arrow works like a main volume control for the synthesizer. The greater the distance from the lower arrow (indicated by the blue bars), the more the volume is affected by incoming velocity messages. The lower arrow indicates the level when you play pianissimo (velocity =1). You can adjust the modulation range and intensity simultaneously by grabbing the bar and moving both arrows at once. Note that as you do so, they retain their relative distance from one another.
Rate This defines the speed (frequency) of modulation. If you set values to the left of zero, the LFO phase is locked to the tempo of the song—with phase lengths adjustable between 1/96 bar and 32 bars. If you select values to the right of zero, it will run freely.
Int Via Vel The upper arrow controls the upper modulation intensity setting for the modulation envelope, if you strike a key with the hardest fortissimo (velocity = 127). The lower arrow controls the lower modulation intensity setting for the modulation envelope, if you strike a key with the softest pianissimo (Velocity = 1). The green bar between the arrows displays the impact of velocity sensitivity on the (intensity of the) modulation envelope.
Voices The number displayed is the maximum number of notes which can be played simultaneously. Each ES1 instance offers a maximum of 16 voice polyphony. Fewer played voices require less CPU power. If you set Voices to legato, the ES1 will behave like a monophonic synthesizer with single trigger and fingered portamento engaged.
22 ES2 22 The virtual-analog ES2 synthesizer offers an exciting and extensive array of features and functions. This chapter covers all details of the ES2’s powerful tone generation system. A brief summary is followed by an in-depth description of its parameters. At the chapter’s end you will find tutorials, where well-known sound designers explain how to program ES2 sounds. Concept and Function The ES2 is one of the most versatile virtual-analog synthesizers ever designed.
The ES2’s three Oscillator concept is reminiscent of the Minimoog and EMS VCS 3. The oscillators can be synchronized and ring-modulated. Pulsewidth modulation is also possible. Oscillator 1 can be modulated in frequency by Oscillator 2, and is capable of producing FM synth sounds. Further to the classic, standard waveforms, the ES2’s Oscillators also feature 155 singlecycle waveforms, known as DigiWaves. Each has a totally different sonic color.
The ES2 Parameters If given just a few words to explain the principles behind a subtractive synthesizer, it would go something like this: The Oscillator generates the oscillation (or waveform), the Filter takes away the unwanted overtones (of the waveform), and the Dynamic Stage sets the volume of the permanent oscillation (the filtered waveform) to zero as long as no key is pressed.
Oscillators Tune Tune sets the pitch of the ES2 in cents. 100 cents equals a semitone step. The range is ±50 cents. At a value of 0 c (zero cents), a' is tuned to 440 Hz or concert pitch. Analog Analog alters the pitch of each note, plus the cutoff frequency in a random fashion. Much like polyphonic analog synthesizers, all three oscillators used by each synthesizer voice maintain their specific deviation, but are shifted by the same amount randomly.
The CBD (Constant Beat Detuning) parameter matches this natural effect by detuning the lower frequencies in a ratio proportionate to the upper frequencies. Besides disabling CBD altogether, four values are at your disposal: 25%, 50%, 75%, 100%. If you choose 100%, the phasing beats are (almost) constant across the entire keyboard range. This value, however, may be too high, as the lower notes might be overlydetuned at the point where the phasing of the higher notes feels right.
You can switch between monophonic and polyphonic modes by clicking on the Poly and Mono buttons. Legato is also monophonic, but with one difference: The envelope generators are only retriggered if you play staccato (release each key before playing a new key). If you play legato (press a new key while holding the old one), the envelope generators are only triggered with the first note you play legato, and then continue their curve until you release the last legato played key.
Osc Start The oscillators can run freely, or they can begin at the same phase position of their waveform cycle each time you hit a key (every time the ES2 receives a note on message). When Osc Start (Oscillator Start) is set to free, the initial oscillator phase startpoint is random, with each note played. This gives the sound more life and a less static feel— just like an analog hardware synthesizer.
Frequency Switch Switches the pitch in semitone steps over a range of ±3 octaves. As an octave consists of 12 semitones, the ±12, 24, and 36 settings represent octaves. You can click on these options to quickly set the corresponding octave. The value display works as follows: the left numbers show the semitone (s) setting, the right numbers show the cent (c, 1 cent = 1/100th semitone) setting. You can adjust these two values independently.
Oscillator 1 Waveforms Oscillator 1 outputs standard waveforms—pulse, rectangular, sawtooth, and triangular waves—or, alternately, any of the 155 available DigiWaves. It can also output a pure sine wave. The sine wave can be modulated in frequency by Oscillator 2 in the audio frequency range. This kind of linear frequency modulation is the basis on which FM synthesis works.
During a Fourier transformation, complex oscillations can be divided into their basic sine components. In additive synthesis, complex oscillation forms can be resynthesized. The most simple additive synthesizer is the drawbar organ (the Hammond organ, for example). With such an organ, you can mix nine sine choirs with drawbars. Try selecting sine waves for all three oscillators and the following semitone settings: −12 (16'), 0 (8') and +7 (5 1/3'), and set all oscillators to the same level.
Classic synthesizer literature indicates the use of the sawtooth wave to create a sound similar to that of a violin. The rich and full sound of the sawtooth wave is the most popular synthesizer waveform, and serves as a basis for synthetic string and brass sounds. It is also handy for synthesized bass sounds. Screenshot of the ES2’s rectangular wave. The 50% rectangular wave contains all odd harmonics, the amplitudes of which decrease proportionately with their number.
Linear Frequency Modulation The principle of linear frequency modulation (FM) synthesis was developed in the late sixties and early seventies by John Chowning. It’s such a flexible and powerful method of tone generation that synthesizers were developed, based solely on the idea of FM synthesis. The most popular FM synthesizer ever built is Yamaha’s DX7. FM synthesis is also found in other models of the Yamaha DX range and some Yamaha E-Pianos.
Oscillator 1’s frequency modulated sine wave, modulated by Oscillator 2 set to sine wave. Oscillator 2 was set to three times the frequency of Oscillator 1 (+19 semitones). The modulation intensity is low (Wave control at about 12 o’clock). As the wavelength (the duration period) of the modulating Oscillator is a third of that of the modulated Oscillator, the sine is accelerated and slowed down three times within a phase.
Waveforms of Oscillators 2 and 3 Basically, Oscillators 2 and 3 supply the same selection of analog waveforms as Oscillator 1: sine, triangular, sawtooth, and rectangular waves. The pulse width can be scaled steplessly between 50% and the thinnest of pulses, and can be modulated in a number of ways (see “Pulse Width Modulation” section, on page 236).
Screenshot of an ES2 sample, created with the Bounce function of Logic. The pulse width is modulated. It is easy to see how the width of the pulses varies between a rectangular shape and very thin pulses. An LFO is selected as the modulation source, and its waveform is a sine wave. You can see about half a phase of the sine wave. If a rectangular wave had been selected for the LFO, you would see the pulse width periodically changing between the two fixed extreme values.
Synchronized sawtooth wave as output by Oscillator 2. Oscillator 1 is set to 0, Oscillator 2 to +22 semitones. The dots in the graphic indicate the phases of Oscillator 1. Synchronized rectangular wave as output by Oscillator 2. Oscillator 1 is set to 0, Oscillator 2 to +22 semitones. The dots in the graphic indicate the phases of Oscillator 1. Output signal of Oscillator 2 (sawtooth), synchronized to Oscillator 1.
A ring modulator has two inputs. At it’s output you will find the sum and difference frequencies of the input signals. The graphic shows the output signal of the ring modulator, appearing as the output signal of Oscillator 2. The amplitude (or the elongation, to be more exact) of the output of Oscillator 2 changes with the phase of Oscillator 1. Oscillator 2 is set to a higher frequency than Oscillator 1. As the frequency ratio is odd (irrational), the resulting waveform always changes over time.
White and Colored Noise (Oscillator 3 only) Unlike Oscillator 2, Oscillator 3 is not capable of producing ring modulated signals nor sine waves. Its sonic palette however, is bolstered by the inclusion of a noise generator. By default, Oscillator 3’s noise generator generates white noise. White noise is defined as a signal that consists of all frequencies (an infinite number) sounding simultaneously, at the same intensity, in a given frequency band. The width of the frequency band is measured in Hertz.
The position of the cursor can be controlled via the vector envelope, just like the cursor position in the Track Pad (the Square), which we’ll look at in “The Square” section, on page 272. Note that the vector envelope features a loop function. This addition extends its usefulness, allowing you to view it as a luxurious pseudo-LFO with a programmable waveform. It can be used for altering the positioning of the Triangle and Square cursors.
Filters The ES2 features two dynamic filters which are equivalent to the Voltage Controlled Filters (VCF) found in the world of analog synthesizers. The two filters are not identical. Filter 1 features several modes: lowpass, highpass, bandpass, band rejection, peak. Filter 2 always functions as a lowpass filter. Unlike Filter 1, however, Filter 2 offers variable slopes (measured in dB/octave). The Filter button bypasses (switches off ) the entire filter section of the ES2.
In the graphic below, the filters are cabled in parallel. If Filter Blend is set to 0, you’ll hear a 50/50 mix of the source signal routed via Filter 1 and Filter 2, which is fed into the mono input of the dynamic stage. There it can be panned in the stereo spectrum, and then fed into the effects processor. Filter Blend: Cross-Fading the Filters You can cross-fade the two filters.
In conjunction with the overdrive/distortion circuit (Drive) and a series wiring configuration, the ES2’s signal flow is anything but commonplace. The graphics illustrate the signal flow between the Oscillator Mix stage (the Triangle) and the dynamic stage, which is always controlled by ENV 3. The signal flow through the filters, the overdrives and the bypassing sidechains is dependent on the Filter Blend setting. Filter Blend in parallel filter mode. Filter Blend in Serial filter mode.
Filter Blend and Parallel Filter Configuration Tip The overdrive/distortion circuit is always wired after the Oscillator Mix stage and before the filters. The filters receive a mono input signal from the overdrive circuit’s output. The outputs of both filters are mixed to mono via Filter Blend. Note: If Drive is set to 0, no distortion occurs. Drive The filters are equipped with separate overdrive modules. Overdrive intensity is defined by the Drive parameter.
To check out how the overdrive circuit between the filters works, program a sound as follows: • Simple static waveform (a sawtooth) • Filter set to Serial mode • Filter Blend set to 0 (center position) • Set Filter 1 to peak Filter Mode • Set a high Resonance value for Filter 1 • Modulate Cutoff Frequency 1 manually or in the Router • Set Drive to your taste • Filter away (cut) the high frequencies with Filter 2 to taste The sonic result resembles the effect of synchronized oscillators.
Note: The dynamic lowpass filter is the most essential module in any subtractive synthesizer. This is why Filter 2 always operates in lowpass mode. Note: As opposed to the filter and EQ effect plug-ins in Logic, the ES2’s filters are dynamic, which means that the Cutoff Frequency parameter can be modulated extremely quickly and severely in real-time—even on modulation signals in the audio frequency range.
Filter Mode (Lo, Hi, Peak, BR, BP) Filter 1 can operate in several modes, allowing the filtering (cutting away) and/or emphasis of specific frequency bands. • A lowpass filter allows frequencies that fall below the cutoff frequency to pass. When • • • • set to Lo, the filter operates as a lowpass filter. The slope of Filter 1 is 12 dB/octave in Lo mode. A highpass filter allows frequencies above the cutoff frequency to pass. When set to Hi, the filter operates as a highpass filter.
Lowpass Filter Lowpass-filtered sawtooth wave, Cutoff Frequency one octave above the frequency of the sawtooth. As the slope isn’t infinite, harmonics are still visible. Note that the waveform is rounded. Highpass Filter Highpass-filtered sawtooth wave, Cutoff Frequency one octave above the frequency of the sawtooth. The basic harmonic is rejected by about 12 dB, as the slope is 12 dB/ octave.
Band Rejection In this image, the second partial is rejected. The basic oscillation and all other harmonics are present. Sawtooth is still the input waveform. Bandpass Filter The bandpass filter distorts the picture of the sawtooth wave. Filter 2 FM The cutoff frequency of Filter 2 can be modulated by the sine wave of Oscillator 1, which means that it can be modulated in the audio frequency range.
Note: Filter 2 can be driven to self-oscillation. If you set a very high value for Resonance, it will produce a sine wave. This self-oscillating sine wave will distort at the maximum Resonance value. If you mute all oscillators, you’ll only hear this sine oscillation. By modulating the Cutoff Frequency, you can produce effects similar to those produced by modulating the frequency of Oscillator 1 with Oscillator 2.
Dynamic Stage (Amplifier) The dynamic stage defines the level—which means the perceived volume—of the played note. The change in level over time is controlled by an envelope generator. ENV3 and the Dynamic Stage ENV3 is hard-wired to the dynamic stage—envelope generator 3 is always used for control over the level of the sound. For detailed explanations of the envelope parameters, see “The Envelopes (ENV 1—ENV 3)” section, on page 268.
The Router The ES2 features a modulation matrix, called the Router. Any modulation Source can be connected to any modulation Target—much like an old-fashioned telephone exchange or a studio patchbay. The modulation intensity—how strongly the Target is influenced by the Source—is set with the associated vertical slider. Note: To set the modulation intensity to zero, just click on the little zero symbol (the small circle) right beside via.
In the example below, the lower half of the slider knob defines the vibrato intensity when the modulation wheel is turned down. The upper half defines the vibrato intensity that takes place when the modulation wheel is set to its maximum value. Note: To invert the effect of the via modulation source, simply activate the Via invert (inv) parameter in the Router.
Pitch 1 This target allows modulation of the frequency (pitch) of Oscillator 1. Slight envelope modulations can make the amount of detuning change over time, when Oscillator 1 is sounding in unison with another (unmodulated) Oscillator. This is useful for synthesizer brass sounds. Pitch 2 This target allows modulation of the frequency (pitch) of Oscillator 2. Pitch 3 This target allows modulation of the frequency (pitch) of Oscillator 3.
For further information on the effects of these modulations, please read the “Pulse Width Modulation” section, on page 236. Also take a look at the “Linear Frequency Modulation” section, on page 234, “White and Colored Noise (Oscillator 3 only)”, on page 240, and the “Digiwaves” section, on page 233.
Osc1WaveB If wavetable modulation is active for a DigiWave using Osc1Wav, you can use this target to modulate the shape of the transition. Osc1 FM mode: When compared with the hardwired Osc1-FM and the Osc1Wave modulation target, the Osc1WaveB modulation target offers much higher FM intensities. Osc2WaveB If wavetable modulation is active for a Digiwave using Osc2Wav, you can use this target to modulate the shape of the transition.
Resonance 1 (Reso 1) This target allows modulation of the Resonance of Filter 1. See the “Cutoff and Resonance” section, on page 246. Cutoff 2 This target allows modulation of the Cutoff Frequency of Filter 2. Resonance 2 (Reso2) This target allows modulation of the Resonance of Filter 2. LPF FM A sine signal, at the same frequency as Oscillator 1, can modulate the Cutoff frequency of Filter 2 (which always works as a low pass filter).
Amp This target modulates the dynamic stage, or level (the voice’s volume). If you select Amp as the target and modulate it with an LFO as the Source, the level will change periodically, and you will hear a tremolo. Pan This target modulates the panorama position of the sound in the stereo spectrum. Modulating Pan with an LFO will result in a stereo tremolo (auto panning). In Unison Mode, the panorama positions of all voices are spread across the entire stereo spectrum.
Env2Dec Env2Dec (Envelope 2 Decay) modulates the Decay time of the second envelope generator. In cases where you’ve selected ENV2 Dec as the target and Velocity as the source, the duration of the decaying note is dependent on how hard you strike the key. Selecting Keyboard as the source will result in higher notes decaying more quickly (or slowly). Env2Rel Env2Rel (Envelope 2 Release) modulates the Release time of the second envelope generator.
ENV1 … Envelope Generator 1 is described in “The Envelopes (ENV 1—ENV 3)” section, on page 268. ENV2 … Envelope Generator 2 is described in “The Envelopes (ENV 1—ENV 3)” section, on page 268. ENV3 … Envelope Generator 3 is described in “The Envelopes (ENV 1—ENV 3)” section, on page 268. Note: Envelope Generator 3 always controls the level of the overall sound. Pad-X, Pad-Y These modulation sources allow you to define the axes of the Square, for use with the selected modulation target.
Note: For most standard applications, you’ll probably use the wheel as the via controller. Traditionally, it can be (and is) used for control over the intensity of periodic LFO modulations. Used here, it can be employed for direct, static modulations, such as controlling the Cutoff frequencies of the filters (Target = Cut 1+2). Note: The Least Significant Bit (LSB) controller for the modulation wheel is recognized correctly, as well. Touch Aftertouch serves as modulation source.
Note: In the MIDI specification, for all controllers from 0 to 31, there also is a LSBController defined (32 to 63). This “Least Significant Bit”-controller allows for a resolution of 14 bit instead of 7 bit. The ES2 recognizes these control change messages correctly; the breath or expression controllers, for example. RndN01 RndNO1 (Note On Random1) outputs a random modulation value between −1.0 and 1.0 (same range as an LFO), that changes when a note is triggered or re-triggered.
Pad-X, Pad-Y Both axes of the Square (the Vector Envelope) are available as via sources as well, enabling you to control modulation intensities with them. Kybd Kybd (Keyboard) outputs the keyboard position (the MIDI note number). The center point is C3 (an output value of 0). Five octaves below and above, an output value of −1 or +1, respectively is sent.
This facility is especially helpful if you’ve always wanted to use Controller #4 (foot pedal), for example, as a modulation source. This feature allows you to assign your favorite MIDI real-time controllers as Ctrl A, Ctrl B, and so on. All parameters that allow you to select a MIDI controller feature a Learn option. If this is selected, the parameter will automatically be assigned by the first appropriate incoming MIDI data message.
• LFO 1 is polyphonic, which means that if used for any modulation of multiple voices, they will not be phase-locked. Furthermore, LFO 1 is key-synced: Each time you hit a key, the LFO 1 of this voice is started from zero. To explain, when used on polyphonic input (a chord played on the keyboard) the modulation is independent for each voice (note). Where the pitch of one voice may rise, the pitch of another voice might fall and the pitch of a third voice may reach its minimum value.
Rate This parameter defines the frequency or speed of the modulation. The value is displayed in Hertz (Hz) beneath the slider. Wave This is where you select the desired LFO waveform. Check out the waveforms while a modulation of Pitch123 is engaged and running. You should find the symbols quite selfevident. Triangular Wave The triangular wave is well suited for vibrato effects. Sawtooth Wave and Inverted Sawtooth The sawtooth is well suited for helicopter and space gun sounds.
Rate (LFO 2) The LFO2 Rate (frequency) control allows for the free (in the upper half of the slider range) or song-tempo synchronized (in the lower half of the slider range) running of LFO 2. The rate is displayed in Hertz or rhythmic values, dependent on whether the song tempo synchronization is engaged or not. Rates range from speeds of 1/64-notes through to a periodic duration of 32 bars. Triolic and punctuated values are also possible.
