Logic Express 7 Plug-In Reference
Apple Computer, Inc. © 2004 Apple Computer, Inc. All rights reserved. Under the copyright laws, this manual may not be copied, in whole or in part, without the written consent of Apple. Your rights to the software are governed by the accompanying software licence agreement. The Apple logo is a trademark of Apple Computer, Inc., registered in the U.S. and other countries.
1 Preface 7 8 Contents Introducing Logic’s Plug-ins About This Manual Chapter 1 11 11 14 16 17 17 Basics Using Plug-ins The Plug-in Window Plug-in Settings Plug-in Automation Plug-ins From Other Manufacturers Chapter 2 19 19 20 Instruments and Effects Effect Plug-ins Instrument Plug-ins Chapter 3 23 23 24 24 25 Equalizer Channel EQ Silver EQ DJ EQ Individual EQs Chapter 4 27 27 30 30 32 33 34 Dynamic Compressor Silver Compressor Noise Gate Silver Gate Limiter Preset Multipressor Chapter 5
42 Phase Distortion Chapter 6 43 43 46 48 48 Filter AutoFilter Fuzz-Wah High Cut/Low Cut High Pass/Low Pass Filter Chapter 7 49 49 50 52 Delay Sample Delay Tape Delay Stereo Delay Chapter 8 53 53 54 55 55 57 57 Modulation Modulation Delay Chorus Flanger Phaser Tremolo Spreader Chapter 9 59 59 60 61 64 Reverb AVerb SilverVerb GoldVerb PlatinumVerb Chapter 10 67 67 68 Special Pitch Shifter II Denoiser Chapter 11 71 71 72 74 Helper Tuner Gain Levelmeter Chapter 12 75 75 76 76 Synthesiz
84 85 Modulator and Carrier The Output Section Chapter 14 87 87 ES M Parameters of the ES M Chapter 15 89 89 ES P Parameters of the ES P Chapter 16 93 93 ES E Parameters of the ES E Chapter 17 95 95 ES1 Parameters of the ES1 Chapter 18 103 KlopfGeist Chapter 19 105 106 107 111 134 135 136 EXSP24 Using Instruments File Organization Sample File Import EXSP24 Key Commands A Brief History of Sampling MIDI Controller List Chapter 20 139 139 GarageBand Instruments About GarageBand Instrument
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. Given a fast enough computer, you could conceivably arrange and mix an entire song using several software instruments, such as Logic’s ES1, or EXSP24, amongst others.
Logic’s plug-ins include the following features: • Real-time processing of audio. • Support for sample rates up to 96 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 and Audio Unit 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.
You can set all plug-in parameters in the Plug-in window. For further information please read “The Plug-in Window” on page 14. Closing the Plug-in window leaves the plug-in active. 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.
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 m m m m Adjusting Parameters 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 EXSP24 instance was loaded on Audio Instrument channel 1, switching between these tracks would automatically update the Plug-in window to show the ES1/EXSP24, respectively. Bypass The Bypass button allows a plug-in to be deactivated, but not removed from the insert/ instrument slot.
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.
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 11.). This allows processing of all audio and instrument signals.
Bus Effects When you use bus effects, a controlled amount of the signal is sent to the effect. Buses are typically used for effects that you want to apply to several signals at the same time. 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.
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.
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, 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 four frequency bands.
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. When your mouse cursor is in the access area of a band, its individual curve and parameter area will be highlighted and a pivot point appears. When you click-hold the mouse button directly on the (illuminated) pivot point of a band, vertical movements (up/down) will change its Q value. Horizontal movements (left/right) change the Frequency of the band.
Individual EQs Parametric EQ The Parametric EQ offers the following three parameters: • Hz: Center frequency • dB: Cut/Boost • Q: Quality A symmetrical frequency range on either side of the center frequency is boosted or cut. You can adjust the width of this frequency range with the Q control. High Shelving EQ/Low Shelving EQ • The Low Shelving Equalizer only affects the frequency range below the selected frequency.
