Logic Express 8 Instruments and Effects
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1 Preface 9 9 Contents Introduction to the Logic Express Plug-ins Logic Express Effects and Instruments Chapter 1 13 13 15 Amp Modeling Bass Amp Guitar Amp Pro Chapter 2 21 22 22 23 24 Delay Echo Sample Delay Stereo Delay Tape Delay Chapter 3 27 28 29 30 31 32 33 Distortion Bitcrusher Clip Distortion Distortion Distortion II Overdrive Phase Distortion Chapter 4 35 37 41 43 45 47 48 49 52 53 54 Dynamics Compressor DeEsser Ducker Enveloper Expander Limiter Noise Gate Preset Multipressor Silver
Chapter 5 55 57 60 61 62 63 64 EQ Channel EQ DJ EQ Fat EQ Single Band EQs Silver EQ Frequency Ranges Used With EQ Chapter 6 65 66 70 75 85 88 Filter AutoFilter EVOC 20 Filterbank EVOC 20 TrackOscillator Fuzz-Wah Spectral Gate Chapter 7 91 91 94 Imaging Direction Mixer Stereo Spread Chapter 8 95 96 97 97 98 Metering BPM Counter Correlation Meter Level Meter Tuner Chapter 9 99 100 100 101 102 102 103 105 110 112 113 114 Modulation Chorus Ensemble Flanger Microphaser Modulation Delay Phaser Ri
Chapter 11 123 124 125 126 129 132 Reverb AVerb EnVerb GoldVerb PlatinumVerb SilverVerb Chapter 12 133 134 136 137 138 139 140 Specialized Denoiser Enhance Timing Exciter Grooveshifter Speech Enhancer SubBass Chapter 13 143 144 145 146 Utility Gain I/O Test Oscillator Chapter 14 149 150 150 150 151 152 153 154 159 161 163 165 167 168 169 169 170 171 172 EVOC 20 PolySynth Vocoder Basics What Is a Vocoder? How Does a Vocoder Work? How Does a Filter Bank Work? Using the EVOC 20 PolySynth EVOC 20 Pol
Chapter 15 175 176 177 178 179 180 EFM1 Global Parameters Modulator and Carrier FM Parameters The Output Section MIDI Controller Assignments Chapter 16 181 ES E Chapter 17 183 ES M Chapter 18 185 ES P Chapter 19 187 187 195 ES1 The ES1 Parameters MIDI Controller List Chapter 20 197 198 199 203 210 218 220 233 236 241 242 250 251 252 255 255 268 ES2 The ES2 Parameters Global Parameters Oscillator Parameters Filters Dynamic Stage (Amplifier) The Router The LFOs The Envelopes (ENV 1 to ENV
Chapter 21 275 277 279 281 282 283 284 294 312 334 337 338 EXS24 mkII Learning About Sampler Instruments Loading Sampler Instruments Working With Sampler Instrument Settings Managing Sampler Instruments Searching for Sampler Instruments Importing Sampler Instruments Parameters Window The Instrument Editor Setting Sampler Preferences Configuring Virtual Memory Using the VSL Performance Tool Chapter 22 339 339 340 External Instrument External Instrument Parameters Using the External Instrument Chapter 2
Preface Introduction to the Logic Express Plug-ins The Logic Express music and audio production software features a comprehensive collection of powerful plug-ins. These include innovative synthesizers, high quality effect plug-ins, and a powerful software sampler. This manual will introduce you to the individual effects and instruments—and their parameters. All plug-in parameters are discussed in detail.
Effect category Included effects Dynamic           Compressor (p. 37) DeEsser (p. 41) Ducker (p. 43) Enveloper (p. 45) Expander (p. 47) Limiter (p. 48) Noise Gate (p. 49) Preset Multipressor (p. 52) Silver Compressor (p. 53) Silver Gate (p. 54) EQ      Channel EQ (p. 57) DJ EQ (p. 60) Fat EQ (p. 61) Single Band EQs (p. 62) Silver EQ (p. 63) Filter      AutoFilter (p. 66) EVOC 20 Filterbank (p. 70) EVOC 20 TrackOscillator (p. 75) Fuzz-Wah (p. 85) Spectral Gate (p.