Note: Both ENV 2 and ENV 3 are velocity sensitive, making it unnecessary to set via to Velo in the Router channel: You can leave via switched off. The Parameters of ENV 1 At first glance, ENV 1 appears to be rather poorly equipped. Its few parameters, however, are useful for a vast range of synthesizer functions. Trigger Modes: Poly, Mono, Retrig In Poly mode, the envelope generator behaves as you would expect on any polyphonic synthesizer: Every voice has its own envelope.
Decay/Release ENV 1 can be set to act as an envelope generator with an Attack time and Decay time parameter or with an Attack time and Release time parameter. Switching between both modes is achieved by clicking on the D or the R above the right ENV 1 slider. • In its Attack/Decay mode, the level will fall to zero after the attack phase has completed, no matter whether you sustain the note or not. It will decay at the same speed even if you release the key.
Decay Time The Decay time parameter defines the length of time it takes for the level of a sustained note to fall to the Sustain level, after the attack phase is over. If Sustain level is set to its maximum, the Decay parameter has no effect. When the Sustain level is set to its minimum value, Decay defines the duration or fade-out time of the note. The Decay parameter appears as a modulation target in the Router for ENV 2 and ENV 3 individually (ENV2Dec, ENV3Dec).
The Square The Square has two axes: The X and Y axes have positive and negative value ranges. They are bipolar, in other words. By touching and moving the cursor with the mouse, the values of both axes are continuously transmitted. As you can modulate one freely selectable parameter with the X value, and another freely selectable parameter with the Y value, you can use the mouse like a Joystick.
Vector Int—The Modulation Intensity The maximum intensity, sensitivity, and polarity of the modulation is set with the Vector X Int and Vector Y Int parameters. The Vector Envelope The Triangle and Square are the most special and unusual elements of the ES2’s graphical user interface. Whilst the Triangle controls the mix of the three oscillators, the X and Y axes of the Square can modulate any (modulation) target.
To define a point as the Sustain Point, click on the turquoise strip above the desired point. The selected point will be indicated by an S between the point and its number, on the turquoise strip. Loop Point Any point can be declared Loop Point. Given that the note is sustained long enough, the envelope can be repeated in a loop. The looped area is the time span between Sustain Point and Loop Point. In between, you can define several points which describe the movements of the Square and Triangle cursors.
Default Setting of the Vector Envelope The default setting of the Vector Envelope consists of three points. Point 1 is the startpoint, point 2 is defined as the Sustain Point, and point three is the end point, by default. The impact of the Vector Envelope on the Oscillator Mix or on the Square is switched off by default. This allows the ES2 to behave as a synthesizer without a Vector Envelope generator. This traditional starting point is more convenient when creating patches from scratch.
Solo Point This function basically switches off the entire Vector Envelope generator. If Solo Point is set to on, no dynamic modulations are applied by the Vector Envelope. In this scenario, the currently visible cursor positions of the Triangle and Square are permanently in effect. These cursor positions match the currently selected Vector Envelope point. If you select another Vector Envelope point (by clicking on it), you will engage its Triangle and Square cursor positions immediately.
Curve The Curve parameter sets the shape of the transition from point to point. You can choose between nine convex and nine concave shapes. There are also the two extreme forms; hold+step and step+hold, which allow stepped modulation. Where step+hold jumps at the beginning of the transition time, hold+step jumps at the end. Note: You can use hold+step to create stepped vector grooves with up to 15 steps.
• Backward When Loop is set to Backward, the Vector Envelope runs to the Sustain Point and begins to repeat the section between the Sustain Point and Loop Point periodically, always in a backward direction. • Alternate When Loop is set to Alternate, the Vector Envelope runs to the Sustain Point and returns to the Loop Point and back to the Sustain Point periodically, alternating in both a backward and forward direction.
• If Loop Rate is set to Sync or Free, the Loop Smoothing Time will be displayed as a percentage of the loop cycle duration. • If Loop Rate is set to as set, the Loop Smoothing Time will be displayed in milliseconds (ms). Loop Count The loop cycle of the Vector Envelope doesn’t need to be performed infinitely—you can have it cycle just a few times. Following the defined number of repetitions, the Vector Envelope will run from the sustain point onwards, just as with Loop Mode off.
In cases where a loop which was synchronized to the song tempo had been engaged (Loop Rate = sync), pressing Fix Timing will also switch the Loop Rate to as set, thus preserving the absolute rate. Effect Processor The ES2 is equipped with an integrated effect processor. Any changes to this processor’s effects settings are saved as an integral part of each sound program. The entire output of the dynamic stage is processed in true stereo.
Note: The phaser is based on a mix of a delayed and an original signal. The delayed element is derived from an all-pass filter, which applies a frequency-dependent delay to the signal. This is expressed as a phase angle. The effect is based on a comb filter, which is basically an array of inharmonic notches (rather than resonances, as with the flanger), which also wander through the frequency spectrum.
You can restrict the random sound variation to the parameter groups listed below: All All ES2 parameters, with the exception of the parameters mentioned above, will be altered. All except Router and Pitch All ES2 parameters, with the exception of all Router parameters and the basic pitch (semitone settings of the oscillators), will be altered. The oscillator fine tuning will be varied. This will result in more musically useful sounds.
Vector Env Mix Pad The Oscillator mix levels (Triangle cursor positions) of the Vector Envelope points are altered. The rhythm and tempo of the modulation (the time parameters of the points) will not be altered. Vector Env XY Pad The Square cursor positions of the Vector Envelope points are altered. The XY routing won’t be altered. The rhythm and tempo of the modulation (the time parameters of the points) will not be altered.
• Set Filter Blend to its left-most position, which will allow you to listen to Filter 1 in isolation. In many circumstances, you’ll probably prefer to use Filter 2, but Filter 1 has its advantages. In addition to the lowpass filter with 12 dB/octaves slope (Lo), Filter 2 also offers: a highpass, peak, bandpass (BP), and band rejection mode (BR). Filter 1’s lowpass sounds softer in comparison to Filter 2. It is best-suited to sounds where the filter effect is/should be less audible (Strings, FM-Sounds).
• Increase Drive or Distortion. • Set Env 2 to be velocity sensitive. This allows for velocity sensitive filter modulations. • Insert a stereo delay effect in the audio instrument channel strip of the ES2. In order to delay several Audio Instruments, you might prefer to insert the effect into a Bus, which is accessed via each channel’s Send. Logic incorporates reverb and delay effects which are essential for many synthesizer sounds.
Tutorial Setting: FM Start Topic: FM Intensity and Frequency These are the first steps to take when learning about linear Frequency Modulation (FM) synthesis. You’ll first hear an un-modulated sine sound, generated by Oscillator 1. Oscillator 2 is switched on and set to produce a sine oscillation as well, but its level is set to 0: Just push the cursor in the triangle in the uppermost corner. In the ES2, Oscillator 1 is always the carrier, and Oscillator 2, the modulator.
• Following such drastic augmentations to the modulation range, the sound will have become uneven. In the lower and middle ranges, it sounds nice, but in the treble range the FM intensity appears to be too severe. You can compensate for this effect by setting the target Osc 1 Wave to be modulated by the keyboard position (kybd) in modulation channels 5 and 6. This results in a keyboard scaling of the FM intensity.
Tutorial Setting: FM Megafat Topic: Distorted FM in monophonic Unison This sound is hard core, and is well-suited for distorted basses and guitar-like sounds. In its treble range, this sound gets rather “rude”. This cannot be compensated for by scaling, but not every sound has to be “nice” over the entire keyboard range! • Check out extreme detunings by adjusting the Analog parameter. • Check out the Flanger with this sound.
Tutorial Settings: PWM Start, PWM Slow, PWM Fast, and PWM Scaled Topic: Slow and Fast Pulse Width Modulations with Oscillator 2 Pulse Width Modulation (PWM) is one of the most essential features of any sophisticated analog synthesizer. Use this setting to manually control the pulse width of a rectangular wave, set via the Wave control Note: Avoid Drive and Distortion with PWM sounds.
Tutorial Setting: Ringmod Start Topic: Ring Modulation A ring modulator takes its two input signals and outputs the sum and difference frequencies of them. In the ES2, Oscillator 2 outputs a ring modulator, which is fed with a square wave of Oscillator 2 and the wave of Oscillator 1, when Ring is set as Oscillator 2’s waveform. Odd intervals (frequency ratios) between the oscillators, in particular, result in bell-like spectra, much like those heard in the “RingMod Start” setting.
Note: Pulse width modulation is also available via the synchronized square wave of Oscillators 2 and 3. A modulation of the wave parameters of these two Oscillators results in a PWM if the synced square is selected. Tutorial Setting: Vector Start and Vector Envelope Topic: First Steps in Vector Synthesis In this tutorial section you’ll find some useful hints for the programming of vector envelopes. In the “Vector Start” setting, the mix of the Oscillators is controlled by the vector envelope.
• Switch on Solo Point, in order to more easily listen to the settings for the single points. • Click Point 1. You will only hear Oscillator 1’s sawtooth. • Move the cursor in the Square to the hard left, which results in a low Cutoff Frequency • • • • • for Oscillator 2. Click Point 2. You will only hear Oscillator 2’s rectangular wave. Move the cursor in the Square all the way down, which results in the right-most Panorama position. Click Point 3. You will only hear Oscillator 3’s triangular wave.
• Switch off the Vector Envelope by setting Solo Point to on. This allows you to audition the individual points in isolation. • Take the opportunity to alter the cursor positions in the Square according to your taste. As in the example above, the X/Y axes of the Square control the Cutoff Frequency of Filter 2, and the Panorama position. Adjustments to these make the sound more vivid. • Activate the Vector Envelope by setting Solo Point to off.
Tutorial Settings: Vector Perc Synth and Vector Punch Bass Topic: Percussive Synthesizers and Basses with Two Filter Decay Phases (Vector Envelope) As with the “Vector Kick”, this setting uses the Vector Envelope to control the Filter Cutoff Frequency (with two independently adjustable decay phases). This would not be possible with a conventional ADSR envelope generator.
Let’s have a look at its architecture: Osc 1 and 3 provide the basic wave combination within the DigiWave field. Changing the DigiWaves of both (in combination) delivers a huge number of basic variations— some also work pretty well for electric piano-type keyboard sounds. Osc 2 adds harmonics with its synced waveform, so you should only vary its pitch or sync waveform. We kept the “noisiness” we were after in mind.
Velocity is set up to be very responsive because many synthesizer players don’t touch keys in the manner of a piano player’s weighted-action-punch. As such, we ask that you play this patch softly, or you may find that the “slap” tends to sweep a little. Alternately, you can adjust the sensitivity of the filter modulation’s velocity value to match your personal touch.
Please feel free to find your own values for customization of this setup. While doing so, keep in mind the fact that there are two modulation “couples”, which should only be changed symmetrically (Mod. 2 and 3 work as a pair of “twins”, and also Mod. 6 and 7). So, if you change Pitch 2’s maximum to a lower minus value, remember to set Pitch 3’s maximum value to the same positive amount (same goes for modulation pair 6 and 7).
Oscillator 3 generates a DigiWave, which we considered “brassy” enough within the overall wave mix. As an alternative to the DigiWave, we could have used another modulated pulse wave to support the ensemble, or another sawtooth wave to achieve more “fatness” when detuning it with Oscillator 1’s sawtooth wave. What we were after, however, was to have a little bit of growl, achieved through a short wavetable “push”, as described for the Stratocaster patch, on page 294.
Try a careful, very slow movement of the modwheel, and you’ll hear drastic changes within the wave configuration. Each incremental position of the wheel offers a different complete digital pad sound. No frantic movements, please, otherwise this will sound like an AM-radio. Another potential modification procedure is hidden in the modulation intensity of Oscillator 1, 2, and 3 wave’s parameter.
If you move the wheel, you can scroll through the spectrum of harmonics that we’ve programmed for real-time changes. Any modification here starts with the pitch of Oscillator 2 itself, which we’ve set to three semitones below the overall pitch. Feel free to start with a different pitch for Oscillator 2; it won’t effect the patch’s tuning. The next modification may be modulation 7’s intensity (or the interval).
23 Ultrabeat 23 Ultrabeat is a rhythm synthesizer with integrated step sequencer. Ultrabeat’s synthesis engine is optimized for creating electronic and acoustic drum and percussion sounds. The sonic diversity possible with Ultrabeat is due to its various synthesis engines. In addition to a new type of Phase Oscillator, sample playback, FM, and physical modeling are also put to use. Special attention has been paid to achieving the greatest possible range of dynamics for its sounds.
The distribution of drum voices across the MIDI keyboard is simple and easily explained: the first (starting from the bottom) 24 MIDI keys are each assigned a single drum voice. The 25th drum voice is an exception, and can be played chromatically over three octaves. You can compare Ultrabeat to a drum machine that features 24 drum pads plus a builtin three octave keyboard. Ultrabeat’s 24 drum pads are assigned to the first 24 keys of a standard MIDI keyboard (corresponds to MIDI notes C1-B2).
The step sequencer is located at the bottom of the Ultrabeat window. It can be used in place of, or in addition to, MIDI notes entering Ultrabeat’s input section (from Logic) to control sounds. Saving and Loading Settings In order to save and reload Settings, Ultrabeat uses the procedure common to all Logic plug-ins: as is customary, settings are saved and loaded via the Settings menu. An Ultrabeat setting contains: • The drum kit, which consists of 25 sounds, inclusive of assignment and mixer settings.
You can also select the sound by using MIDI note input. To do so, activate the Voice Select button in the upper left hand corner of the Plug-in window. Note: The automatic sound selection function activated by the Voice Select button is useful for quick selection of different sounds, which are then displayed for editing.
The Drum Mixer The assignment section contains a mixer for the 25 sounds found in an Ultrabeat drum kit. It allows you to adjust each sound’s volume and pan position, and also offers a Mute and Solo button. Volume The individual volumes of all sounds are indicated by blue bars, providing a complete overview of all levels within the kit. You can adjust the volume of the sound, in relation to Ultrabeat’s total output level, by dragging the blue bar beneath the sound name.
The Synthesizer The majority of Ultrabeat’s user interface is dedicated to creating and shaping individual drum sounds; in short, Ultrabeat’s synthesizer. The parameters of the drum sound selected in the assignment section are displayed in this synthesizer section. Note: Despite its vast feature set, Ultrabeat’s user interface only requires a single Plugin window.
Modulation Ultrabeat was developed with special attention paid to dynamic sound shaping possibilities. To this end, almost every sound parameter can be modulated. Ultrabeat provides two powerful LFOs, four new types of envelope generators (Env 1–4), velocity, and four freely-definable MIDI controllers as modulation sources. The settings for the LFOs and envelope generators are represented graphically. They are located above and below Ultrabeat’s output section.
Control Elements All parameters can be set by clicking or grabbing them, and moving in an upward or downward direction. Note: If you hold Shift before clicking and moving a control, its value can be finetuned. Repeated clicks on buttons steps through different operating states. Move the mouse vertically while holding down the mouse button to adjust values in number fields.
The Synthesizer Parameters In this section, you’ll find a description of the individual parameters found in Ultrabeat’s synthesizer section. A discussion of the signal flow can be found in “The Signal Flow” section, on page 306. The Sound Sources A drum voice in Ultrabeat has four sound sources: two multi-synthesis capable oscillators, a noise generator and a ring modulator. Oscillator 1 To use Oscillator 1, you need to first turn it on.
The curved slider to the left of the Volume knob controls the pitch of the oscillator in half step intervals. If you press Shift, you can adjust the pitch of Oscillator 1 in cent intervals. The pitch value is displayed numerically to the left of the slider. You can change the displayed value by click-holding directly on the value field, and moving the mouse vertically. Pitch can be modulated by the sources found in the Mod and Via menus.
The Asym parameter can be modulated by the sources found in the Mod and Via menus. This allows you to create dynamic sound changes at the oscillator level. The effect of the Mod and Via modulations are adjusted with the small sliders to the left and right of the Asym slider. The range affected by the modulations is colored blue (Mod) and green (Via). If no source is selected in the Mod and Via menus (set to Off ), the Mod and Via sliders remain hidden.
Note: The Filter Bypass button simply determines the signal flow. It doesn’t turn the oscillator on or off. Use the oscillator On/Off button for this (see above). Oscillator 2 To use Oscillator 2, you first need to turn it on. This can be done with the On/Off button in the lower left corner of the Oscillator 2 section. When active, the button is red. Note: When you program a drum sound, you can turn the individual sound sources on or off with the corresponding On/Off buttons.
Phase Oscillator The waveform of the Phase Oscillator can be “twisted” with the Slope, Saturation, and Asymmetry parameters, and shaped into almost any basic synthesizer waveform. The effects of these three parameters are graphically illustrated in the waveform display within the oscillator section. Setting all three parameters to zero values will cause the oscillator to produce a sine wave. The Slope parameter determines the slope or steepness of the waveform.
Samples are selected in a dialog box, which can be reached by clicking on the arrow (or no sample loaded text) in the upper left corner (or top) of the waveform display. In addition to the supplied Ultrabeat multi-layer samples, it is also possible to use this dialog to select and load an audio file of your own choice. It should be noted, however, that the velocity layering function is not available for such samples.
Model This oscillator type offers a physical model of a string instrument for the creation of percussive sounds. The parameters at your disposal are based on the physical properties of a real string. Two contrasting exciters, each with different sound characteristics, are available. You can toggle between them with the corresponding buttons (Type 1 and Type 2). Note: In Ultrabeat’s oscillator 2 Model, an exciter is the agent or triggering device used to initiate the vibration of the string.
In contrast to the other parameters of the Model oscillator, Resolution does not reproduce a pre-defined real-world property of the physical model, but affects the modelling process itself: higher values lead to an improved calculation resolution which results in more overtones. Lower values reduce the precision of the calculations, leading to fewer overtones and often to inharmonic spectra.
Between the ring modulator and the filter section you’ll find a signal flow switch that controls the routing (Filter Bypass button). Repeated mouse clicks will send the signal to the filter (Filter Bypass button turns red), or bypass the filter and send it directly to the EQ section (Filter Bypass Switch remains gray). The direction of the arrow on the Filter Bypass button illustrates the routing. Note: The Filter Bypass button determines the signal flow. It doesn’t turn the ring modulator on or off.
The bandpass (BP) filter only allows a certain frequency range (a frequency band) centered around the Cutoff frequency to pass. It can be used in the upper, as well as at the lower, end of the frequency spectrum to reduce the highs and lows of a sound. The Cut knob determines the Cutoff frequency, and defines the point in the frequency spectrum where reduction begins. Depending on the type of filter you select, you can make a sound darker (LP), thinner (HP) or more nasal (BP) by adjusting the Cut value.
The Filter Section The output signals of both oscillators, the ring modulator and the noise generator are passed on to Ultrabeat’s central filter section (if they haven’t bypassed it through use of the various Filter Bypass buttons). The filter section offers a multimode filter and a distortion unit. The order that sounds are passed through the filter and distortion unit is determined by the red arrow found at the “equator” of the filter section.