4 Dynamic 4 This chapter introduces Logic’s Dynamic plug-ins. This includes the Compressor, Silver Compressor, Noise Gate, Silver Gate, Limiter, and Preset Multipressor 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. This “evening out” of the loud and soft passages means that the peak level remains pretty constant, and the overall loudness—the perceived volume—of a track is increased.
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. 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).
The three rotary knobs (at the top) influence the dynamic response of the noise gate. If you want the gate to open extremely quickly, say for percussive signals such as drums, set the Attack knob to the lowest value by turning it as far as it will go counterclockwise. If the signal fades in a bit more softly, as is the case with string pads and the like, a noise gate that opens too quickly can wreak havoc with the signal, causing it to sound unnatural.
When you’re working with noise gates, you’ll run across scenarios where the useful signal and the noise signal have levels that are near enough to be perceived as identical. A typical example is the crosstalk of a hi-hat—its signal tends to bleed into the snare drum track when you’re recording a drum kit. If you’re using a noise gate to isolate the snare, you’ll find that the hi-hat will also open the gate in many cases. To avoid this effect, the Noise Gate offers Side Chain filters.
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.
The interface of the Preset Multipressor features 12 radio buttons that allow you to choose between settings optimized for various genres; the names of the presets are pretty much self-explanatory. Make use of the different presets and use your ears to determine which one best fits your needs.
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 The Guitar Amp plug-in simulates the sound of several famous guitar amplifiers. You can process guitar signals directly within Logic, allowing you to reproduce the sound of high-quality guitar amplification systems.
Guitar Amp offers a range of Amplifier and EQ models that can be combined in a number of ways. The EQ models are equipped with the Bass, Mid, and Treble controls typical of guitar amplifiers. Four different amplifier models can be accessed via the Model radio buttons at the top. • British Clean—simulates the classic British Class A combos which have been continuously produced since the 1960s to the present, without any significant modification. This model is ideally suited for clean or crunchy rhythm parts.
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.
6 Filter 6 This chapter covers Logic’s filter effects. The filter effects include the AutoFilter, Fuzz-Wah, Low/High Pass Filter, and Low/High Cut plug-ins. AutoFilter The AutoFilter is an extremely versatile, resonance-capable lowpass filter, that offers a couple of truly unique features. The most important parameters are located to the right side of the Plug-in window: The Cutoff Freq. knob determines the point where the filter kicks in.
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 Logic Pro 7 EVD6’s Wah effect. Its parameters 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. Wah Mode There are simulations of several classic wah effects, as well as some basic filter types available. Available models are: off, ResoLP, ResoHP, Peak, CryB, Morl1, Morl2.
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).
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, Tremolo, 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. The modulation section consists of two LFOs, with variable frequencies (0 to 20 Hz).
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.
Tremolo The tremolo effect is a cyclic modulation of the amplitude, resulting in periodic volume changes of. As opposed to the vibrato effect which can be achieved with the Modulation Delay plug-in, the amplitude (not the frequency) is the modulated parameter. You’ll recognize this effect from vintage guitar combo amps (where it is sometimes incorrectly referred to as vibrato). The intensity of modulation is set with Depth. Rate defines the speed (frequency) of the modulation.
9 Reverb 9 This chapter describes Logic’s reverb effects. This includes AVerb, SilverVerb, GoldVerb, and PlatinumVerb AVerb Although the AVerb is based on a simple reverb algorithm, it delivers remarkably good results. The actual reverb algorithm is controlled by just four parameters: • As its name implies, Reflectivity defines how reflective the imaginary walls, ceiling, and floor will be. • Room Size challenges your architectural skills—use it to define the dimensions of simulated rooms.
SilverVerb The SilverVerb algorithm is controlled by just three parameters: As its name implies, Reflectivity defines how reflective the imaginary walls, ceiling, and floor will be. Room Size challenges your architectural skills—use it to define the dimensions of simulated rooms. The graphic display visually represents these parameter settings. Predelay determines the delay between the original signal and the reverb tail.