The abbreviation BR stands for Band Rejection filter. In this mode, the area (the frequency “band”, to be more exact) around the Cutoff frequency is filtered out while frequencies that lie further away (from the Cutoff frequency) are allowed to pass. The mid frequencies become softer and the low and high frequencies remain unchanged. Below the filter type buttons, you’ll find two buttons labeled 12 and 24. These allow you the select the slope of a filter.
The Distortion Unit Depending on the order determined by the red arrow in the filter section, the distortion unit is inserted either before or after the multimode filter. It provides either a bit crusher or distortion effect. The desired mode is activated by clicking on the Crush or Distort button. The active effect is indicated in red.
The shelving EQ is activated by pressing the upper of the two EQ Type buttons. The peak EQ is activated by pressing the lower of the two EQ Type buttons. Note: The shelving filter in band 1 offers a low shelving EQ while the shelving filter in band 2 features a high shelving EQ. Low shelving means that the frequencies below the set frequency are affected. High shelving affects frequencies above the set frequency. Note: Shelving EQs function similarly to synthesizer lowpass and highpass filters.
Pan Modulation/Stereo Spread The EQ’s output signal is passed along to the pan/spread section. In the pan/spread section, the placement of the sound in the stereo field (set in the assignment section’s mixer) can be modulated (Pan Modulation mode), or the stereo basis of the sound can be broadened (Stereo Spread mode). Activate the desired mode by clicking on the appropriate button (Pan Mod or Spread). If neither mode is activated, the signal passes through unaffected.
Trigger and Group Menus The manner in which Ultrabeat reacts to a succession of incoming notes is individually defined for each sound. These parameters are found in the output section, below the Voice Volume knob. Clicking the button below the Trigger label opens the Trigger menu, allowing you to choose between Single and Multi trigger modes. • Single: A new trigger note cuts off the (same) note that is currently playing.
Modulation Numerous sound parameters can be controlled dynamically (modulated) in Ultrabeat. The setting of modulation routings follows a universal principle that is explained in this chapter.
As soon as a modulation source is selected in the Via menu (Vel in this example), a movable slider appears on the Mod ring. Grabbing and moving this slider with the mouse allows you to set the maximum modulation value that can be reached through use of the Via source (0.90 in this example). So much for the settings.
Here’s another example: Cutoff is again set to 0.50, Env 1 now drives the value down to 0.25, and a maximum Ctrl A value reduces the Cutoff frequency down to 0. Here is another example that illustrates the simplicity and speed of Ultrabeat’s modulation options: In this example, you won’t just be changing the modulation intensity of Env 1 (which affects Cutoff ) with the dynamics of your performance (Vel), but you’ll also control its direction as well.
Note: The Max setting produces a static modulation at maximum level. When the Mod value is set to Max, the Via parameter is routed directly to the modulation target. This way, velocity can be used as a direct modulation source, even though Vel is not available as a source in the Mod menu. Another example would be to set up an external MIDI fader unit with Ctrl A, B, C, or D (see below).
Note: The speed of the LFO in Ultrabeat can reach up to 100 Hz which, when compared to analog synthesizers, offers a number of far-reaching possibilities. Ultrabeat has two LFOs that offer identical feature sets. The parameters for each are described jointly; you can, of course, adjust LFO 1 and LFO 2 completely independently of each other. The buttons labeled 1 and 2 select the corresponding LFO, allowing adjustments of each LFO’s parameters.
The Cycle parameter can also determine whether the LFO (waveform) is started from the beginning (at a zero-crossing point) with each note trigger, or whether it simply continues oscillating. A Cycle value of Inf (Infinity) forces the LFO to run freely. It is not reset by incoming MIDI note on messages. When Cycle is set to values under 100, the LFO will be reset by each new MIDI note on message (Note On Reset).
In the envelope graphic, you can see various junction points of two different sizes. Both of the larger handles on the x-axis (the horizontal, or time axis) control the attack and decay times, respectively. A vertical line extends up from the first of the two handles, and divides the envelope into an attack and decay phase. Both segments each have two small curved junction points. You can move these in any direction to deform the contour of the envelope, and freely shape its amplitude.
Note: When you modulate Time, increasing velocity values lead to a reduction in length of the envelope segment. Lower velocity values increase the length of the envelope segment. Sustain Activation of the Sustain button causes a red handle (and vertical line) to appear on the x-axis. This can be moved horizontally—but only within the decay segment area. The amplitude that the envelope reaches at the Sustain junction point is retained until the MIDI note is released.
The Step Sequencer Principle The basic idea behind analog step sequencers was to set up a progression of control voltages, and output these step by step. In early analog sequencers, three control voltages were usually created per step, in order to drive different parameters. The most common usage was control of a sound’s pitch, amplitude, and timbre (Cutoff ) per step.
• The actual sequencing takes place in the step grid above. In this section, a pattern of 32 steps is shown for each sound. The pattern grid of the sound that is currently selected in the assignment section is shown. You can add or remove events to the grid by simply clicking at the desired step position. Parameter values in the individual steps are altered by grabbing and moving them with the mouse. Global Parameters A description of the parameters that apply globally to all internal sounds follows.
Pattern Parameters A pattern has a maximum number of 32 steps and contains the total events of all 25 sounds. At the bottom edge of the Plug-in window you can select from one of 24 patterns, and set global parameters (for each pattern) that apply to all sounds. Pattern # (Pattern Number) Click the field next to the Pattern # label to open the Pattern menu, which allows you to choose one of the 24 patterns.
Step Grid (The Heart of the Sequencer) In the step grid area, the pattern is rasterized and displayed in numerous rows and steps. The displayed grid applies to the sound that is currently selected in the assignment area. Choosing a different sound (either via the onscreen keyboard/ assigment section or via MIDI) switches the grid display to match the newly chosen sound.
Swing Enable Activation of the blue Swing button to the left of the trigger row stipulates that the grid of the currently selected sound will be played in accordance with the Swing knob setting. Only even-numbered steps are affected by the Swing parameter; exactly which beats this corresponds to depends on the selected Resolution parameter setting, as demonstrated by the following example. At a Resolution of 1/8 and a Length of 8, the notes on steps 1, 3, 5, and 7 represent quarter notes in the measure.
• Alter Gate (Time): Randomly changes the note lengths of all steps while retaining the selected beats (the trigger row remains unchanged). • Randomize Vel(ocities): Same as Alter Velocities, but random parameter alteration is more pronounced. • Randomize Gate (Time): Same as Alter Gate, but random parameter alteration is more pronounced. Using MIDI to Control the Sequencer As mentioned earlier, pattern performance can be influenced by incoming MIDI notes.
• Sustain—The reception of a MIDI note starts the pattern and it continues playing in an infinite loop until the corresponding MIDI note is released. • Toggle—The reception of a MIDI note starts the pattern and it continues playing in an infinite loop until the next note is received. If it is the same note, the pattern stops immediately. If it is a different note, the sequencer immediately switches to the new pattern.
Importing Sounds Ultrabeat allows you to import individual sounds. Please proceed as follows: 1 Click on the Import field to the right of the Voice Auto Select function. 2 Navigate the dialog box as you would normally until you find the setting that you want to import sounds from. 3 After selecting the setting, a list of all the sounds found in this setting will open up in a “drawer” next to the mixer section. 4 Select a sound in the list and click on its name while holding down the Control key.
Tutorial: Creating Drum Sounds in Ultrabeat Now that you’re acquainted with all of Ultrabeat’s features, we’d like to offer you a few specific sound creation tips in the following section. Please take the time to explore the vast and complex possibilties available to you in Ultrabeat, using the following programing tips as a starting point. You’ll discover that there is hardly a category of electronic drum sound that Ultrabeat can’t create easily.
Kick Drum Electronically produced kick drum sounds are primarily based on the sound of a deeply tuned sine wave. To program this type of sound in Ultrabeat, please proceed as follows: 1 Load the Default Tutorial setting. Note that Oscillator 1 is in Phase Oscillator mode. 2 Find a suitably tuned pitch in the lower octaves by soloing the bass drum along with other important tonal elements of the song (a bass or pad sound, for example). Use the Osc 1 Pitch slider to adjust the pitch until appropriate.
5 Change the Mod amount (the blue control) of Osc 1 Pitch again (see step 1). The interaction of this parameter with the envelope’s Decay time provides numerous possibilties for shaping the “kick” or “punch” of the bass drum sound. Note: This simple bass drum sound is called “Kick 1” in the tutorial set, at a pitch of C1. Removing Tonality One advantage of bass drums based on sine waves is that their sound can be precisely tuned to match the song.
4 Make the settings shown in the following graphic to the filter section: Filter type = LP 24, Cutoff value = 0.10, Mod Source for Cut = Env 3, Mod Amount for Cut = 0.60, Resonance = 0.30. 5 Set the Attack time of Env 3 to zero. Use the Decay time of Env 3 to shape the sound of the filtered bass drum. 6 You may also choose to control the filter resonance with an envelope. Make sure you dedicate a single envelope to this function (in this case, use Env 2 as a Mod source for Res).
More Contour… In our example, all four envelopes are being used. Take some time to play with the shapes of the envelopes, while maintaining the Attack and Decay settings. Experiment with the junction points of the Decay phase in the different envelopes to familiarize yourself with the sound shaping options available.
Snare Drum The sound of an acoustic snare drum primarily consists of two sound components: the sound of the drum itself and the buzzing of the snare springs. Try to approximate this combination in Ultrabeat with a single oscillator and the noise generator. Start programming your snare drum as follows: 1 Begin again with the Default Tutorial setting. Deactivate Oscillator 1, and switch Oscillator 2 on (in Phase Oscillator mode).
Now refine the snare drum sound using FM synthesis: 1 Turn on Oscillator 1 in FM mode. Use Env 1 to control the volume of Osc 1 as well. 2 Choose a pitch for Oscillator 1 that’s about an octave lower than Oscillator 2. Consciously avoid even intervals between the oscillators and detune them slightly from each other. As an example, try a pitch setting of F#2 in Osc 2 and E1 in Osc 1, then fine tune Osc 1 a few cents higher by holding Shift while adjusting its Pitch slider.
To complete the 808 emulation, add some noise: 1 Switch the noise generator on, and activate the highpass mode in its filter (HP). Set the Cutoff value to about 0.65, Resonance to 0.35 and add a little Dirt (around 0.06). 2 The noise generator provides the sustained snare sound. It should be shaped by its own envelope, independent of the decay phases of both oscillators, in order to get 808-like results. Changing the volume of the noise generator simulates the “snap” parameter of the 808.
You can increase the performance dynamics with the following steps: 1 First, reduce the values of the individual volumes by turning down the Volume knobs in both oscillators and the noise generator. Note how the Mod ring and its Via sliders also move back. Change the Via slider positions until all three Volume knobs look like this: 2 If you use differing intensities for each Volume knob when completing step1, you’ll have the potential of individual velocity reactions for each sound component.
5 Set the additional control that appears (as shown), to control the character of the sound with velocity: 6 Repeat this with the other parameters of Oscillator 2, as well as pitch: 7 Within the noise generator, you’ll be working with negative modulation (the position of the marking is below that of the base parameter value), and will modulate one parameter directly (Max setting in the Cut Mod menu), and another indirectly (the LFO 2 setting in the Dirt Mod menu): Our sound is now nothing like an 808 sna
The Kraftwerk Snare A further classic electronic snare drum sound is the highly resonant lowpass filter of an analog synthesizer that quickly closes with a “snap”. This sound was used extensively by Kraftwerk. Proceed as follows to recreate this sound with Ultrabeat: 1 Choose the “Snare 1” sound to start with. 2 Direct the signals of both oscillators and the noise generator to the main filter. 3 Modulate Cutoff with Env 1 (this is already modulating the volume of the noise generator).
Hi-Hats and Cymbals Electronic hi-hat sounds are very easy to create in Ultrabeat. Please proceed as follows: 1 Load the Default Tutorial sound. 2 Switch off Oscillator 1 and turn on the noise generator. Choose the following settings for the noise generator: 3 In the GUI detail above, you can see, that the Cutoff parameter is modulated by Env 1. The modulation is negative, the position of the Mod slider is below that of the base parameter value. 4 Use rather short Decay values for Env 1 and Env 4 ().
Metallic Sounds If you want to create metallic sounds with Ultrabeat, the ring modulator and the Model oscillator are the ideal tools. To use the ring modulator, proceed as follows: 1 Load the Default Tutorial sound. 2 Activate a Phase Oscillator and the Model oscillator. Choose a pitch for each oscillator above C3 so that a slightly detuned interval is created. 3 In the Material Pad of the Model oscillator, choose a setting with plenty of overtones as in the graphic below.
Programming in Building Blocks As you become familiar with drum sound programming, you may begin thinking in “building blocks”. By this we mean that you might realize that drum sounds usually consist of different components. Once you’ve mentally, or physically, written down your “list” of components, you should try to emulate each component that contributes to the sound’s character—making use of the different sound generators available in Ultrabeat.
24 Sculpture 24 Sculpture is a synthesizer plug-in that generates its sound based on a simulated vibrating string or bar. To keep things clear, we will always refer to the “String” throughout this chapter, even though many of the possible sounds that you can create with Sculpture have nothing in common with what you’d expect from a stringed instrument! The Graphical User Interface (GUI) of Sculpture is broken down into three main areas. The silver section at the top contains the sound engine.
Sculpture uses a method of synthesis called component modelling. This approach to tone generation shares some aspects and parameters with other synthesis techniques, such as those found in additive and subtractive synthesizers. As such, many of the parameters used by Sculpture will be immediately familiar to you, such as LFO’s, Vibrato, Envelopes, and so on. Many others, however, will be very new.
The String is the central synthesis element. It offers a range of parameters that allow you to adjust its material—what it’s made of, in other words. Up to three Objects of different types are used to excite or disturb the vibration of the String. These Objects can be positioned anywhere along the String, and offer multiple parameters for adjustments to their properties. The String itself doesn’t make a sound unless it is stimulated by the Objects.
Sculpture goes far beyond the mere creation of an infinite number of base timbres, however. One of the key differences between Sculpture’s String and a traditional synthesizer’s waveform is that the base timbre (provided by the String) is in a constant state of flux. By this, we mean that if Sculpture’s String is still vibrating for a specific note, retriggering that same note will interact with the ongoing vibration.
Sculpture’s Parameters The following sections discuss the parameters of Sculpture. Before touching on them, however, a key factor with component modelling is the interaction between various sections of the synthesis engine. This can lead to some truly unique sounds, but can also lead to unexpected results. Sculpture is very different to traditional synthesizers, and requires a more measured approach to achieve a particular end result.
Most parameters can be reset to their default value by clicking on the desired control, while holding down Option. Note: When adjusting most parameters, keep an eye on the small help tag that pops up while the mouse button is depressed. This will provide you with an accurate indication of the current parameter value, allowing you to make precise changes. Global Parameters These are found across the top of the Sculpture GUI, unless otherwise specified.
Voices When Keyboard Mode is set to Poly, this parameter limits the number of simultaneously sounding voices to the set value. A value of 16 voices is the maximum polyphony of Sculpture. Keyboard Mode Here, you can select between the Mono, Legato or Poly Keyboard Modes. You can switch between mono and poly modes by clicking on the Poly and Mono buttons. The Portamento, the duration of which is set by the Glide Time parameter, affects legato performances.
Bender Range Up/Down These parameters are found below Object 3, on the left-hand side of the Sculpture GUI. Separate settings are available for upwards and downwards pitch bends—using your MIDI keyboard’s pitch bend controller. When Bender Range Down is set to Linked, the Bender Range Up value is used for both (up/down) directions. The range is set in semitone increments: +12 and−36 respectively.
The string Release parameters impact on the vibrations of the string once the key has been released. The Hide button is handy for avoiding accidental parameter changes, and simplifies the interface. The Material Pad The following two string material parameters determine the general timbre, and are controlled by the “ball” (which correlates to the X and Y co-ordinates) within the Material Pad. The “crosshair” is a handle for the Key Scale/Release Scale diamonds in cases where these are hidden by the ball.
Inner Loss Scale Release Values above 1.0 cause the inner losses to increase when the key is released. This is quite unnatural, as this would mean that the string material would change after the note was released. In practice, however, the use of this parameter in combination with Media Loss Scale Release allows a natural simulation of strings that are dampened at note-off time. To adjust, first select the Release button, then click-hold on the blue Release line, and drag up/down to the desired position.
The String Parameters Around the Material Pad Resolution (Harmonics) This parameter determines the maximum number of harmonics contained in (and spatial resolution of ) the sound at C3. This is roughly proportional to the required CPU power, so the more harmonically-rich/the higher the Resolution setting of the sound, the more processing muscle will be required. Note: As you alter the Resolution value, you are changing the interaction of the string with the Objects.
Media Loss Release The blue slider (in the outer ring of the Material Pad) controls the Media Loss release time. To activate it, you must first click on the Release button, to the bottom right of the Material Pad. Values above 1.0 cause media losses to increase when the key is released. This parameter can be used to simulate a string that is dropped into a bucket of water after initially vibrating in air, for example.
Excite/Disturb Object Parameters The following parameters are used to excite, disturb or dampen the String. Important: At least one Object must be used, as the String itself does not make any sound! As you’ll discover shortly, there are a number of different string excite “models” such as blow, pluck, bow, and so on.
The repositioning of Objects changes the timbre of the String. If emulating, say a guitar, changing an Object position could be viewed as similar to picking or bowing a string at various spots along a fretboard. About Objects and Velocity Sensitivity It is important to note that: • Object 1 is velocity sensitive. • Object 2 is only velocity sensitive when a Type that actively excites the string is selected. • When damping Objects are used, Object 2 is not velocity sensitive.
The following table lists all excite Types available for Objects 1 and 2, and information on the “controls” available for each. Click-hold on the Type panel for these objects, and select from the list.
The following chart lists all Disturb/Damp Types available for Objects 2 and 3. Click-hold on the Type panel of these Objects, and select from the list. 370 Name Description Strength controls Timbre controls: Disturb A disturb object that is placed at a the hardness of fixed distance from the string’s the object “resting” position the distance from the resting position.
Gate Determines when the Object is active—in other words, when it disturbs/excites the String. Options are: • KeyOn—between note on and note off. • Always—between note on and the end of the release phase. • KeyOff—triggered at note off, and remaining active until the voice is released. Note: If using an Object Type such as Gravity Strike, the note may retrigger when you release the key. To avoid this artefact, set Gate mode to Always.
Position (morphable) Determines the position of each Object along the String (A value of 0.0 means one end, and a value of 1.0, the other end of the String). To adjust, simply click-hold and drag the corresponding numerical slider handle (the 1, 2 or 3 “arrows”) for each Object. Adjustment of these Object pickup positions will disturb/excite a given portion of the string. Object 1 can be an exciter. Object 3 can be a damper. You’ll note that Object 2 has two arrows.
Processing The “processing” tools covered in this section act on a per-voice basis, as do the String parameters discussed in the previous chapter. Pickup Parameters • The transparent bell curves represent the position and widths of Pickups A and B. • The green horizontal line within the Pickups window represents the String. As the Stiffness of the String is increased, the line will become thicker. • The vertical orange lines represent the positions of disturb/excite Objects 1, 2, and 3.
Stereo (Key) Panning position (left/right) is determined by MIDI note number. Dependent on settings, the further up/down the keyboard you play, the more the voice signal is panned left/right. Stereo (Pickup) Spreads the two pickups across the stereo base. In other words, the pickup position, combined with this parameter, will be spread further towards/from the left/right stereo channels.
You can adjust both slider halves simultaneously by dragging in the space between them. Important: The Attack time parameters of the Amplitude Envelope have a major impact on the way a single note is retriggered. When both Attack Soft and Hard are set to value of zero, the vibrating string is retriggered. If either of these parameters is set to a value above zero, a new note will be triggered. Decay Defines the Decay time.
Input Scale (morphable) This is a bipolar parameter. Negative values attenuate, and positive values amplify, the input signal prior to processing by the Waveshaper. When set positively, this results in a richer harmonic spectrum. The level increase introduced by the parameter is automatically compensated for by the Waveshaper. Given its impact on the harmonic spectrum, Input Scale should be viewed/used as a timbral control, rather than a level control.
Filter Parameters These parameters offer further timbral/spectral control over your sound. They should be pretty familiar to you if you have any experience with synthesizers. On/Off The Filter button activates/deactivates the filter. Filter Type Buttons The five buttons at the bottom of the filter section determine the filter mode. Choices are: • Hipass—Allows frequencies above the Cutoff Frequency to “pass”. As frequencies below the Cutoff Frequency are suppressed, it’s also known as a Low Cut Filter.
Resonance (morphable) Sets the filter resonance value. For highpass and lowpass modes, the Resonance parameter emphasizes the portions of the signal which surround the frequency—as defined by the Cutoff value. In Peak, Notch, and Bandpass modes, Resonance controls the bandwidth. Key This knob adjusts the key tracking of the Cutoff Frequency. Put plainly, the further up/ down the keyboard you play, the more bright/mellow the sound becomes.
Post Processing The “post processing” tools covered in this section impact on the summed stereo signal of all voices, rather than on a per-voice basis. Stereo Delay This is a (song) tempo-syncable stereo delay. It may also be set to run freely (not synchronized). On/Off The Stereo Delay button enables/disables the Delay section. Wet Level The Wet Level knob sets the level of the Delay output (wet signal). The parameter value is expressed as a percentage (%). Dry level is 100%.
Input Balance This parameter provides true stereo panning. Adjustments allow you to move the (stereo) center of the Delay input to the left or right, without the loss of any signal components. This makes it ideal for “ping-pong” delays. Delay Base Time This parameter, coupled with the Sync setting, enables you to set the delay base time. This can be in either: musical note values—1/4, 1/4t (t = triplet) and so on—or in milliseconds.
Spread Positive values on the “y” axis (above the default, centered position) increase the delay time of the right delay line and decrease the delay time of the left delay line—in effect, “smearing” the delay times of the left and right channels. Negative values reverse this. Spread is useful for “wide” stereo delay effects. Groove This parameter (on the “x” axis) allows you to reduce the delay time of one delay line by a given percentage, while keeping the other delay line constant.
Model The Model pull-down menu allows you to select the model from the following list. Any selection will be reflected in the graphic window to the right: • Low Mid Hi—A broadband equalizer module, with individual controls (via the knobs) of Low, Mid, and High frequency ranges. • Guitar 1—Emulates the body of an acoustic guitar. • Guitar 2—Emulates the body of another acoustic guitar. • Violin 1—Emulates the body of a violin. • Violin 2—Emulates the body of another violin.
Formant—Shift This parameter shifts the formants logarithmically. A value of −0.3, for example, shifts all formants one octave downwards, and a value of +0.3 shifts the formants up one octave. A value of +1.0 shifts up by a factor of 10—from 500 Hz to 5000 Hz, for example. Formant—Stretch Stretches the formant frequencies, relative to each other. In other words, this parameter alters the width of all bands being processed by the Body EQ, extending or narrowing the frequency range.
• Click-dragging horizontally on the graph allows you to control the Formant Shift parameter. Level/Limiter Level Controls the overall output level for the instrument. Drag to adjust. Level Limiter Mode Clicking on the desired button activates/deactivates the integrated limiter. Options are: • off—disables the limiter. • mono—a monophonic limiter on the summed signal of all voices. • poly—a polyphonic limiter, that processes each voice independently. • both—a combination of both limiter types.
Each modulation generator allows you to select one (or in most cases, two) of the core synthesis parameters as a modulation “target”. The same target can be selected for all modulators, if desired. Modules such as the LFOs and Control Envelopes also offer “via” modulations (a freely modulatable amount/intensity—the scaling factor—of the modulation source output level). These are “sidechain” modulations.
The two Sample&Hold waveform settings output random values. A random value is selected at regular intervals, defined by the LFO rate. A modulation of Pitch leads to the effect commonly referred to as a “random pitch pattern generator” or “sample and hold”.
• a decay to a zero amplitude level. Phase Allows the choice between strictly monophonic or polyphonic LFO modulations with either; similar phases, completely random phase relationships, key-synced phase … or anything in-between. • If used polyphonically for modulation of multiple voices, the modulations will not be phase-locked. To explain, when used on polyphonic input (a chord played on the keyboard) the modulation is independent for each voice (note).
Target1/2 These two modulation destinations can be assigned per LFO, with an optional, additional “via” modulation. These parameters determine modulation destinations 1 and 2. To activate, click on either the 1 or 2 buttons (which will highlight the Target and via text/pull-down menus), then click-hold on the Target pull-down menu(s). Select the desired target, and release the mouse button. Via (source) 1/2 These parameters determine the source that controls the modulation scaling for each LFO.
There are two “special” rectangular waves: rect01 and rect1—the former switching between values of 0.0 and 1.0 (unipolar), and the latter between values of −1.0 and +1.0 (bipolar, like the other waveforms). See the “LFO 1 and 2” section, on page 385. Curve Allows you to define a freely-variable number of waveform variations, resulting in subtle/drastic changes to your modulation waveforms. The Curve parameter can even influence the Sine waveform type. • Curve value of 0.
Random Variations Jitter 1+2 The two jitter generators are special LFOs, designed to produce continuous, random variations—such as those of smooth bow position changes. The jitter generators are equivalent to general purpose LFOs set to a “noise” waveform. To activate the routing of the Jitter generators, click on the 1 or 2 buttons. Note: Jitter modulation of pickup positions as the Target produces great chorus-like effects.
Velocity Modulations The Excite Objects and the Filter have dedicated velocity sensitivity controls. Many other modulation routings also allow you to select Velocity as a via input source. In some cases, it may be of use/interest to directly control other synthesis core parameters by velocity. This can be done in this section—where two independent destination/amount/velocity curve slots are available. To activate either, click on the 1 or 2 buttons.
The Control Envelopes The two Control Envelopes are somewhat special, as they can be used as: • “traditional” four segment envelopes. • MIDI controller modulations. • a combination of both: as MIDI controller movement recorders (with ADSR-like macro parameters), for polyphonic playback. To select Envelope 1 or 2, click on one of these buttons. Mode—Ctrl/Env Buttons Selects either controller (“run” mode) or envelope functionality.
A large number of possible Targets are available, including; string, object, pickup, waveshaper, and filter parameters. To select, simply click-hold on the panel below the word Target, and make your choice from the list. Via (source) 1/2 The Via parameter defines the modulation amount for Envelopes 1 and 2. The Via panels allow the selection of sources that are used to scale the modulation amount of the envelopes.
• The lines on the background grid are placed 100 milliseconds apart. • The background lines are placed 1000 ms apart for very long displayed envelope times. In sync mode, this is displayed as 1 quarter. • The envelope is zoomed automatically after releasing the mouse button. This allows the display of the entire envelope at the highest possible resolution for the graphic Envelope window. • This behavior can be disabled/enabled by pressing the Autozoom button—the small magnifying glass.
You cannot move a node beyond the position of the preceding node. You can, however, move nodes beyond the position of the following node—even beyond the right-hand side of the Envelope window—effectively lengthening both the envelope segment and the overall envelope. When you release the mouse button, the Envelope window will automatically zoom to show the entire envelope. To adjust the level of each node, click on the desired handle, and drag it up or down.
To record an Envelope… To provide you with an Envelope recording example: • Set the Record Trigger Mode to Note+Ctrl. • Enable record by clicking on the R button. • Play, and hold, a key—and start moving the modwheel or whatever controllers are assigned to Envelope controls 1 and/or 2. To end an Envelope recording… An Envelope recording ends as soon as at least one of the following conditions is met: • The record button (R) is disengaged manually by clicking on it. • All voices are released.
Envelope Parameters The following parameters are only active if the envelope functionality is engaged (Mode set to either Env or Ctrl+Env). A-Time Velosens This slider is used to set the velocity sensitivity for the Attack time of the envelope. Positive values will make the Attack time shorter at minimum velocities, and negative values will make the Attack time shorter at maximum velocities.
Compare Following the initial (original) recording of an envelope, and any subsequent edits, this button allows you to toggle between the original recording and the edited version. Obviously, this is only available as an option if an envelope curve has actually been recorded. Sync/ms Buttons These parameters allow you to select between a free-running envelope (with segment times displayed in milliseconds) or a tempo-synced envelope with note value options, such as 1/8th or 1/4.
VariMod allows you to select a modulation source, and amount, to control the strength of the “deviation”. As an example; if you recorded a tremolo, as shown. You could trim the loop by dragging the envelope “handles” to fit the Attack peak, and the start and endpoints of a tremolo loop. • To do so, grab and drag the vertical lines that intersect the handles. (These are also highlighted when the mouse pointer touches them).
• Now, set up the VariMod source to CtrlA (which might be set to Touch, for example), and an amount above 0. You’ll end up with an interesting tremolo (or whatever you’re modulating with this envelope) with an arbitrary waveform, and a modulation depth that can be controlled by Key Pressure (Touch) or whatever controller is assigned to CtrlA. Copying Envelopes If you Control-click on the Envelopes, a context menu opens, offering Copy, Paste, and Clear options.
• The Morph Envelope, that can be edited either by segment (with the mouse), or recorded via MIDI controller movements. With a vector stick (Morph X/Y controllers) or mouse movements of the Morph Cursor (the “ball”), for example, on the Morph Pad. Morph Point Selection and Randomization There is always one of the five Morph Pad points (A/B/C/D/Center) that is selected for editing. This selected point is indicated by two concentric circles that surround it.
Randomizing The randomize feature allows you to create random variations of selected morph points. When combined with the copy/paste functionality that’s also available, randomizing lends itself to using the Morph Pad as a kind of sound cell culture device. Use of the Morph Pad can yield an interesting/inspiring composite sound. You can copy this sound to a corner of the Morph Pad (or several corners) and randomize it by a definable amount.
A Brief Randomizing Tutorial • Select the top button (5 points) in the Points section. Ensure that Auto Select is active. • Select the Int(ensity) slider, and drag it to a value of say 25%. • Now press the Rnd button. As you’re doing so, take a look at the parameters in the core synthesis engine. You will see a number of them move after the mouse click. • Now, click-hold on the Morph Cursor (the “ball”), and drag it to each of the corners in the Morph Pad.
The remaining entries in the Morph Pad contextual menu are to do with grouping of random parameters. Put another way, the following options allow you to determine which type of parameters you would like to randomize (via the Rnd button and Int(ensity)) slider. • All (Random Group)—this is your ticket to “wacky” sounds as all parameters in the • • • • following three groups are randomized. This can lead to some interesting results, but can be uncontrolled.
• As you move your mouse cursor along the line, or hover over the nodes directly, the current Envelope segment is highlighted. • You can create your own envelopes manually, by manipulating the nodes and lines, or you may record an envelope, as discussed in the “Morph Envelope/Record Path” section, on page 405. • To adjust the time between nodes, click on the desired handle, and drag it left or right. As you do so, the overall length of the Morph Envelope will change—with all following nodes—being moved.
Morph Envelope Parameters Mode Activates the Morph Envelope, and allows you to select from between the following modes: • Off—Morph functionality is disabled. • Pad only—envelope is deactivated, and morph functionality is controlled by the • • • • morph cursor (the “ball”) and/or X/Y MIDI controllers only. Env only—envelope is running, but the morph cursor and X/Y MIDI controllers are deactivated.
The loop and sustain point icons (the small “L” and “S”) can be directly grabbed and repositioned. Note that doing so can potentially alter the loop (and the overall Morph Envelope) length. • When set to finish, the Morph Envelope runs in “one shot mode” from its beginning to its end—even if the note is released before the envelope has come to its end. The other loop parameters are disabled.
Transition Provides control over the “transitions” between the morph points. This can range from the original (possibly recorded) movement to linear connections, and beyond this to “stepped” transitions. By the latter, we mean remaining at one morph state during the entire Morph Envelope segment, and then abruptly switching to the morph state set at the following envelope point. This parameter (and the Morph Envelope itself ) can lead to interesting evolving sounds or even rhythmic patches.
CtrlEnv 1/CtrlEnv 2 Sets the controller assignments for the two Control Envelopes—used as a modulation signal or an offset—in cases where the Control Envelope is set to Ctrl only or Ctrl+Env modes. It also is used to define the source for recording controller movements. Morph X/Morph Y Determines the controller assignments for the X and Y co-ordinates of the Morph Pad.
Basics Throughout the manual, we have followed the signal flow of the core synthesis engine. When programming “from scratch”, this is the approach you should also take, working on each component of the sound in isolation. Obviously, when you’re starting out with Sculpture, you won’t be familiar with the impact of each parameter on your end results. Don’t sweat it, we’ll provide some pointers for particular types of sounds in a few moments … but let’s get back to it.
• Press the Keyscale button at the bottom of the Material Pad “ring”, as shown in the graphic. • Strike and hold and/or repeatedly strike middle “C” on your keyboard. Middle “C” is the default pitch/key of the string. • While doing so, grab hold of the “ball” on the Material Pad by click-holding on it with your mouse, and move it around. Listen to the sonic changes as you move between the Nylon, Wood, Steel, and Glass materials.
• Adjust the Strength knob, by click-holding it, and moving your mouse vertically for large changes, or horizontally for fine adjustments. Strike a note repeatedly while doing so. The three string Object dials/controls are shown, along with the Pickup section at the center left. • Drag the Timbre and VeloSens arrowheads to different positions while striking a key • • • • to audition the changes that they bring.
• Do the same with the Pickup A and Pickup B sliders. You’ll note that changes to the Pickup positions result in quite different String vibrations (and tonal qualities). Adjust the Level control (directly opposite the Pickup section, on the right of Sculpture’s GUI) to increase volume, if desired. • Adjust the pickups of the other two Objects. Adjust each Object’s Strength, Timbre, and Variation parameters to alter the tone. Make use of the tables in the “Type” section, on page 368.
Processing From the Pickups, the signal is sent to the processing section, which consists of the ADSR-equipped Amplitude stage, a Waveshaper (with selectable types of waveshaping curves) and a multi-mode Filter. These processors and the Pickup parameters are covered in “Processing” on page 373. Feel free to experiment with these, while referring to the individual parameter descriptions. All elements that we’ve covered thus far exist on a per-voice basis.
A good understanding of the physical properties of the instrument that you are trying to emulate is obviously advantageous. This type of knowledge, however, is not common to most people, but it can be found online. You can certainly do some detailed research, but for most sound creation tasks with Sculpture, you can follow this general breakdown formula when creating your “string”.
Important: Before beginning, we’d like to stress that the following examples are just that—examples. There are many ways to model each component of the sound. We encourage you to experiment with the settings that are suggested to create your own “versions” of patches. You’ll note that specific parameter values are rarely given, and if they are, feel free to use another. Just to balance the ledger a little. Subtle changes—particularly when it comes to Keyscaling parameters—result in more “controlled” sounds.
Bells At a basic level, bell-like sounds are quite easy to produce with Sculpture. The creation of truly interesting bells involves a little more effort, but the harmonic richness and detuning during the decay/release phase makes all the difference … • Load your “plain vanilla” patch. • Set Object 1’s Type to Strike. • Move the Material Pad ball to the very bottom of the pad, and place it pretty much halfway between the Steel and Glass entries.
Brass Brass instruments are notoriously difficult to recreate with electronic instruments. Samplers do a reasonable job in the right hands, and with the right sample library, but they lack the “organic warmth” of a real live brass player. This is a simple and generic brass patch that can be played as a solo instrument or as a brass section. • Load your “plain vanilla” patch. • Set Object 1’s Type to Blow. • Activate Object 2, and set its Type to Noise. • Adjust the Strength of Object 1 to around 0.90.
Flute This can be used as the basis for most instruments in the “wind” family, including flutes, clarinets, shakuhachi, pan pipes, and so on. • Load your “plain vanilla” patch. • Keyboard Mode should, theoretically at least, be set to “mono”, as flutes and other wind instruments are monophonic. After you’ve set up the patch, experiment with this parameter while playing, and make your choice. • Set Object 1’s Type to Blow. • Set Object 2’s Type to Noise. • Set the Gate of both Objects to Always.
Guitar Guitar, lute, mandolin, and other plucked type instruments, including harps, can be created from this basic patch. • Load your “plain vanilla” patch. • Set the Voices parameter to a value of 6—there’s only 6 strings on a guitar. Obviously, pick 7 for a banjo, as many as possible for a harp. • Set Object 1’s Type to Impulse, if it’s not already there. • Activate Object 2 and set its Type to Pick. • Now move Pickup A’s position to the extreme right. • Move Object 2’s Pickup position to a value of 0.14.
Organ Organ sounds are amongst the easiest and quickest sounds to emulate in Sculpture as they have no release phase. This simplifies programming as there is no real need to set up Keyscaling parameters to create the basic tone. You may, however, wish to do so at a later stage—for modulation routing or specific sound design purposes. We encourage you to play notes/chords while you are adjusting parameters. This way, you can hear what each parameter is doing to the sound. • Load your “plain vanilla” patch.
Percussion Percussive sounds, such as drums, tend to share a similar type of envelope. They contain a “strike” element, where most of the sonic character is exhibited, followed by a short decay phase. The release phase will vary—dependent on the instrument itself (a snare drum as opposed to a woodblock), and the ambient space it is placed in—a cavern, a bathroom and so on. • Load your “plain vanilla” patch. • Set Object 1’s Type to Strike. • Activate Object 2, and set its Type to Disturb 2.
• Click-hold on the Stereo Pickup dial, and drag upwards until the 10.30/1.30 positions are reached. • Set the Level Limiter to both. • Save Setting as… with a new name. We encourage you to set up your own modulations for this sound. The most common thing that springs to mind is the introduction of vibrato into the sound after a short period. The creation of higher pitched solo string instruments is much the same as the example above but special attention must be given to ALL Keyscaling parameters.