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 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 61. 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 001/011 button 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.
10 Special 10 This chapter introduces Logic’s special plug-ins. This includes the Pitch Shifter II and Denoiser plug-ins. Pitch Shifter II The Pitch Shifter II takes a minimalist approach—with just a few parameters available in the Editor view. Semi Tones is used to set the transposition value—in semi-tone increments, within a range of one octave upwards or downwards. Cents controls detuning in increments equivalent to 1/100th of a semi-tone step.
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.
If the noise floor of your recording is very high (on recordings from cassette—more than −68 dB), you should be content with reductions of 83 to 78 dB, provided that there aren’t any plainly audible side effects. After all, you have reduced the noise by more than 10 dB, which is less than half of the original volume.
11 Helper 11 This chapter introduces you to Logic’s Helper plug-ins. This includes the Tuner, Gain, and Levelmeter plug-ins. Tuner The ET1 Tuner plug-in can be used to tune acoustic instruments. This ensures that software instruments, existing samples or recordings are perfectly tuned to any new acoustic recordings you may make. You would normally insert the ET1 Tuner into an Input fader channel. Use couldn’t be simpler. There is a single tuning control at the bottom of the ET1 Tuner interface.
The numeric semicircle around the top of the ET1 interface displays the amount of variance—in cents—from the original pitch. The range is displayed in single semitone steps ±6 cents, and then in larger increments to a maximum of ±50 cents. If the incoming note is slightly flat, the indicator segments to the left will be illuminated. If the incoming note is slightly sharp, the indicator segments to the right will be illuminated. When the pitch is perfect, the center segment is lit.
Parameters The following parameters are available in the Gain plug-in: Gain This control adjusts levels from −96 to +24 dB, in steps of 0.1 dB. Press Shift while dragging on the Gain parameter to adjust in fine increments. 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.
Levelmeter The stereo Level Meter shows the current signal level on a logarithmic scale—using two blue bars. If the level is higher than 0 dB, the portion of the bar above the 0 dB point will turn red. The current peak values are displayed numerically (in dB increments), next to the Level Meter. The values are reset by clicking into the display. The Level Meter plug-in is switchable between Peak and RMS characteristics.
12 Synthesizer Basics 12 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.
13 EFM 1 13 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.
14 ES M 14 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.
15 ES P 15 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.
16 ES E 16 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.
17 ES1 17 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.
18 KlopfGeist 18 KlopfGeist is an instrument that is optimized to provide a metronome click in Logic. KlopfGeist is inserted on Audio Instrument channel 16 by 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 16.
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.
19 EXSP24 19 This chapter introduces Logic’s EXSP24 sampler. The EXSP24 sample player offers all of the playback 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 EXSP24 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 EXSP24 offers numerous synthesis options, enabling you to tailor sounds to meet your needs. Last, but not least: as a highly optimized Logic instrument, the EXSP24 offers great performance, even on slower machines. The EXSP24’s performance is scalable, so you can look forward to enhanced functionality and increased polyphony on future computer technology. The number of possible Sampler Instruments available for simultaneous playback is directly related to the computer’s processing and RAM resources.
File Organization File Types and File Organization The EXSP24 uses the following file types and hierarchical structures: Audio File A single sample on your hard disk. The EXSP24 is compatible with all audio file formats supported by Logic. Sampler Instrument A Sampler Instrument points to one or more audio files, and organizes them as multi samples or drum maps, respectively. Note: Audio files are not contained in a Sampler Instrument.
Important: The settings that can be stored and recalled in the Plug-in window are not part of the Sampler Instrument being loaded. Settings reside above the Sampler Instruments in the hierarchy: A setting contains a pointer to a Sampler Instrument, and when a new setting is selected, the Sampler Instrument it points to is automatically loaded. As such, settings are convenient for organizing and accessing your favorite Sampler Instruments.
When selecting a Sampler Instrument from a sub menu, a bold entry at the top of the root menu is added, to indicate the current selection. The sub menu that contains the selected Sampler Instrument is also shown in bold type, as are further sub menus. This makes it easy to trace the file path of the currently loaded Sampler Instrument.
3 When saving any newly created or modified Sampler Instruments, ensure that you use the “Save as” function and browse to the “Sampler Instruments” folder inside the new song folder. When saving on a per-song basis, you should observe the following folder hierarchy: • 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.
Sample File Import The EXSP24 is compatible with the EXS(P)24, AKAI S1000 and S3000, SampleCell, Gigasampler, and SoundFont2 sample formats, as well as the Vienna Library.
Importing SoundFont2 Files To make use of this functionality, simply copy or move your SoundFont2 files into the Sampler Instruments folder. Select the file name in the EXSP24 Sampler Instrument load flip-menu and the file will automatically be converted. An EXS Instrument file will be created in the Sampler Instruments folder which contains the original SoundFont2 file. The raw samples associated with the Sampler Instrument will be placed in a SoundFont Samples folder within the Logic program folder.
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 EXSP24 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.
This way, you can load and audition all of an AKAI CD-ROMs programs and files within Logic. Later, at your convenience, you can make use of your operating system’s file management utilities to remove or reorganize your imported AKAI sounds, as discussed in “File Organization” on page 107. To convert AKAI files 1 Select Options > AKAI Convert. This will launch a window similar to that shown above, with the text “Waiting for AKAI CD” spread across the four columns.
6 Any audio files imported will be stored within a folder which matches the name of the Volume. This folder is created within the Logic > AKAI Samples folder. The Sampler Instrument(s) created by the import procedure matches the Program name(s). It is placed inside the Sampler Instruments folder, or a sub folder as determined by the Save converted instrument file(s) into sub folder parameter discussed in “AKAI File Organization” on page 116.
• If converting an entire CD ROM, you can create a shortcut with the sample CD’s name—“Dance MegaSynth” for example. This can be placed in the Sampler Instruments folder directly, or as a sub-folder within the AKAI Samples folder. The advantage with the second method is that all imported AKAI Instruments will be placed under the AKAI Samples sub-menu within the EXSP24’s load window flipmenu.
To explain, many CD-ROMs created for AKAI samplers may feature several programs that contain single velocity layers for an instrument. AKAI samplers require the loading of an entire volume, or all necessary single programs, to be able to hear/play all velocity layers. All of these single programs are automatically assigned to the same MIDI channel and also react to the same MIDI program change number.
Vienna Library The EXSP24 features an additional interface for the Vienna Symphonic Library— Performance Set. The Performance Tool software provided by VSL needs to be installed to allow access to this interface. For details please refer to the VSL documentation. Plug-in Window Parameters Legato/Mono/Poly Buttons These switches determine the number of voices used by the EXSP24: • When Poly is selected, the maximum number of voices is set via the numeric field alongside the Poly button.
Unison Mode This mode plays multiple EXSP24 voices when each key is triggered: • In Poly mode, two voices per note. • In Mono or Legato mode, you can adjust the number of voices per note with the voices parameter (this value is limited to 8—which is more than enough for fat unison sounds!) The voices are equally distributed in the panorama field and are symmetrically detuned, dependent on the Random knob value.
• Save instrument as allows the storage of the currently opened Sampler Instrument under a different name. When invoked, a file dialog box will open. • Delete instrument will delete the opened Sampler Instrument. • (Recall default EXS24 mkI settings) does almost the same as the first entry, but the • • • • • settings for the former version of the EXS are recalled for the selected Instrument, especially the former modulation paths (see “EXSP24 mkI Modulation Paths” section, from page 131 onwards).
Hold Pedal and Crossfades Hold via This parameter determines the modulation source used to trigger the sustain pedal function (hold all currently played notes, and ignore their note off messages until the modulation source’s value falls below 64). The default is controller number 64 (MIDI standard). You can change it if there are reasons to prevent Sustain from using CC 64, or if you wish to trigger Sustain with another modulation source.