We’ll be using this basic pad sound for the following examples. Don’t be shy about doctoring the “vanilla pad”—anything goes, so make use of the Filter, the Delay, EQ, and Waveshaper to create new sounds. Evolver • Load your “vanilla_pad” patch. • Click on the LFO 1 tab at the bottom of the GUI. • Press the 1 button, and play the keyboard. Not much difference, there, huh? • Now, click-hold on the Min/Max sliders, and drag left and right, while holding down a chord. Finally settle on a value of 0.15.
Modulations The modulation options can be very important for the emulation of acoustic instruments. As a simple example, the introduction of vibrato into a trumpet sound over time. Many “classic” synthesizer sounds also rely as much on modulation as they do on the basic sound source components—the VCO, VCF, and VCA. Here’s a number of quick modulation tips … • Let’s say you want to modulate the timbre of Object 2, with the LFO, for example.
Programming: In Depth This tutorial explains how you can program sounds with Sculpture from scratch. Based on Sculpture’s string model, you’ll learn how to use the individual sound shaping parameters in order to recreate the physical properties of an instrument in detail. Note: You will find the settings for these tutorials in the Tutorial Settings folder in the settings menu (in the head of the Sculpture Plug-in window).
The number of frets differs from bass to bass and depends on the scale length. We don’t need to worry about pitches higher than a single ledger line C; the actual functional range of this instrument is primarily in its two lower octaves—between E 0 and E 2. We should also mention the fretless electric bass. Like all instruments of this type, it is freely tunable and possesses a distinctive, individual sound.
The vibration of the strings is, of course, naturally hampered by several physical factors: the radius of motion of the string (antenode) is impeded by the left bridge or by the first fret that’s pressed down upon (and the frets in between). This can lead to the development of overtones which can take the form of anything from a slight humming or buzzing to a strong scraping or scratching sound.
Play some notes in the lower range. you’ll note that the sound is very muffled, hollow, and distorted. Before we adjust further parameters in Object 1, we need to set the position of the Pickup. This is accomplished in Sculpture’s Pickup window located to the left of the Material Pad (see the GUI detail above). You’ll find three trapezoidally shaped sliders, representing Objects 1 to 3. Both of the transparent bell-shaped curves help you to visualize the position and width of Pickup A and Pickup B.
In order to recreate the material properties of a set of round wound strings: 1 Move the ball in the Material Pad up and down at the left edge. Pay attention to how the overtones react. Move the ball to the lower left hand corner. The sound should vaguely remind you of the sound of a low piano string. As the overtones sustain too long, the tone sounds somewhat unnatural. 2 Move the ball upwards until you hear an acceptable sound.
We can simulate these disturbing elements with Object 2: 1 Activate Object 2, and choose the Bouncing Type menu item. The sound should now vaguely remind you of a mandolin tremolo. This is way too strong an effect for the kind of sound we’re after. 2 Move Object 2 all the way to the right (a value of 1.00). 3 Experiment with Object 2’s parameters. A discrete and realistic result can be achieved with the following parameter values: Strength: 0.33, Timbre: −1.00 and Variation: −0.69.
Once activated, the key-scaling function is used to adjust the timbre of the sound, independent of pitch. To do this, we’ll initially employ the Resolution parameter. This is normally used to set the balance between DSP load and sound quality. As the overtone spectrum is reduced at low Resolution values, this parameter can also be used to shape the sound.
When playing, you’ll recognize the smooth transition that takes place between the wiry, overtone-rich sound at the bottom end and the extremely dampened sound in the upper register. This (exaggerated) setting was chosen to clearly demonstrate the scaling principle in stringed instruments. In order to achieve an authentic sound and timbre, we recommend the following setting: In basses in particular, low notes sustain far longer than the high notes.
Modifying the Frequency Spectrum of our Basic Bass The scope for sound design, by altering the frequency spectrum of electromagnetic instruments, is far more flexible than that offered by acoustic instruments. In addition to the number of pickups, the choice of amplifier, the equalization setting within the amplifier and, last but not least, the physical properties of the speakers and their enclosing cabinet also play a major role.
8 Save this sound Setting, as we’ll need it for further modifications later. Please name it “E-Bass Fingered Basic EQ1”. Pick Bass Our basic bass is played with the fingers. In the following example we will simulate playing the strings with a pick. We’ll also use the Pick Object Type for this sound. We’ll make use of the Timbre parameter to adjust the relationship between the speed and intensity at which the string is struck.
Damping Playing with a pick is often combined with a damping technique that employs the ball of the thumb. The right hand, which also holds the pick, should physically lay on top of the strings at the bridge. This technique results in the sound having less overtone content but become more percussive and “punchy” at the same time. You can variably control the timbre of the sound through the angle and pressure of your hand while playing.
Harmonics Harmonics are single partials (overtones) of the overall sound. They can be heard by damping certain points along the string. This is done by lightly laying the fingers of the left hand (assuming a right-handed bass player) on the string (not pressing down) before the note is articulated. The first overtone, the octave, is achieved by placing your finger at the exact middle of the string, in effect separating the string into two halves.
Tip: If you turn Object 3 off, you’ll hear a sound that is reminiscent of a 1970’s Fender Precision Bass. 4 Save this Setting as “Flatwound Pick Damped”. To get a nice percussive sound a la “Bert Kaempfert”, proceed as follows: 1 Turn Object 3 back on. 2 Move both Pickups a little to the left (position 0.08). 3 Our virtual pick (Object 1) can also be moved a little further to the outside (position 0.10). 4 We can add the icing to the cake with the Body EQ. Turn the Low dial to its maximum value (1.00).
5 Move the ball down a little, and the sound becomes more wiry. The ball should now be located directly above the word “Steel” on the horizontal axis. From the models at our disposal, Strike is the most suitable for simulating a thumb physically striking the strings from above. This model is not, however, as appropriate for the slapped (popped) strings. It makes the most sense to choose the Pick model for this purpose. 6 To be safe, turn the Level dial to −25 dB. 7 Select the Pick model for Object 1.
Let’s adjust Object 2’s parameters to the following settings: 1 Set Timbre to 0.39. This corresponds to a fingerboard that runs almost parallel to the string. 2 Set the Strength parameter to 0.33. Note: Try some higher values as well. You’ll see that the sound becomes softer and softer until it’s completely dampened by the obstacle. 3 An appropriate value for Variation is 0.64. Despite the overtone-rich reflection, the string can still vibrate freely.
4 Adjust Object 2’s parameters to the following values: Strength 0.14, Timbre −0.05, Variation −1.00. 5 Object 2’s Pickup position remains at the far right; please enter a value of 0.99. You’ll note that the range between C2 and C3 already sounds quite acceptable, but the buzzing in the lower notes is still too strong. It is somewhat sitar-like, so keep this disturb model in mind when it comes to creating a “home-spun” sitar.
13 Vary the Object 3 Strength parameter. You’ll discover that the overtone content of the buzzing can be controlled very effectively. A Strength value of 0.25 is recommended here. 14 Save this Setting as “Fretless Roundwound#1”. Modulation and Detuning Detuning and ensemble effects are normally achieved using a modulation effect or by combining doubling and detuning. When using a fretless bass for a solo part, a broad chorus effect adds a nice touch.
8 In order to hear the effect, the modulation intensity (amount) has to be set. Familiarize yourself with this effect by moving the slider labeled Amt (amount) gradually to the right. Set it to a final value of 0.15, a moderate rate that doesn’t “wobble” too much. 9 Save this Setting as “Fretless Chorus Dry”. Tip: At the maximum stereo breadth, effects based on detuning are not as prominent, especially when the “beats” heard in the sound result from signal differences between the left and right channels.
8 Reduce the total level of the effect by setting the Wet Level dial to a value of 25%. 9 Save this Setting as “Fretless Chorus+Ambience”. This example shows that the stereo delay section can be used as a substitute reverb for small spaces. For sophisticated reverb effects, it’s best to process Sculpture’s output with one of Logic’s reverb plug-ins. Creating a “drowned” in delay effect: 1 Reload the “Fretless Chorus Dry” Setting. 2 Switch the stereo delay section on.
In the following sections, we’ll be demonstrating Sculpture’s ability to create somewhat spacy and less “organic” sounds, using several pad patches as examples. After having studied the modeling of bass sounds in the preceding sections, we’ll now introduce you to a totally different group of sounds. Provided that you’re willing to supply the necessary level of curiosity and time investment for experimentation, you’ll discover a wide variety of interesting and animated sounds.
Please test the abovementioned Object Types one after the other, and move the Object 1 (Pickup) slider, responsible for the exact position of the exciting agent, up and down the string while you’re playing. You will come to two conclusions. First, the sound is now sustained for as long as you hold the key down. Secondly, shifting the Object 1 slider, with the Bow Type selected, results in the most pronounced sonic changes.
The Envelope can now be recorded. We assume that your MIDI keyboard has a modulation wheel which outputs the corresponding MIDI controller message (CC number 1) and that option 1 ModWh is selected for control of Envelope 1 (CtrlEnv 1) in the dark bottom edge of the Sculpture window. 4 Click on the button labeled R, located in the upper right corner of the Envelope section below “Record Trigger”, to prepare the Envelope for recording. Select the Note+Ctrl option for the recording.
Select Loop Alternate in the Sustain mode menu below the Envelope graphic display. As the sustain point is found at the end of the Envelope, the Envelope repeatedly travels from the beginning to the end and backwards from the end to the beginning, creating a continuous flux within the sound. Summing up: we now have a rudimentary, but appealing and organic-sounding pad which we will use as the foundation for further shaping and refinement.
You will hear a pleasant “beating” or chorus effect in the sound, which makes it broader and more full, alleviating the unpleasant, dry character (preset 0002 lfo>pick pad). Another unpleasant aspect is that the sound is too strong in the mid frequency range and could use some equalization. We’ll use the Body EQ to correct this. Activate the Body EQ, and experiment with the Lo Mid Hi model (which is the standard setting). Our favorite setting is when Mid is reduced 0.5 and Mid Frequency is set to 0.
3 Adjust the Intensity to −0.40 with the slider below the Target menu, and reduce the Rate parameter to 1 Hz. There should be subtle inconsistencies in the pressure applied by the bow to the string. To better recognize the effect, temporarily increase the Intensity level. We’ll use the second Jitter modulator for random position deviations with the modulation target Pickup Pos A+B (pickup position A and B). 4 Activate Jitter 2 and choose the Pickup Pos A+B option in the Target menu.
To vary the sound with the Morph Pad: 1 When you move the ball to one corner, the corresponding “partial” sound is selected; you can recognize this by the gray blue arches that light up in the graphic display. Choose each of the four corners one after the other, and vary the sound by altering several parameters directly in Sculpture’s GUI.
3 Perform the randomization by clicking on the Rnd button. When you next move the morph ball, you’ll hear the variations you just created. We have now reached the end of our programming tutorial. By demonstrating how to create basic sounds, detailed emulations of various electric bass sounds and explaining how to approach the generation of synthesized sounds, we hope to have given you a few insights into the interplay of Sculpture’s functions and parameters.
25 KlopfGeist 25 KlopfGeist is an instrument that is optimized to provide a metronome click in Logic. KlopfGeist is inserted on Audio Instrument channel 64by default. Logic automatically assigns this channel to the Metronome Object, making KlopfGeist the synthesizer responsible for the metronome click. Theoretically, any other Logic or third-party instrument could be used as a metronome sound source on Audio Instrument channel 64.
KlopfGeist can operate as a monophonic or polyphonic (4 voice) instrument, as determined by the Trigger Mode radio buttons. There are two tuning parameters; Tune for coarse tuning in semitone steps, and one for fine tuning (Detune) in cents. The Tonality parameter changes the sound of KlopfGeist from a short click to a pitched percussion sound—similar to a Wood Block or Claves. Damp controls the release time. The shortest release time is reached when Damp is at its maximum (1.00) value.
26 EVB3 26 This chapter introduces Logic’s EVB3 virtual Hammond organ. An electro-mechanical Hammond organ, with tonewheels and a Leslie sound cabinet, cannot be replaced by anything other than an electro-mechanical Hammond organ, with tonewheels and a Leslie sound cabinet! Having said that: The EVB3 offers a truly portable alternative to this classic pop, rock, and jazz instrument. Concepts and Function The EVB3 software instrument mimics the sound, and use, of the Hammond B3 and Leslie sound cabinet.
If you’re familiar with the original B3, you’ll remember the inverted (black) keys of the lowest octave on each manual. These inverted keys are switches that recall preset registrations (a preset of your drawbar settings). This feature is emulated by the EVB3 as well, but has been improved significantly, given that you won’t need a screwdriver to change the registration settings of your presets.
Logic records the channel information of incoming notes. With most other MIDI and software instruments, this information is not used at all. This is due to Logic’s MIDI channel setting (in the instrument parameters), which has priority, and overrides the original channel information. This can be circumvented by setting MIDI Cha = All, which is recommended for the EVB3. This will force the original channel information to be used.
Keyboard Split The EVB3 can also be played perfectly with a single MIDI keyboard (one manual) that is only capable of transmitting on one MIDI send channel. You can split the keyboard in order to play Upper, Lower, and Pedal sounds on different keyboard zones. In the parameter field in the bottom center of the GUI, set Keyboard Mode = Split. Set the keyboard zones with the UL Split and LP Split parameters, in conjunction with the Set buttons. The abbreviations are for: Upper/Lower and Lower/Pedal.
If you select MIDI Mode = RK, every drawbar responds to a specific MIDI control change number, commencing with CC #70. (Non-drawbar parameters may be set using control change messages up to CC #118). If you select MIDI Mode = HS, all of the EVB3 drawbars will be controlled by just a few control change numbers—CC #80–82. Its values are intelligently mapped to all drawbars. The resolution of this technique is not particularly high (much like the original B3), but it works well.
The EVB3 Parameters The graphical user interface (Editor view) opens when you double-click on the EVB3 Input slot of an Audio Instrument object. You can open and close the wooden lid by clicking the button underneath the Volume control. Keep it open while reading this section of the manual, because we’ll be discussing every parameter in detail. Drawbars The principles of additive synthesis with sine drawbars is further explained in “Additive Synthesis With Drawbars” on page 481.
Pedal Drawbars The organ features two drawbars for the bass Pedals. The waveform of the bass is not a pure sine wave, but a mixed waveform, that realistically simulates the Hammond B3 bass. The two registers differ in pitch, and in the following ways: • the Lower 16' register contains more octaves • the 8' register has a more prominent fifth portion.
Tune The simulated tone wheel generator can be tuned in cents (percentages of a semitone). 0 c is equal to A = 440 Hz. Scanner Vibrato The vibrato of the organ itself must not be confused with the Leslie effect, which is based on rotating speaker horns. The EVB3 simulates both. The Scanner Vibrato is based on an analog delay line, consisting of several lowpass filters. The delay line is scanned by a multipole capacitor, which has a rotating pickup.
Note: Check out the chorus and vibrato effects, and compare them with the sound of the rotor cabinet simulation! The organ’s chorus sounds different to modern chorus effects (such as Logic’s Chorus plug-in). Many organ players rarely use the Scanner Vibrato, preferring to work with a Leslie, in isolation. Others, like B3 virtuoso Brian Auger, prefer the integrated organ vibrato over the Leslie. Percussion Percussion is only available for the Upper manual—as it does on an original B3.
If you engage percussion on a B3, the volume of the normal, non-percussive registers is reduced slightly. The Up Level parameter simulates this behavior, allowing you to define the volume of the Upper manual, with percussion engaged. Preset Keys and Morphing The Hammond B3 is equipped with 12 switches, located below the lowest octave of both keyboard manuals. These are the preset keys, and are laid out as an inverted keyboard octave (black keys, white sharps). They are used to recall drawbar registrations.
If you hold the cancel key (C) on your master keyboard with the small finger of your left hand, and sustain a chord with your right hand, you can trigger the chord with different registrations, by pressing the preset keys with the other fingers of your left hand. This results in an organ-specific gater type effect, which wouldn’t be possible with the right hand alone. Morphing You can switch between the presets of the Upper manual with any keyboard.
Max Wheels Calculating (emulating) all tone wheel generators consumes considerable CPU processing power. A reduction of this parameter value reduces the EVB3’s hunger for processing resources. Note: This will diminish some overtones, so you should not reduce this value if you’re after an ultra-realistic simulation. Tonal Balance Tonal Balance changes the mix relationship of the higher/lower sounding tone wheels. Positive values result in a lighter and brighter sound.
Condition Technical limitations of electro-mechanical drawbar organs with tonewheels cause some strange tonal artefacts, such as crosstalk. These quirks form an integral part of the B3’s charm. You can adjust the following parameters to define the age of your EVB3. Note: Read more about the Click parameters in “Click” on page 468. Drawbar Leak Even if all drawbars are at their minimum position, the tonewheel generators of the B3 aren’t completely quiet.
Filter Age The high frequency output signals of the B3’s tone wheel generators are filtered by bandpass filters. The center frequency of these filters varies as the capacitors age. Filter Age allows you to alter the center frequencies of the filters. This colors the sound of the jitter applied by Random FM and the background noise resulting from Leakage. (See “Leakage” on page 467 and “Random FM” on page 467.) This parameter also influences the intonation of the organ, if you use the pitch bender.
Pitch Compared to the original B3, the EVB3 offers several parameters to change its pitch behavior. Note: The Trans UM, Trans LM and Trans Ped parameters are explained in “Transposition (Octave Range)” on page 458. Stretch The EVB3 is tuned to an equal-tempered scale. As a deviation from this standard tuning, you can stretch the tuning in the bass and treble ranges, much like acoustic pianos (especially upright pianos).
When applying Warmth and Stretch, you should consider that these parameters may result in a detuned sound, which is similar to the overuse of a chorus effect. Straight tuning is nice too, so set Warmth to 0 if you’re after a “pure” sound. Pitch Bender, Brake Effect The Hammond organ has no pitch bender. As such, use of the pitch bender is not suitable for realistic simulations, but it does provide a number of creative options.
Effect Chain and Effects Bypass Effect Chain The EVB3’s signal flow is as follows: the organ’s signal runs through the equalizer, wah wah and distortion effects. You can choose between four different signal flow routings for the equalizer, wah wah and distortion effects in the Effect Chain pull-down menu. This treated signal is then fed into Reverberation and finally passed to the Rotor effect. A “classic” B3 patch would be: an EQ’ed organ, plugged into a wah wah pedal, amplified by an overdriven Leslie.
Reverb Box, Small, Medium, Large, Big, and Spring are the names of the reverb algorithms. Reverb level is defined by the Reverb parameter. A Reverb =0 value conserves processing resources. You can also select Bypass in the Reverb Mode pull-down menu, if you want to disable the reverb without changing its level. The reverb is always patched after the EQ, wah wah and distortion effects, but before the rotor effect. This means that the reverb always sounds as if it is played back through the rotor speaker.
Mode Mode allows you to enable/disable the wah wah effect. If you select Mode off, the effect is disabled. There are six different filter types available: • ResoLP (Resonating Low Pass Filter) In this mode, the wah wah will work as a resonance-capable low pass filter. At the minimum pedal position, only low frequencies can pass. • ResoHP (Resonating High Pass Filter) In this mode, the wah wah will work as a resonance-capable high pass filter. At the maximum pedal position, only high frequencies can pass.