If planning to do so, however, please keep in mind the fact that all sources (except Velocity and Key) cause all velocity layers to run simultaneously—using up as many voices as there are layered Zones. This also happens in cases where the Zones are not audible at the current control level. If a continuous controller (such as the modulation wheel) is chosen, you can step through the velocity layers during playback.
Pitch Bend Up (▲) The amount of pitch bend (in semitones) that can be introduced by moving the pitch bend wheel to its maximum position. Pitch Bend Down (▼) The amount of pitch bend (in semitones) that can be introduced by moving the pitch bend wheel to its minimum position. When Linked is selected, the Pitch Bend Up value is used. Remote The Remote parameter allows you to easily pitch complete EXSP24 Instruments in realtime.
Please note that the upper half of the Pitcher slider can be set above the center position, and the lower half below the center position. When the Pitcher sliders are set in this fashion, lower velocity values cause the pitch to rise from the lower setting back to the original note pitch, while higher values cause it to fall from the higher setting back to the original value. In other words: the polarity of the pitch envelope can be changed according to velocity values.
Highpass (HP) The Highpass Filter is a 2 pole (12 dB/Oct.) design. A Highpass filter reduces the level of frequencies that fall below the cutoff frequency. It is useful for situations where you would like to suppress the bass and bass drum in a sample, for example, or for creating classic highpass filter sweeps. 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.
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 EXSP24.
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 EXSP24 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 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).
Select the format in which the EXSP24 handles the loaded sample data via the Sample Storage parameter. When set to Original, the samples are loaded into RAM at their original bit depth, and are converted to Logic’s internal 32 Bit floating point format on playback. When 32 Bit Float is selected, the samples are stored and loaded in this format.
Ignore Release Velocity This option also refers to the Release Trigger function and should always be set to on for this purpose. Regardless of whether or not your keyboard is able to send Release Velocity, you would want your samples played by the Release Trigger function to be louder or softer than the original Sample, or at the same volume, regardless of the initial velocity. When playing with Release Trigger, you would want the Release Velocity value to have the same value as the Initial Velocity value.
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. By the 15th century, organ builders had managed to simulate violins, flutes, trumpets, and even human-like sounds with their instruments. Some years later, organs were perfected that could imitate birdsong. Following the inception of film sound, several instruments were built that used film for the storage and playback of sound.
MIDI Controller List Common Pitch Filter 136 Chapter 19 EXSP24 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 Chapter 19 EXSP24 • 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.
20 GarageBand Instruments 20 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 EXSP24, ES1 (or other equivalent Logic instrument) instance simultanously.
Glossayr Glossary AD converter or ADC Short for analog/digital converter; a device that converts an analog signal to a digital signal. aftertouch MIDI data-type generated by pressure on keys after they have been struck. There are two types: Channel aftertouch, the value of which is measured by a full length keyboard sensor. It affects all played notes. Polyphonic aftertouch (rare) is individually measured and transmitted for each key. Aftertouch is also known as pressure.
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 Configuration window Logic window that provides an overview of all audio routing. Allows the copying of the entire audio configuration between Logic songs, and assists in renaming tasks.
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. The Audio Unit format is part of the Mac OS X operating system.
bit depth The number of bits a digital recording or digital device uses. The number of bits in each sample determines the theoretical maximum dynamic range of the audio data, regardless of sample rate. Also known as bit resolution, word length, or bit rate. bit rate See bit depth bit resolution See bit depth blue noise Highpass-filtered white noise, sounds like tape hiss. boosting The act of raising an audio level.
CD Audio Short for Compact Disc—Audio; current standard for stereo music CDs: 44.1 kHz sampling rate and 16 bit depth. cent A tuning subdivision of a semitone. There are one hundred cents in a semitone. Many of Logic’s software instruments contain a Fine parameter that allows sounds to be tuned in cent steps. channel strip A channel strip is a virtual representation of a channel strip on a mixing console.
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. This view is provided to allow access to additional parameters and to use less onscreen space. Core Audio Standardized audio driver system for all Macintosh computers running Mac OS X version 10.2 or higher.
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. All files used in Logic are digital. Also see analog for comparison. disclosure triangle A small triangle you click to show or hide details in the user interface. distortion The effect produced when the limit of what can be accurately reproduced in a digital signal is surpassed, resulting in a sharp, crackling sound.
envelope The envelope is the variation that a sound exhibits over time, an envelope basically determines how a sound starts, continues, and disappears. Synthesizer envelopes usually consist of Attack, Decay, Sustain, and Release phases. Environment The Environment is Logic’s brain: it graphically reflects the relationships between hardware devices outside your computer and virtual devices within your computer.
float window Window with special status which always “floats” on the surface above all other windows, but can only be operated with the mouse. Any Logic window can be opened as a float window by holding down Option while opening it. frame Unit of time. A second in the SMPTE standard is divided into frames that correspond to a single still image in a file or video.
interface 1) A hardware component such as a MIDI or audio device that allows Logic to “interface” (connect) with the outside world. You need an audio or MIDI interface to get sound/MIDI into and out of your computer. Also see audio interface. 2) A term that is used to describe Logic’s graphical elements that can be interacted with. An example would be the Arrange window, where graphical interface elements such as Regions are interacted with to create an arrangement, within the overall Arrange interface.
loop An audio clip that contains recurring rhythmic musical elements or elements suitable for repetition. Logic also supports Apple Loops. Loop function Loop is a Region parameter in Logic that creates “loop repetitions” for an Audio or MIDI Region. These repetitions will repeat until the song end point, or until another Region or folder (whichever comes first) is encountered on the same track in the Arrange window. LFO Abbreviation for Low Frequency Oscillator.
MIDI Multi mode Multi-timbral operating mode on a MIDI sound module where different sounds can be controlled polyphonically on different MIDI channels. A Multi mode sound module behaves like several polyphonic sound modules. General MIDI describes a 16-part multi mode (the ability to control 16 different parts individually). Most modern sound generators support multi mode. In Logic, multi mode sound modules are addressed via Multi Instrument Objects.
multitimbral This term describes an instrument or other device that can play different sounds at the same time, using several MIDI channels at the same time. Multi Trigger mode This term is associated with synthesizers such as the ES 1. In this mode, a synthesizer envelope usually is retriggered by every note played. mute Switch off an Audio Object or track’s audio output. You can mute a track by clicking the Track Mute button in the Track List.
pan, pan position The placement of mono audio signals in the stereo field, by setting different levels on both sides. Parameter box Field on the left side of Logic’s windows used to adjust the parameters of the selected Regions or Objects. 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.
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.
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. sequencer A sequencer is a computer application that allows you to record both digital audio and MIDI data and blend the sounds together in a software mixing console.
stereo Short for stereophonic sound reproduction of two different audio channels. Compare with mono. Sustain pedal A momentary footswitch that is connected to MIDI keyboards. It transmits MIDI controller number 64, which is recorded and played back by Logic. synthesizer A device (hardware or software) that is used to generate sounds. The word is derived from early attempts with mechanical and electronic machines to emulate (or synthesize) the sounds of musical instruments, voices, birdsong, and so on.
Undo function Function which reverses the previous editing operation. velocity Force at which a MIDI note is struck; controlled by the second data byte of a note event. virtual memory Area of the hard disk used as an extension of RAM memory by the computer. The disadvantage is its very slow access time, in comparison to physical RAM. WAV, WAVE The primary audio file format used by Windows-compatible computers.
zoom level The amount that a window’s contents (tracks, Regions, and Objects, for example) are magnified. Zooming in to a high level allows you to make more precise edits. Conversely, you can zoom all the way out to see the entire song and work on very large sections.