The Tone control only affects the distorted portion of the sound, while the dry signal portion remains unaffected. This allows for very warm overdriven sounds that won’t become “scratchy” if you try to get more treble out of the instrument. Drive controls the amount of overdrive distortion. The output level is automatically compensated for, so there’s no need for another master volume control adjustment facility.
Proline & Horn IR This setting uses an Impulse Response of a Leslie with a more open enclosure. Split & Horn IR This setting uses an Impulse Response of a Leslie with the bass rotor signal routed more to the left side, and the treble rotor signal routed more to the right side. Rotor Speed The Rotor Speed switches work as follows: Chorale = slow movement, Tremolo = fast, and Brake stops the rotor. Speed Control Organ players generally switch between Choral and Tremolo.
Note: The Hammond B3 isn’t equipped with a Sustain Pedal. This allows you to make use of your MIDI master keyboard’s Sustain Pedal as a speed switch. Rotor Fast Rate Rotor Fast Rate defines the maximum possible rotor speed (Tremolo). While moving the slider, the Tremolo rotation speed is displayed in Hertz. Acc/Dec Scale The Leslie motors need to physically accelerate and decelerate the speaker horns in the cabinets, and their power to do so is limited.
Another Mode is Sync: The acceleration and deceleration of the horn and bass drum are about the same. This sounds as if the two were locked, but is only clearly audible during acceleration/deceleration. Additional Parameters A number of additional parameters are accessible via the 001/011 button at the top of the EVB3 Plug-in window. The Upper/Lower Stop Position sliders allow you to set an exact stop position for the Leslie horn or bass rotator, respectively.
This table describes the MIDI Control Change Message number assignment when MIDI Mode is set to RK. This is the correct setting if you use a Roland VK series or Korg CX-3 drawbar organ as a remote controller for the EVB3.
Controller Number MIDI Mode RK: Parameter Name 111 Distortion Drive 112 Distortion Tone Click Levels 113 Click On Level 114 Click Off Level Balance 115 Main Volume 116 Lower Volume 117 Pedal Volume Rotor Fast Rate 118 Rotor Fast Rate Only the controller assignments 80 to 82 differ, when MIDI Mode is set to HS. This setting matches the controller mapping of Hammond XB-series organs. The other controllers remain as described above.
Controller Number MIDI Mode 0: Parameter Name 24 Lower Drawbar 4' 25 Lower Drawbar 2 2/3' 26 Lower Drawbar 2' 27 Lower Drawbar 1 3/5' 28 Lower Drawbar 1 1/3' 29 Lower Drawbar 1' Vibrato 31 Upper Vibrato on/off 30 Lower Vibrato on/off Brightness Vibrato Attack Time Chorus Intensity Percussion Sostenuto Percussion on/off Release Time Percussion Harmonic (2nd/3rd) Sound Variation Percussion Volume Harmonic Content Percussion Time Equalizer 90 EQ Low 70 EQ Mid 5 EQ High Disto
Additive Synthesis With Drawbars The Hammond B3 is the classic drawbar organ. As with an acoustic pipe organ, the registers (drawbars, or “stops” on a pipe organ) can be pulled out, in order to engage them. But in contrast to a pipe organ, the B3 allows seamless mixing of any drawbar registers. The more you drag the drawbars down, the louder they will become.
Note: 2 2/3' is the fifth over 4'. 1 3/5', is the major third over 2'. 1 1/3' is the fifth over 2'. In the bass range, this can lead to inharmonic tones, especially when playing bass lines in a minor key. This is because mixing 2', 1 3/5' and 1 1/3' results in a major chord. Residual Effect The residual effect is a psychoacoustic phenomenon. Human beings can perceive the pitch of a note, even when the fundamental is completely missing.
Hammond also holds the patent for the electro-mechanical spring reverb, still found in countless guitar amplifiers today! The Hammond B3 was manufactured between 1955 and 1974. It is the Hammond model preferred by jazz and rock organ players such as: Fats Waller, Wild Bill Davis, Brother Jack McDuff, Jimmy Smith, Keith Emerson, Jon Lord, Brian Auger, Steve Winwood, Joey DeFrancesco, and Barbara Dennerlein.
The Leslie Don Leslie developed his rotor cabinets in 1937, and began marketing them in 1940. Laurens Hammond wasn’t keen on the concept of rotating speakers at all! Leslie’s approach was to simulate a variety of locations in the pipes, resulting in a new spatial perception for every note. The rotor speaker cabinets could simulate this effect, and the sense of space that they impart is incomparable, when placed side-by-side with any fixed speaker.
27 EVD6 27 This chapter covers everything about the EVD6, a virtual emulation of the Hohner Clavinet D6. The sound of the Hohner Clavinet D6 is synonymous with funk, but was also popularized in the rock, pop, and electric jazz of the 1970s, by artists and groups such as: Stevie Wonder, Herbie Hancock, Keith Emerson, Foreigner, and the Commodores.
You’ll appreciate the perfect integration of the EVD6 into Logic. In use, it’s much easier to handle than a real world Clavinet. There’s no need to transport a bulky and heavy instrument, or to attach any cables to it. In addition, the EVD6 eliminates the problems of reliability, getting new parts, and tuning—all of which are becoming increasingly difficult with the original instruments.
Global Parameters The Global Parameters are found in the lower-left portion of the EVD6 Graphical User Interface (GUI). Voices The voices parameter allows you to set the maximum number of voices that can sound simultaneously. Lowering the value of this parameter limits the polyphony and also the processing requirements of the EVD6. Minimal CPU power is used when the instrument is operated monophonically. There are two monophonic settings: mono and legato.
Warmth Amount of random deviation from an equal-tempered scale. High values add “life” to sounds. It can be useful for simulating an instrument which has not been tuned for a while, or for slightly “thickening” a sound. When playing chords, the Warmth parameter creates the warm detuning or beating effect between the chord’s notes. Click-hold, and use your mouse as a slider to adjust. Stretch The EVD6 is tuned to an equal-tempered scale.
Filter Switches The four filter switches emulate the original switches on the D6, with one exception. When all switches are set to off, you’ll hear the unfiltered sound, rather than the original D6’s “humming silence”. Simply click anywhere on each switch to toggle between its on/off position. Active switches are indicated by pale green lettering, and by being depressed towards the bottom of the Plug-in window. You may use the filter switches in any combination of on/off positions.
Stereo Spread Pickup—While the original D6 only has a mono output, the EVD6 has stereo capabilities. When both pickups are active (upper+lower and upper−lower modes), the two pickup signals can be spread across the stereo spectrum. To adjust the position of the stereo spread, click-hold on the up/down arrows in the lower half of the circular button—in the Pickup section.
Model The Model parameter allows you to select a basic type of tone, or “model”. Each model has its own unique tonal characteristic, and each is suitable for the creation of very different sounds. Each model is an instrument in its own right, and can immediately be played, without any further editing. We will discuss each model below, and encourage you to experiment with each.
Domin(ation) A powerful model with a strong and punchy attack—reacts more aggressively to velocity than other models. GuruFnk (Guru Funk) In the lower bass-octave ranges, the string oscillations become increasingly resonant over time, until they finally collapse (after 20 to 30 seconds). Higher notes have a much shorter decay, which also applies to their resonating behavior. This model invites heavy, funk-style bass playing in the lower octaves.
When using a setting with both pickups quite close to the upper end of the strings and Brilliant + Treble filter switches active, the fundamental tone is quite weak in the output signal. As such, you will mostly hear the overtones that are not exactly in tune for inharmonic models (Wood, for example). Try moving the pickups to the center, and deactivate all filter switches to circumvent this detuned effect. Level Sets the (post -Effects) level, in dB (decibels). Click-hold, and drag, to adjust.
Velocity Curve There are nine preset velocity curves available for the EVD6. These allow you to set up a curve which is suitable for your playing style, or the sound. The nine curves available are: fix25%, fix50%, fix75%, fix100%, convex1, convex2, linear (the default), concave1, and concave2. Excite Parameters Excite describes the string excitation, the physical power which stimulates the string to oscillate.
Random Controls the amount of random click level variations across the keyboard. This slider simulates the wearing of some hammers, but not all of them, emulating the real-world wear and tear of the original. The further to the right the slider is moved, the greater the variation between key clicks on some keys. If all the way to the left, all keys have the same level of key click. Range: 0.00 to +1.
Damping The Damping parameter allows you to modify the damping of strings. Damping is essentially a faster decay for the higher partials/harmonics in a sound, and is a property of the string material used (high damping for catgut strings, medium damping for nylon strings, low damping for steel strings). Sonically, damping results in a more mellow and rounded, or woody sound, dependent on the Model in use.
Pickup Parameters The original D6 is equipped with two electromagnetic pickups, much like those found in electric guitars: one below the strings (lower) and one above (upper). Pickup Position In contrast to the fixed pickups of the original instrument, the EVD6 pickups can be set to arbitrary positions and angles. To do so, simply click-hold on one end of the desired pickup (Upper or Lower) and drag the end to another position. Release the mouse button when done. Both values can be moved simultaneously.
Pickup Mode Pressing the AB and CD switches will change the virtual wiring of the two pickups. The current wiring, the EVD6 calls it Pickup Mode, is displayed in the Pickup Mode panel. You can also click directly on the Pickup Mode panel, and select the desired mode from a pull-down menu. • • • • C + A = Lower C + B = Upper D + A = Lower−Upper D + B = Lower+Upper Also see “Stereo Spread” on page 490, and “Pickup Switches” on page 489.
Distortion The integrated distortion effect can be adjusted in both intensity and tone. Range: Tone −2000 Hz to 20,000 Hz, Gain −0 dB to 20 dB. Using low Tone and Gain settings allows the Distortion unit to create warm overdrive effects. Bright and screaming distortion effects are produced with high Tone and Gain settings. Compressor Please note that the Distortion effect is always preceded by a compression circuit (shown in the panel above the Tone knob—with a ratio of 1:19.
Wah The typical Wah effect is generated by a dynamically moving filter. The EVD6 offers simulations of several classic wah effects, as well as some basic filter types. Possible values are: off, ResoLP, ResoHP, Peak, CryB, Morl1, Morl2. The abbreviations are for Resonant Low and High Pass filters, Peaking filter, CryBaby, Morley 1, and Morley 2. The latter three are famous effects pedal models that continue to be manufactured.
Wah Ctrl The Wah Ctrl parameter allows you to define the MIDI Controller (number/name) used as a manual wah effect control—a MIDI foot controller for example. You can also use MIDI Velocity to control the wah effect. Just click into the respective parameter field and select velocity from the ensuing pop-up menu. MIDI-control/coupling can be disabled by selecting off. Note: Both envelope and a manual controller can control the wah simultaneously.
High values lead to very deep, self-oscillating phase shifts, for those cutting (and ear and speaker damaging, so take care!) sounds. Chorus The Rate parameter adjusts the speed of the Chorus effect, and the Intensity parameter adjusts its depth. High Intensity values lead to ensemble-type effects. Ranges: Rate −0.10 Hz to 10 Hz, Intensity −0 to 100 Flanger The Rate parameter adjusts the speed of flanging, and the Intensity parameter adjusts the depth of flanging. Ranges: Rate −0.
A Brief History of the Clavinet German Company, Hohner, was the manufacturer of the Clavinet. Hohner were known mainly for their reed instruments (harmonicas, accordions, melodicas, and so on), but had made several “classic” keyboards, prior to the first incarnation of the Clavinet, known as the “Cembalet”. Musician and inventor, Ernst Zacharias, designed the Cembalet in the 1950’s. This was intended to be a portable, amplifiable version of the Cembalo, or Harpsichord.
The early models—Clavinet I with built-in amp, Clavinet II with tonal filters, Clavinet “L” with its bizarre triangular shape, all led to the Clavinet model “C”. This, in turn, was refined into the D6—a portable, amplifiable keyboard. The D6 used a hammer striking a string against a metal surface to produce its tone. It had a fully dynamic keyboard— as the striker is directly underneath the key, meaning the harder you hit, the louder and more vibrant the tone.
28 EVP88 28 This chapter covers Logic’s EVP88 virtual e-piano. The sounds of various Fender Rhodes pianos are among the most popular keyboard instrument sounds used in the second half of the twentieth century. The various Rhodes models have been popularized in a wide range of musical styles, ranging from pop and rock, electric jazz, jazz rock, soul, and in countless ballads, plus recent house and hip hop genres.
Incorporated into the EVP88’s front panel, you will discover an integrated effects processor which provides a number of classic effects popularly used on electric piano sounds. The algorithms featured in the effects processor have been specifically designed, adapted, and optimized for the EVP88. Included are: a great sounding equalizer, an overdrive, a stereo phaser, a stereo tremolo and stereo chorus.
Voices The voices parameter allows you to set the maximum number of voices that can sound simultaneously. Lowering the value of this parameter limits the polyphony and processing requirements of the EVP88. When the parameter is set to 1, the instrument is monophonic, and uses minimal CPU power. The maximum setting is 88, allowing for glissandi over the entire keyboard range with the sustain pedal depressed. A setting of 88 will, of course, be more processor-intensive.
Release The release parameter determines the amount of “damper” applied after the keys are released. Extremely long settings allow you to play the piano like a vibraphone. Bell Bell determines the level of the inharmonic treble portion of the tone. It is useful for emulating a number of classic and typical electric piano sounds. Damper This parameter sets the level of the damper noise caused by the damping felt hitting the vibrating tine.
Stretch and Warmth The EVP88 is tuned to an equal-tempered scale. As a deviation from this standard tuning, you can stretch the tuning in the bass and treble ranges, much like acoustic pianos (especially upright pianos). You can also modulate the tuning of each note randomly. The main tuning parameter is Tune. Note: The tones of upright pianos, and—due to their longer strings, less so—grand pianos have inharmonicities in their harmonic structure.
Effects Equalizer Treble This is a conventional filter for the high frequency range. Depending on the model selected, shelving or peak type filters are utilized, with optimized frequency ranges for each model pre-selected. Bass This is a conventional filter for the low frequency range. Depending on the model selected, shelving or peak type filters are utilized, with optimized frequency ranges for each model pre-selected.
Gain The Gain control determines the amount of harmonic distortion. Phaser Phaser pedals used by electric guitarists are “classic” effect tools for electric pianos as well—especially in the electric jazz, jazz-rock and pop styles of the seventies. Classical four-stage phasing effects are based on phase shifting using modulated all-pass filters. Mixing the phase-delayed signal with the original signal results in characteristic notches in the frequency response curve, also known as the comb-filter effect.
Tremolo A periodic modulation of the amplitude (level) of the sound is known as tremolo. The modulation is controlled via an LFO. The Fender Rhodes suitcase piano features a stereo tremolo and many other electric pianos have a simple, but quite obtrusive, mono tremolo that can introduce a strange kind of polyrhythmic feel to performances. Note: The original Wurlitzer piano has a mono tremolo with a fixed modulation rate of 5.5 Hz. For an authentic Wurlitzer sound, choose 0°. For Rhodes sounds, select 180°.
Additional Parameters The EVP88 features a number of additional parameters that are accessible via the 001/ 100 button at the top of the EVP88 Plug-in window. The Volume slider sets the overall output level of the EVP88 (Range: −20 to +20 dB). The Bend Range Down/Up sliders determine the pitch bend range in semitone steps. The Chorus Rate slider sets the speed of the Chorus effect, in Hz.
The Rhodes piano was also made available as a suitcase piano (with pre-amp and twochannel combo amplifier) and as a stage piano, without amplifier. Both of these 73-key “portable” versions have a vinyl-covered wooden frame and a plastic top. In 1973, an 88 key model was introduced. Smaller “Celeste” and bass versions were less popular. The Mk II (1978) had a flat top instead of a rounded one. This allowed keyboardists to place extra keyboards on top of the Rhodes.
Rhodes Models: • Suitcase MkI • Suitcase V2 • Bright Suitcase • Stage Piano MkI • Stage Piano MkII • Bright Stage MkII • Hard Stage MkII • MarkIV • Metal Piano • Attack Piano The Metal Piano and Attack Piano models feature sound qualities that can be “aimed at” with the original Rhodes instruments, but not to the extent of these models. They do not sound realistic, but they are included as sound “ideals” that the Rhodes technicians might have had in mind when preparing their keyboards.
The model Funk Piano does not sound realistic in the bass. We’ve added this special synthetic sound of the piano engine as a bonus. Hohner Electra Piano Not to be confused with the all-electronic RMI Electrapiano, the extremely rare Hohner Electra Piano offers striking hammers like those of the Rhodes, but a stiffer keyboard action. It was designed to resemble the look of a conventional acoustic upright piano.
MIDI Controller List E-Piano Stretched Tuning Equalizer Overdrive Phaser Tremolo Chorus Chapter 28 EVP88 Volume 11 Model 64 Voices 72 Tune 73 Decay 65 Release 66 Bell 67 Damper 68 Intensity 71 Lower 69 Upper 70 Warmth 74 Treble 75 Bass 76 Gain 77 Tone 78 Rate 79 Color 80 Stereophase 81 Speed 82 Intensity 83 Stereophase 84 Intensity 85 517
29 EXS24 mkII 29 This chapter introduces Logic’s EXS24 mkII sampler. The EXS24 mkII offers all of the facilities that you would expect to find in a hardware sampler, without the cost and bulk of this type of device. As a purely software-based instrument, the EXS24 is perfectly integrated into Logic, and makes use of your computer’s RAM and hard disks. This integration within the computer environment offers instant access to all audio data and Sampler Instruments used in a Logic song file.
The EXS24 is compatible with the EXS24, AKAI S1000 and S3000, SampleCell, ReCycle, WAV, AIF(F), Gigasampler, and SoundFont2 sample formats, as well as the Vienna Library, allowing access to large and comprehensive sampler libraries. The EXS24 offers numerous sample processing and synthesis options, enabling you to tailor sounds to meet your needs. Last, but not least: as a highly optimized Logic instrument, the EXS24 offers great performance, even on slower machines.
Change the sound by twisting the knobs, pressing switches and moving sliders—and don’t worry—you can’t destroy the original Sample Instrument. Creating and Editing Instruments in the Instrument Editor Instruments are created and edited in the Instrument Editor. It is also used to organize samples and to convert foreign sample formats (AKAI S1000/3000 and so on).
To create a new Instrument and a Zone 1 Select Instrument > New from the Editor window’s menu. A new instrument is created. Note: In order to hear your edits, please ensure that the correct Instrument is loaded into the EXS24 instance assigned to the currently selected track, and is selected in the editor. 2 Go to Zone > New Zone to create a new zone. A small window will appear to the left of the editor window. 3 Click on the empty field alongside the Audio File label.
To move a Zone: 1 Move the mouse cursor (it will change to a two-headed arrow) over an existing zone bar (this is one of the bars displayed directly under the keys). 2 Click-hold and drag the zone to the desired position below the onscreen keyboard. To change the start/end note of a Zone: 1 Move the mouse cursor to the beginning or end of a zone bar. The cursor will change to a left (start) or right (end) bracket, surrounded by arrowheads.
Editing Samples You may have noticed the small E buttons next to the start, end and loop point parameters. Clicking on these will launch the selected sample in Logic’s Sample Editor, allowing you to edit the sample borders graphically, and use all of the Sample Editor’s functionality. When loop is activated, you can also edit the loop points graphically: the LS marker indicates the loop start point and LE, the loop end point.
Given that up to 64 EXS24 instruments (dependent on your version of Logic) can be used simultaneously, opening several instances of the EXS24 provides the advantage of a dedicated channel strip for each and every sound you use. This allows full control over the sound (via EXS and effects parameters) during composition and mixdown. To assign a Group to a Zone 1 Select Group > New Group in the editor’s menu to create a new Group. A Group window will appear on the right-hand side of the editor.
• Each Group offers separate ADSR parameters for offsetting the ADSR volume envelope settings made in the Plug-in window: The Attack, Decay, and Release time parameters can be adjusted by ±9999 ms, the Sustain level by ±50%. • Similarly, the Cutoff and Resonance settings of the Plug-in window can be offset by ±50% for each Group.
Note: You can store your Sampler Instruments in any folder on any of your computer’s hard drives. To do so, you must create an alias pointing to this folder within the Sampler Instruments folder located in the Logic program folder. Please refer to “File Organization” on page 526. You can manually load Sampler Instruments from other locations into the EXS Instrument Editor at any time. Such Instruments also appear in the EXS24’s Sampler Instrument load pull-down menu.
To organize your Sampler Instruments into a preferred hierarchy: 1 Simply create a folder—“Basses” for example—within the Sampler Instruments folder, with your operating system’s file management utilities. 2 Drag and drop the desired EXS24 Sampler Instruments into this newly created folder. Their menu structure will be reflected when clicking on the EXS24 Sampler Instruments pull-down menu.
It works as follows: When opening a Logic Project, the EXS24 initially looks for a subfolder named “Sampler Instruments” in the folder that contains the song file. If such a sub-folder exists, all Sampler Instruments found in this folder are added to the Sampler Instrument pull-down menu in the EXS24 GUI. This new entry in the Sampler Instrument pull-down menu will appear as a sub-menu item that matches the song file name.
This will change the folder hierarchy as follows: • The Project folder contains the song file and the Sampler Instruments folder. • The Sampler Instruments folder contains all Sampler Instruments that are used in this song exclusively—vocals, for example. • A separate folder containing the audio files associated with the respective Sampler Instrument for each Sampler Instrument used.
The Clear Find option in the Sampler Instrument pull-down menu will display the full menu but does not clear the actual search term typed into the search dialog. To return to the limited menu, simply select Enable Find. The selection of Enable/Clear Find allows you to toggle between the two without re-typing the search term. If you wish to use a different character string, select the Find option a second time and type in the desired search term.
2 When the Sampler Instrument is loaded, ensure that the appropriate CD-ROM is in the computer’s CD-ROM drive. If the appropriate CD-ROM (the one that contains the desired Sampler Instrument and its associated audio files) is in the drive, the EXS24 will automatically search for the associated samples on all local media. It will locate the CDROM and will load the Sampler Instrument. 3 If the CD-ROM is not present, you will be required to insert the appropriate disc and reload the Sampler Instrument.
Note: You can store your imported Sampler Instruments in any folder on any of your computer’s hard drives. To do so, you must create an alias pointing to this folder within the Sampler Instruments folder located in the main Logic program folder. Care should be taken when importing samples to ensure that when a song is loaded, the associated Sampler Instruments will be found. Sampler Instruments are only searched for in the Sampler Instruments folder (or an alias to it).
Importing Giga Files The importation of Giga format files is as per that of SoundFont2 files. Simply copy or move your Gigasampler files into the Sampler Instruments folder. Select the file name in the EXS24 Sampler Instrument load flip-menu and the file will automatically be converted. An EXS Instrument will be created in the Sampler Instruments folder which contains the original Giga file.
Converting AKAI Files This section discusses the AKAI import procedure. The EXS24 can import samples saved in the AKAI S1000 and S3000 sample formats. The AKAI Convert function can be used to import: • an entire AKAI format CD ROM • an AKAI Partition • an AKAI Volume • an AKAI Program • an Individual Audio File (sample) These options have been provided to give you the most flexible and efficient method of dealing with your sample library.
Note: Reading of a CD-ROM may take some time, dependent on the amount of sample data and file structure of the disc. In addition, the speed of the CD-ROM mechanism, bus speed, memory, and other factors can affect performance. 3 To view the contents of the Partitions, click once on the appropriate entry with the mouse button. This will display the Volume information contained within the Partition.
When a Program is imported, these programs appear as Program.EXS in the Sampler Instruments folder. Sampler Instrument management works with AKAI samples imported from CD ROM, in the same fashion as with other sample formats. Given the different file structures used by many AKAI format discs, however, you should take care to follow these guidelines. • Create a shortcut to any folder on your hard disk/s which contains your AKAI sample library (or where you wish to store it).
Default instrument output volume (head room) This parameter is extremely useful for many AKAI CD-ROMs. Please select this option before converting a CD-ROM. • For drum CDs, select a headroom value of −− 3 up to zero dB. • For piano/string/pad CDs, a headroom value of −− 9 dB is recommended, or the sound may/will clip with polyphonic use of these types of instruments. • In cases where you’re not sure of which headroom value to select, choose − 6 dB (average). Merge programs (same MIDI cha. and prog.
Prelisten Function The AKAI Import window features a Prelisten button, which is found below the Audio Files column. This facility allows you to individually audition AKAI audio files before deciding whether or not to import them. To prelisten an audio file: 1 Select an individual file (sample) within the Audio Files column: 2 Press Prelisten. This will start playback of the selected audio file and the Prelisten button will update, with the word “Stop” appearing on the face of the button.
Voices This parameter determines the number of voices (polyphony) that the EXS24 is supposed to play. The used field indicates the number of voices that are actually used. If both fields tend to show the same value most of the time (probably causing a noticeable number of samples to drop out), you should set a higher voices value. Unison Mode This mode plays multiple EXS24 voices when each key is triggered: • In Poly mode, two voices per note.
Options Button Clicking the Options button launches a menu that offers the following options: • Recall default EXS24 settings recalls a neutral setting for all parameters in the Plug-in window. • Recall settings from instrument command manually recalls the original parameter settings of the loaded Sampler Instrument. This parameter is extremely useful if you’ve been over zealous with your tweaking.
• Virtual Memory opens a settings window for the EXS virtual memory functions. Virtual memory allows samples of almost unlimited length to be played back using streams that are fed directly from the hard disk. Switch off this option if you have enough RAM for your current work. The Active checkbox switches virtual memory on or off. In the General Settings, you can set the Disk Drive Speed and the Hard Disk Recording Activity.
Amount This is the range of velocity (or other modulation source) values in which the crossfade takes place. The Select Range setting of all Zones will be expanded by this value, with the crossfade taking place in the expanded area. When the Amount parameter is set to 0, the EXS24 will switch between sample Zones in exactly the same fashion as earlier versions (Velocity Switching).
Pitch Parameters Tune Offsets the pitch of the sample(s) in semitones by up to ±2 octaves. The middle position of the knob (which can be set by clicking the small 0 button) leaves the pitch unaltered. Transpose This parameter allows you to transpose the EXS24. In contrast to the Coarse Tune parameter, Transpose not only affects the pitch, but also moves the Zones in accordance with the Transpose setting. Random This rotary knob controls the amount of random detuning which will apply to every played note.
Remote The Remote parameter allows you to easily pitch complete EXS24 Instruments in realtime. To do so, set the Remote parameter to the key of your MIDI Keyboard that you would like to use as the original pitch. All keys in a range of ±1 octave around this key will now pitch the entire Instrument. This two octave range is similar to the Pitch Bend function, but is quantized to semitones.
When both halves of the pitcher slider are set below or above the centered position, either a low or high velocity will slide up/down to the original pitch. Dependent on the position of the upper/lower halves of the slider in relation to the center position, the time required for the slide up/down to the original note pitch can be adjusted independently for both soft/hard velocities. Filter Parameters Filter On/Off Switch This button switches the filter section on or off.
Bandpass (BP) The Bandpass Filter is a 2 pole (12 dB/Oct.) design. A Bandpass filter only allows the frequency bands directly surrounding the cutoff frequency to pass. Frequencies which fall outside these boundaries will be cut. Drive This knob allows the filter input to be overdriven. Turning Drive up leads to a more dense and saturated signal, with additional harmonics being introduced/becoming audible. Cutoff The cutoff frequency of the lowpass filter.
Volume and Pan Parameters Level via Vel Controls the volume of the sound. The Level parameter can be modulated by velocity: the upper half of the slider determines the volume for maximum velocity, the lower half for minimum velocity. By clicking and dragging in the area between the two slider segments, you can move both simultaneously. Volume The main volume parameter for the EXS24.
LFO Parameters LFO 1 EG This knob allows LFO 1 to be faded out (Decay area) or faded in (Delay area). In the centered position (which can be set by clicking on the small 0 button), the LFO intensity is constant. LFO 1 Rate This is the frequency of LFO 1. It can be set in note values (left area), or in Hertz (right area). In the centered position (which can be set by clicking on the small 0 button), the LFO is halted and generates a constant modulation value at full level (DC = Direct Current).
LFO 2 Rate The frequency of LFO 2. It can be set in note values (left area), or in Hertz (right area). In the centered position (which can be set by clicking on the small 0 button), the LFO is halted, and generates a constant modulation value with full level (DC = Direct Current). Again, don’t overlook this feature if you want to control an LFO-modulated parameter directly via the Modulation Matrix (see following section). LFO 3 Rate There is a third LFO available which always uses a triangular waveform.
You have the option of inserting another modulation source in the middle slot labeled via. In this scenario, the green triangular fader will divide, allowing you to set a range for modulation depth. The size of the modulation range depends on the possible values allowed by the via modulation source. In our example, the key number of the MIDI keyboard (Key) determines how strongly channel pressure controls the Speed of LFO1.
Second Order Modulations The EXS24 also allows the use of second order modulation destinations (such as envelope times, LFO speeds and so on)—functionally outperforming many analog synthesizers. To explain: • The same source can be used as often as desired to control different destinations. • The same destination can be controlled by different sources. The different input values are accumulated.
Modulation Sources Modulation Destinations Relative Volume (auto adjust) Key LFO1 Dcy./Dly Velocity LFO1 Speed Control Nr. 1 LFO2 Speed … LFO3 Speed Control Nr. 120 Env1 Attack Env1 Decay Env1 Release Time Env2 Attack (Amp) Env2 Decay (Amp) Env2 Release (Amp) Hold Note: Controllers 7 and 10 are marked as (not available). Logic uses these controllers for volume and pan automation of the audio object. Controller 11 is marked as (Expression).
Aux Channels for Instrument Plug-ins with Multiple Outputs By default, all outputs of a “Multi Channel” Instrument plug-in are summed and routed to the stereo output of the respective Audio Instrument channel at first. Signals of individual outputs (starting with output 3) are automatically subtracted from the stereo output sum after they have been assigned to Aux channel inputs. Outputs 1 and 2 are always assigned to the respective Audio Instrument. This assignment is fixed and can’t be changed.
Save As… This command also saves the currently loaded/edited Sampler Instrument. When Save As… is used, you will be prompted to give the file a name. Use this command when you want to save a copy of an edited Sampler Instrument, rather than overwriting the original version. Rename This command allows you to rename the loaded Sampler Instrument. The renamed version replaces the previous version on the hard disk. Delete This deletes the currently opened Sampler Instrument.
Extract MIDI region and make new Instrument Use of this menu option will launch a file selection dialog, allowing the selection of a ReCycle file. • Following selection, a new EXS24 Instrument with a name which matches that of the ReCycle loop will be created. Should an EXS24 Instrument of that name already exist, a # sign and a number will be appended, ensuring that the filename is unique within the Sampler Instruments folder. • You will be prompted to enter a velocity factor (see page 534).
Extract Region(s) from ReCycle Instrument This option generates MIDI Regions from EXS24 Instruments which were originally converted from ReCycle file(s). The MIDI Regions are created on the currently selected track, at the current song position (rounded to bars). A single MIDI Regions is generated for each imported ReCycle loop in the currently open Instrument. If no imported ReCycle loop exists in the currently open Instrument, this menu option is disabled.
Cut, Copy, Paste The standard commands for cutting, copying, and pasting values. In addition to values you may also cut, copy, and paste selected Zones and Groups. Note: When multiple Zones and Groups are cut, copied, or pasted simultaneously, the Group assignments of the Zones are retained. Clear Deletes the currently selected Zone or Group. Clear can be undone with the Undo command. Select All Selects all Zones and Groups of the loaded Sampler Instrument.
The Velocity Curve parameter determines the EXS24’s responsiveness to velocity values received from your MIDI keyboard. Negative values increase the response to soft key strikes, and positive values decrease it. The Search Samples On parameter determines the location that instruments samples should be searched in. You may either choose the drives normally used by the operating system or external SCSI, FireWire or USB drives, accessible directly or over a network.
Note: There may be cases where a sound designer has used multiple numbers in a filename, which is common with loops, with one value being used to indicate tempo— “loop60-100.wav”, for example. In this situation, it isn’t clear which, if either of the numbers, indicates a root key or something else: 60 or 100 could indicate the file number in a collection, tempo, root key, and so on. You can set a value of “8” to read the root key at position (letter/character) eight of the filename—namely the 100 (E6).
Zone Menu New Zone Creates a new Zone in the currently loaded Sampler Instrument. Load Multiple Samples Allows several samples to be loaded in one operation. The Instrument Editor reads the key note from samples, and places the samples into new Zones. These Zones are automatically created. The key note is located in the middle of the Zone, with Zone borders being determined by the Zone borders of neighboring Zones. Zones with identical key notes will be layered.
Sort by… These commands determine the order in which the Zone windows are displayed: Sort by None: The Zones are shown in the order in which they were created. Sort by Name: The Zones are sorted alphabetically. Sort by root key (high to low): The Zones are sorted according to their Root Key settings; the higher key notes are displayed at the top of the list. Sort by root key (low to high): The Zones are sorted according to their Root Key settings; the lower key notes are displayed at the top of the list.
Sort by Velocity (high to low): The Groups are sorted according to their Velocity Range settings; the higher velocity ranges are displayed at the top of the list. Sort by Velocity (low to high): The Groups are sorted according to their Velocity Range settings; the lower velocity ranges are displayed at the top of the list. Delete unused Groups This command deletes all Groups that do not have a Zone assignment. This deletion can be undone with the Undo command.
Zone Parameters Zone Name New Zones are numbered consecutively. Double-clicking on a Zone’s number allows you to enter a name instead. Audio File A file selection dialog can be opened by clicking on the gray field next to the Audio File label. This will allow you to select and load a sample into a Zone. When loaded, the sample’s name is displayed in the gray field.
One Shot/Reverse Activating One Shot causes the Zone to ignore the length of notes used to trigger the sample—the sample is always played to the end. This option is useful for drum samples, where you often don’t want the MIDI note length to affect sample playback. Reverse plays the sample from its end to its beginning. This option works nondestructively, leaving the audio data in the sample unchanged. Volume/Pan/Scale Volume adjusts the volume of the Zone. Pan adjusts the pan position of the Zone.
Auto Crossfade In a crossfaded loop, there is no hard “cut” between the loop end and loop start points. Rather, the loop end and start points are crossfaded for a smooth transition. This is especially convenient with samples that are hard to loop, and would normally produce clicks at the transition point—the “join” in the loop. The Auto Crossfade field allows a pre-determined value (in milliseconds) to be used as a default when the option is enabled.
Select Range—The two Velocity Range parameters allow you to set up a velocity range for the Group. The settings made here override the settings in the Zones, if necessary: When a Zone’s velocity range is larger than that allowed by the Group setting, the Zone’s velocity range is limited by the Group setting. Trigger on Key Down (default)—Zones pointing to this group are triggered on key down. Key Release—Zones pointing to this group are triggered on key release.
EXS24 Key Commands A number of key commands are available for the EXS24 which accelerate editing in Logic, and provide additional functionality. They are found in the Key Commands window. These key commands have no default keyboard assignments, so you will need to create them, should you wish to take advantage of these shortcuts and facilities. Please consult your Logic reference manual for information on accessing the Key Commands window and on the assignment of keyboard shortcuts to functions.
View: All/Toggle Mode Toggles between viewing all parameters in Zones and Groups and a limited view which displays the Audio File name in Zones and the Volume/Pan parameters in Groups. View: Next Zone Parameter This key command is designed to aid in the adjustment of the same parameters in each Zone. It limits the Zone display(s) to individual parameters (or rows of parameters) and steps through them from top to bottom.
Move audiofiles of all USED and ACTIVE instruments of current song… Moves the audio files of all (active) Sampler Instruments used by the current song to the target directory of your choice. Folders for the audio files associated with these Sampler Instruments are created in the target location. A Brief History of Sampling The idea of an instrument that could change its sound at any time, and that could imitate any other instrument, dates back centuries.
MIDI Controller List Common Pitch Filter Chapter 29 EXS24 mkII Mono Mode 71 Voices 72 Start Fixed 73 Start via Vel 74 Time via Key 11 Attack Curve 112 • Pitch Bend (up) 9 • Pitch Bend (down) 70 • Transpose 5 Coarse Tune 76 • Fine Tune 77 Glide 78 Pitcher 79 Pitcher via Vel 80 Modulation LFO 81 Mod. Depth Fixed 82 Mod.
Volume LFOs 572 Chapter 29 EXS24 mkII • Output Volume 67 • Key Scale +/− 68 Level Fixed 96 Level via Vel 97 Tremolo/Pan LFO 98 Pan Modulation 99 Tremolo 100 Amp Attack 113 Amp Att. via Vel 114 Amp Decay 115 Amp Sustain 116 Amp Release 117 LFO 1 Dec.
30 GarageBand Instruments 30 GarageBand Instruments are automatically installed with Logic. You can insert them as per other software instruments. GarageBand Instruments are accessible from the Stereo > Logic > GarageBand Instruments sub-menu. About GarageBand Instruments GarageBand Instruments are software instrument plug-ins that are used in Apple’s GarageBand application. Their inclusion makes the importing of GarageBand files into Logic a trouble-free experience.
The interface of GarageBand Instruments consists of a simple silver panel that contains a number of parameter sliders and associated value fields. As an example, here is the Digital Stepper instrument: Many of these parameters are macro parameters, which address specific, useful parameters in the EXS24, ES1 (or other equivalent Logic instrument) instance simultanously.
31 External Instrument 31 The External Instrument plug-in provides a simplified way of handling an external MIDI sound source if you’re feeding its audio output directly into your audio interface. This facility allows you to use one Arrange track for both MIDI recording and audio mixing with effect plug-ins. The External Instrument plug-in can be inserted in Audio Instrument channels (Mono/ Stereo > Logic > External) in place of a software instrument.