A ADC 141 AD converter 141 aftertouch 141 channel 141 polyphonic 141 AIFF 141 AKAI 141 alias 141 aliasing 141 allpass filter 141 amplifier 141 amplitude 141 analog 141 analog synthesizer 75 Arrange window 142 attack 142 attenuating 142 AU Audio Configuration window 142 Audio Instrument 142 Audio Instrument Object 11, 19, 20 Audio Mixer 142 Audio Object 11, 142 Audio Track Object 143 Aux Object 143 Bus Object 144 Input Object 149 Audio Track Object 143 Audio Unit.
Bounce function 7, 22 bpm 144 bus 20, 144 Bus Object 144 bypass 13, 15, 144 C carrier 144 Catch function 144 Catch button 144 cent 145 Channel EQ 23 using as Default EQ 24 channel strip 145 Channel Strip setting 145 Channel Strip Settings 17 chattering effect 31 checkbox 145 Chorus 54 chorus 145 click 145 Clipboard 145 Clip Distortion 41 clip circuit graphic 41 Drive 41 Filter 41 Frequency 41 Gain 41 Input Gain 41 Mix 41 Sum Filter 41 Symmetry 41 Tone 41 comb filter effect 55, 145 Compressor 27 Attack 28 A
Modulator 84 Modulator Pitch 83 Modulator Wave 85 pitch bend 86 Randomize 82, 86 Rate 83 sideband 81 Stereo Detune 85 Sub Osc Level 85 Transpose 82 Tune 82 Unison 82 Velocity 85 Voices 82 Vol Envelope 85 EMF 1 81 envelope 79, 148 attack 79 decay 79 release 79 sustain 79 Environment 148 layer 148 Environment Mixer 148 equalizer 23 Eraser tool 148 ES1 95 2', 4', 8', 16', 32' (octave transposition) 96 ADSR 98, 100 ADSR via Vel 97 AGateR 98 Analog 100 Bender Range 100 Chorus 101 Cutoff 97 Drive 96 Filter 97 Fil
LFO parameters 128 MIDI controller 136 Modulation Matrix 129 compatibility EXSP24 mkI 131 Multiple Outputs 132 Options button 120 Pan parameters 127 Pitch parameters 123 SampleCell 113 sample file import 111 Sampler Instrument 106 loading Sampler Instrument 106, 120 managing Sampler Instruments 108 saving Sampler Instrument song-related 109 searching Sampler Instrument 110 selecting Sampler Instrument 120 Setting 107 SoundFont2 file 112 Unison 120 Vienna Library 119 Voices 119 Volume parameters 127 H highp
M metronome click. See KlopfGeist modifier key 152 modulation 80 Modulation Delay 53 Anti Pitch 53 Constant Mod. 53 Feedback 54 Flanger-Chorus 54 LFO 53 LFO Mix 53 Mix 54 Stereo Phase 54 Vol. Mod.
EXSP24 mkII 105 fine-tunig parameters 14 Gain 72 GarageBand Instrument 139 GoldVerb 61 High Shelving EQ 25, 48 instrument 11, 19, 20 Levelmeter 74 Limiter 33 loading multiple plug-ins 17 Low Shelving EQ 25, 48 Modulation Delay 53 Noise Gate 30 numerical panel 14 operation 14 Overdrive 39 Parametric EQ 25 Phase Distortion 42 Phaser 55 Pitch Shifter II 67 PlatinumVerb 64 resetting parameters 14 rotary knob 14 Sample Delay 49 Setting 16 setting loading default automatically 16 Settings menu 15, 16 Silver Compr
synthesizer 158 analog 75 digital 75 virtual analog 75 T Tape Delay 50 Delay 50 Feedback 50 Flutter Intensity 51 Flutter Rate 51 Freeze 50, 52 Groove 50 High Cut 51 LFO Depth 51 LFO Speed 51 Low Cut 51 Mix 51 Smooth 51 Sync 50 Tempo 50 tempo changes 50 tempo 158 timing 158 toggle 158 Track List 158 Track Mixer 158 transient 158 Index transposition 158 Tremolo 57 graphic display 57 Rate 57 Smoothing 57 Stereophase 57 Symmetry 57 triangular wave 79 Tuner 71 V velocity 159 virtual analog synthesizer 75 vir