Glossayr Glossary AAF Abbreviation for Advanced Authoring Format. This file format, typically used for data exchange with Digidesign ProTools software, can be imported and exported by Logic. It allows multiple audio tracks to be imported, with reference to tracks and Region position, volume automation included. AD converter or ADC Short for analog/digital converter; a device that converts an analog signal to a digital signal.
analog signal A description of data that consists of a constantly varying voltage level, that represents audio information. Analog signals must be digitized, or captured, for use in Logic. Compare with digital. Arrange window The heart of Logic. The primary working window of the program where Audio and MIDI Regions are edited and moved to create a song arrangement. attack Start phase of a sonic event. Also part of an envelope (see envelope). attenuate To lower an audio signal’s level.
audio track A track in Logic’s Arrange window that is used for playback, recording and editing of Audio Regions. Audio Track Object Audio Object in the Environment’s Audio layer. Used to playback audio tracks in Logic’s Arrange window. All data on the audio track is routed to the Audio Object, that was assigned in the Arrange window’s Track List menu. Audio Units (AU) Audio Units is the standard format for real-time plug-ins running on Mac OS X. It can be used for audio effects and software instruments.
Beat Mapping track Component of the Global tracks that helps to make a rhythmically meaningful display of recordings that do not correspond to a strict tempo throughout. It does this by redefining the bar positions of existing musical events, without changing their absolute time position, thereby preserving the audible result with its original timing. beats per minute See bpm. bit depth The number of bits a digital recording or digital device uses.
Catch button The button in the Transport bar featuring the running man icon. Activate this button (blue) to turn on automatic horizontal scrolling during playback. This ensures that the current playback position is always visible. Catch function A window function that makes the currently displayed song section reflect the current song position. Also see Catch button. CD Audio Short for Compact Disc—Audio; current standard for stereo music CDs: 44.1 kHz sampling rate and 16 bit depth.
controller MIDI data type. As examples; sliders, pedals or standard parameters like volume and panning. The type of command is encoded in the first data byte, the value in the second data byte. Controls view All Logic plug-ins (and Audio Units) offer a non-graphical alternative to the Editor views of effect and instrument parameters. The Controls view is accessed via the Controls pull-down menu at the top of each plug-in window.
destructive Destructive audio processing means that the actual data of an audio file is changed, as opposed to just editing peripheral or playback parameters. dialog A window containing a query or message. It must be cancelled or replied to before it will disappear and allow you to continue. digital A description of data that is stored or transmitted as a sequence of ones and zeros. Most commonly, refers to binary data represented using electronic or electromagnetic signals.
Editor view Almost all Logic plug-ins (and Audio Units) offer a graphical view of effect and instrument parameters. The Editor view is used by default, but can be accessed via the Editor pull-down menu at the top of each plug-in window, should the Controls view be visible. effect A type of software algorithm that lets you alter the sound of a track in a variety of ways. Logic includes a set of EQ, dynamics, time-based, modulation and distortion effects in Logic’s native and Audio Unit plug-in formats.
filter effect Filters are effects you can apply to Audio or MIDI Regions (when streamed or recorded as audio). They are designed to reduce a signal’s energy at a specific frequency. A true filter always acts as a subtractive device, and doesn’t add anything to the signal. The names of the individual filters illustrate their function. As an example: A Low Pass filter allows frequencies that are lower than the cutoff frequency to pass.
hierarchical menu Structured menus where choosing an individual entry opens a submenu. high cut filter A high cut filter is essentially a lowpass filter that offers no slope or resonance controls. highpass filter A highpass filter allows frequencies above the cutoff frequency to pass. A highpass filter that offers no slope or resonance controls is generally knows as low cut filter. icon Small graphic symbol. In Logic, an icon may be assigned to each track.
Link mode Link mode is activated by clicking the Link button. It determines the relationships between windows. An editing window in Link mode shows the same contents as the top window. Link button Button featuring the chain link icon in the top left corner of most Logic windows. It controls the linking between different windows. local menu Menu in a window that only contains functions that are relevant to that particular window.
marker Markers serve three purposes in Logic: They mark time-positions in the Arrange window. They hold text notes and they delimit song settings. Markers can be placed in the Marker track, or they can be placed in the Bar Ruler. Markers are generally used for indicating and navigating to different song sections. main menu bar The bar at the top of the computer screen, offering global functions such as opening, saving, exporting or importing songs. It does not offer access to local functions.
modulation matrix The EXS 24 and other Logic instruments contain a grid that allows you to modulate a number of target parameters with a number of modulators. This grid is referred to as the modulation matrix. modulation path A modulation path determines which target parameter will be modulated by a specific modulator (modulation source). modulation wheel A MIDI controller found on most MIDI keyboards. mono Short for monophonic sound reproduction.
normalize This function applies the current Parameter box settings to the selected MIDI events (by altering the actual events themselves), and clears the Parameter settings. When it comes to audio, a different “Normalize” function raises the volume of a recorded audio file to the maximum digital level without altering the dynamic content. notch filter This filter type cuts the frequency band directly surrounding the cutoff frequency and allows all other frequencies to pass.
peak 1) The highest level in an audio signal 2) portions of a digital audio signal that exceed 0 dB, resulting in clipping. You can use Logic’s level meter facilities to locate peaks and remove or avoid clipping. The Search Peak command in the Sample Editor’s Functions menu searches for the sample bit with the greatest amplitude value in the currently selected Audio Region. pink noise A harmonic noise type that contains more energy in the lower frequency range.
PWM Pulse Width Modulation. Synthesizers often feature this facility, where a square waveform is deformed by adjusting it’s pulse width. A square waveform usually sounds hollow, and woody, whereas a pulse width modulated square wave sounds more reedy and nasal. Q factor A term generally associated with equalizers. The Q factor is the “quality” factor of the equalization, and is used to select a narrower or broader frequency range within the overall sonic spectrum of the incoming signal.
reverb Reverb(eration) is the sound of a space. More specifically, the reflections of soundwaves within a space. As an example, a handclap in a cathedral will reverberate for a long time as sound waves bounce off the stone surfaces within a very large space. A handclap in a broom closet will hardly reverberate at all. This is because the time it takes for the soundwaves to reach the walls and bounce back to your ears is very short, so the “reverb”’ effect will probably not even be heard.
semitone Smallest interval between two pitches in the standard diatonic scale, equal to a half tone. Correspondingly a semitone is also called half step or half tone. send Abbreviation for auxiliary sends. An output on an audio device used for routing a controlled amount of the signal to another device. Sends are for example often used to send several signals to the same effect, which is rather advisable for computationallyintensive effects such as reverb.
song Main Logic file, containing all MIDI events and parameter settings (including mixer automation data) plus information about the audio files to be played. Song Settings The Song Settings, accessible from the File menu, are a collection of program settings that are specific to the current song. These are different to the global preferences that affect all Logic songs (see preference). stereo Short for stereophonic sound reproduction of two different audio channels. Compare with mono.
Track Mixer Adaptive Mixer which automatically configures itself to show every audio and MIDI track, in the order that they appear in the Arrange window or in an open Folder. If you move the controls on the Track Mixer while recording, automation data is stored in the relevant tracks as MIDI controller information. transient Position in an audio recording where the signal becomes a lot louder—over a short time span (a signal “spike”, in other words).
word length See bit depth. XY stereo recordings Two cardioid microphones aligned so that they are directed to the left and right of the sound source. Also see MS stereo recordings. zero crossing A point in an audio file where the waveform crosses the zero amplitude axis. If you cut an audio file at a zero crossing there will be no click at the cut point. zoom An action that enlarges (zooms in on) or shrinks (zooms out from) the display in a Logic window.
A AAF 577 Adaptive Limiter 50 Gain 51 Input Scale 51 margin display 51 Out Ceiling 51 ADC 577 AD converter 577 additive synthesis 232, 481 aftertouch 577 channel 577 polyphonic 577 AIFF 577 AKAI 577 alias 577 aliasing 577 allpass filter 577 amplifier 577 amplitude 577 analog 578 analog synthesizer 195 Arrange window 578 attack 578 attenuating 578 AU Audio Configuration window 578 Audio Instrument 578 Audio Instrument Object 13, 23, 24 Audio Mixer 578 Audio Object 13, 578 Audio Track Object 579 Aux Object 57
Mode 63 Resolution 63 bit depth 580 bit rate. See bit depth bit resolution.
Input 161 MS stereo 162 Distortion 62 Drive 62 Output 62 Tone 62 Distortion II 66 PreGain 66 DJ EQ 38 driver 583 DSP Ducking 18 Dudley ,Homer 191 dynamic range 583 E editor 583 Editor view 17, 584 effect 23 bus 23 insert 23 mono 15 plug-in 13, 23 stereo 15 EFM 1 aftertouch 206 Carrier 204 Fine 204 Fixed Carrier option 204 FM 202 FM Depth 203 Glide 202 Harmonic 204 LFO 203 Main Level 206 Modulation Env 203 modulation wheel 206 Modulator 204 Modulator Pitch 203 Modulator Wave 205 pitch bend 206 Randomize 202
Key 217 Level Via Vel 218 LFO Amp 219 LFO Waveform 218 Mix 216 Mod Envelope 220 Out Level 220 Rate 219 Resonance 217 router 219 Sub 216 Tune 220 Voices 221 Wave 216 ES2 223 Amp 252 amplifier 252 Analog 226 Bend Range 227 Blend.
Detune 255 FltBlend 258 Lfo1Asym 259 Lfo1Curve 259 LPF FM 258 Osc1Levl 257 Osc1Wave 256 Osc1WaveB 257 Osc2Levl 257 Osc2Wave 256 Osc2WaveB 257 Osc3Levl 257 Osc3Wave 256 Osc3WaveB 257 OscLScle 257 OscWaveB 256 OscWaves 255 Pan 259 Pitch 1 255 Pitch 123 254 Pitch 2 255 Pitch 3 255 Reso 1 258 Reso 2 258 SineLevl 257 Mono 227 multi trigger 228 muting oscillators 230 noise 240 Oscillators 226 OscLevelX 241 OscLevelY 241 Osc Start 229 Parallel 242 Phaser 280 Poly 227 polyphonic distortion 245 processing power, han
Env Mode 276 Fix Timing 279 Loop Count 279 Loop Mode 277 Loop Point 274 Loop Rate 278 Loop Smooth 278 resetting a point 275 setting point 275 Solo Point 276 Sustain Point 273 switching Vector Envelope off 275 Time Scaling 279 Vector Int 273 Vector Mode 272 Vector X Target 272 Vector Y Target 272 via 253 Via invert (inv) 254 via source 263 Bender 264 ENV1 263 ENV2 263 ENV3 263 Kybd 264 LFO1 263 LFO2 263 ModWhl 264 Pad-X 264 Pad-Y 264 RndNO1 265 RndNO2 265 SideCh 265 Touch 264 Velo 264 Whl+To 264 Voices 228 W
Drawbar 460 Drawbar Leak 467 Drive 474 Effect Bypass 471 Effect Chain 471 EQ High 471 EQ Level 471 EQ Low 471 EQ Mid 471 Expression 461 Filter Age 468 Horn Deflector 476 Keyboard Mode 458 keyboard range of upper and lower manual 457 keyboard split 458 Leakage 467 Leslie 474, 484 Lower Stretch 469 Lower Volume 461 LP Split 458 Max Wheels 466 Mic Angle 476 Mic Distance 476 MIDI CC 465 MIDI controller assignment 477 MIDI Mode 458 MIDI Preset Switching 464 MIDI setup 456 MIDI to Presetkey 464 Mode (Morph) 465 M
Medium 489 MIDI 502 Mode (Modulation) 501 Model 491 Basic 491 Class D6 491 Domin 492 Dulcimer 492 Guru Funk 492 Harpsi 492 Ltl India 492 Mello D6 491 Old D6 491 Picked 492 Pluck 492 Sharp D6 491 StrBells 492 Wood 492 Modulation 501 Phaser 501 Intensity 501 Rate 501 Pickup 489 Pickup Mode 498 Pickup Position 497 Pitch Fall 496 Pressure 488 Random 495 Range 500 Release 495 Shape 494 Soft 489 Stereo Spread 490 Stiffness 496 Stretch 488 String 495 Tension Mod 496 Tone 499 Treble 489 Tune 487 Velocity 495 Velo C
Level 178 oscillators 177 Pitch LFO 185 Poly 177 portamento 180 Rate 186 Ratio c 179 Ratio f 179 Release (Sidechain Analysis In) 181 Release (synthesizer) 181 Resonance (Formant Filter) 184 Resonance (synthesizer filter) 180 Semi 179 Sensitivity 186 Shift LFO 185 Side Chain 175 Sidechain Analysis In 181 Signal 187 single trigger 177 Stereo Width 188 Syn 188 Tuning 180 U/V Detection 186 Unison 177 Unvoiced/Voiced detector 169 Voc 188 Voices 176 Wave 1 178 Wave 2 178 Waveform 178 EVOC 20 TO 79 Analysis In 81
Color 1 154 Color 2 154 Frequency 154 graphic 154 Harmonics 154 Input 154 Expander 42 Auto Gain 42 Ratio 42 Threshold 42 EXS24 mkII 519 AKAI Convert entire CD 536 Convert function 535, 555 Convert window 537 file organization 536 Partition 535 Prelisten 539 Program 536 Volume 536 Amount 543 Amp (Env 2) 548 b/p 551 BP 547 chain symbol (filter) 547 Clear Find 531 compatibility EXS24 mkI 552 Cutoff 547 Dest 550 Drive 547 Edit button 540 Enable Find 531 EXS24 mkI Modulation Path 552 Fat 546 file organization 52
creating 522 creating Sampler Instrument 554 deleting Sampler Instrument 555 loading Sampler Instrument 520, 540 managing Sampler Instruments 527 moving audio files of Sampler Instrument 555 opening Sampler Instrument 554 renaming Sampler Instrument 555 saving Sampler Instrument 554 saving Sampler Instrument song-related 528 searching Sampler Instrument 530 selecting Sampler Instrument 540 Sample Select 543 Second Order modulation 552 Setting 527 SoundFont 2 532 Src 550 Transpose 544 Tune 544 Type 543 Uniso
Conderser button 60 Dynamic button 60 EQ menu 59 FX section 61 Gain 59 Link button 59 Master 60 Mid control 59 Off-Center button 60 Output 61 Presence parameter 59 Reverb section 61 Speaker menu 58 Treble control 59 H Hammond organ 482 Hermode Tuning 27, 585 highpass filter 249, 586 High Shelving EQ 38, 92 Hohner Electra piano 516 I I/O 160 Input 160 Input Volume 160 Output 160 Output Volume 160 impulse response 117 Input Object 586 insert 23 instrument plug-in 13, 23, 24 interface 586 intermodulation eff
Master Gain 55 multi-band graphic 53 Output 55 Peak/RMS 53 Reduction 54 Release 53 multitimbral 589 multi trigger 228, 361 Multi Trigger mode 589 mute 589 N node 589 No HMT option 27 Noise Gate 43 Attack 43 chattering effect 43 Hold 43 Hysteresis 43 Lookahead 44 Monitor 44 Reduction 43 Release 43 Side Chain 44 Threshold 43 O Object mono 15 stereo 15 Object Parameter box 590 No HMT option 27 oscillator 590 Output Object 590 Overdrive 62 Drive 62 Output 62 Tone 62 P Parallel Bandpass Vocoder 191 Parameter
Ensemble 105 Enveloper 67 EnVerb 115 ES1 215 ES2 223 ES E 213 ES M 207 ES P 209 EVD6 485 EVOC 20 FB 72 EVOC 20 PS 175 EVOC 20 TO 79 EVP88 505 Exciter 153 Expander 42 EXS24 mkII 519 External Instrument 575 Fat EQ 37 fine-tunig parameters 16 Gain 159 GarageBand Instrument 573 GoldVerb 111 Guitar Amp Pro 57 High Shelving EQ 38, 92 I/O 160 instrument 13, 23, 24 Levelmeter 166 Limiter 49 loading multiple plug-ins 20 Low Shelving EQ 38, 92 Match EQ 33 Modulation Delay 97 Multipressor 52 Noise Gate 43 numerical pa
OSC 102 Output section 103 Side Chain 102 Single 102 RMS 593 root note 593 Rotor Cabinet 106 routing 593 S Sample & Hold 267 Sample Delay 93 sampler (history) 570 sample rate 593 sampling 593 saturation 593 sawtooth wave 232 Scanner Vibrato 106 Rate Left 106 Rate Right 106 Stereo Phase 106 Sculpture 355 1, 2 and 3 button 368 Always mode 371 amplitude envelope 374 Attack 374 Bandpass 377 Bender Range Down 362 Linked 362 Bender Range Up 362 Body EQ 381 aktivieren 381 component modelling 356 String 357 Contro
velocity sensitivity 368 Peak 377 Pickup 373 setting position 373 Poly mode 361 Position 372 programming electric basses 426 Release 362, 363, 375 Resolution 365 Resolution Scale Low/High 365 Resonance 378 signal path 356 Single Trigger 361 Spread 381 Stereo Base 380 stereo delay 379 activating 379 Stiffness 364 Stiffness Scale Low/High 364 Strength 371 String 357 activating animation 372 Sustain 375 Sync (stereo delay) 380 Tension Mod 366 Tension Mod Scale Low/High 366 Timbre 371 Transpose 360 Tune 360 Var
Filter Envelope 131 Attack Time 132 Break Level 132 Curve Form Node 133 Decay Time 132 End Level 133 Init Level 131 Impulse Response 137 creating 137 digital spike 137 loading 120 randomly generating 120 starter pistol 137 sweep 137 impulse response 117 Input (Crossfeed) 123 inserting Space Designer 118 IR Sample 120 IR Start 124 Latency Compensation 125 Length 122 Low Shelving EQ 136 Freq 136 Gain 136 operating parameters 118 Pre-Delay 124 Preserve Length 122 red progress bar 119 reflectogram 117 reverbera
tempo 595 Test Oscillator 157 Frequency 157 Level 157 Sine Sweep 157 Time 158 Trigger 158 Waveform 157 time signature 595 timing 595 toggle 595 Touch Track 595 Track List 595 Track Mixer 596 transient 596 transposition 596 Tremolo 104 graphic display 104 Rate 104 Smoothing 104 Stereophase 104 Symmetry 104 triangular wave 199, 232 Tuner 158 U Ultrabeat 301 2 band EQ 321 editing graphically 322 808 snare 347 assignment section 302, 303 control element 308 fine-tuning 308 setting to default value 308 distorti
Model button 315 Phase Oscillator 313 Resolution parameter 315 Reverse arrow 314 Sample mode 313 Stiffness parameter 315 Type button 315 Vel Layer parameter 314 Volume parameter 312 Oszillator 1 turning on/off 312 Output section 321 Pan knob 305 Pan Mod button 323 ring modulator 316 Filter Bypass button 317 Volume slider 316 setting 303 loading 303 saving 303 signal flow 302, 306 snare 346 Solo button 305 Spread button 323 step sequencer 308, 332 MIDI control 338 Pattern Mode button 338 Playback Mode menu 3