Effect category Included effects Specialized       Utility  Gain (p. 144)  I/O (p. 145)  Test Oscillator (p. 146) Denoiser (p. 134) Enhance Timing (p. 136) Exciter (p. 137) Grooveshifter (p. 138) Speech Enhancer (p. 139) SubBass (p. 140) The following table outlines the instruments included with Logic Express. Instrument category Included instruments Synthesizer        Drum synthesizer Ultrabeat (p. 343) Software sampler EXS24 mkII (p.
1 Amp Modeling 1 You can add the sound of a guitar and bass amplifier to your audio recordings and software instruments. Using a method known as component modeling, both the sound and functionality of musical instrument amplifiers, particularly those used with electric guitar and bass, can be emulated as an effect.
Bass Amp Parameters  Model pop-up menu: Choose from among nine different amplifier models. The choices are: Model Description American Basic 1970s-era American bass amp, equipped with eight 10-inch speakers. Well suited for blues and rock recordings. American Deep Based on the American Basic amp, but with strong lower-mid frequency (from 500 Hz on) emphasis. Well suited for reggae and pop recordings.
 Mid Frequency slider: Sets the center frequency of the mid band (between 200 Hz and 3000 Hz).  Output Level slider: Sets the final output level for the Bass Amp. Guitar Amp Pro The Guitar Amp Pro can emulate the sound of a variety famous guitar amplifiers and the cabinets/speakers used with them. You can process guitar signals directly within Logic Express, allowing you to reproduce the sound of high-quality guitar amp systems.
 The Microphone Position section is where you set the position of the microphone on the speaker.  The Microphone Type section is where you choose which type of microphone captures the amp’s sound. Amp Section  Amp pop-up menu: Choose the amp model you want to use. The choices are: 16 Model Description UK Combo 30W Neutral sounding amp, well suited for clean or crunchy rhythm parts. UK Top 50W Quite aggressive in the high frequency range, well suited for classical rock sounds.
 Speaker pop-up menu: Choose one of the 15 speaker models. The choices are: Speaker type Description UK 1x12 open back Classic open enclosure with one 12" speaker, neutral, well-balanced, multifunctional. UK 2x12 open back Classic open enclosure with two 12" speaker, neutral, well-balanced, multifunctional. UK 2x12 closed Loads of resonance in the low frequency range, therefore well suited for Combos: crunchy sounds are also possible with low Bass control settings.
 Gain knob: Sets the amount of pre-amplification applied to the input signal. This control has different effects, dependent on which Amp model is selected. For example, when using the British Clean amp model, the maximum Gain setting produces a powerful crunch sound. When using the British Gain or Modern Gain amps, the same Gain setting produces heavy distortion, suitable for lead solos.
Microphone Position and Microphone Type Sections After choosing a speaker from the Speaker menu, you can set the type of microphone emulated, and where the microphone is placed in relation to the speaker. Microphone Position Parameters  Centered button: When selected, places the microphone in the center of the speaker cone, also called on-axis. This placement produces a fuller, more powerful sound, suitable for blues or jazz guitar tones.
2 Delay 2 Delay effects store the input signal—and hold it for a short time—before sending it to the effect input or output. Most delays allow you to feed a percentage of the delayed signal back to the input, creating a repeating echo effect. Each subsequent repeat is a little quieter than the previous one. The delay time can often be synchronized to the project tempo by matching the grid resolution of the project, usually in note values or milliseconds.
Echo This simple echo effect always synchronizes the delay time to the project tempo, allowing you to quickly create echo effects that run in time with your composition. Echo Parameters  Time: Sets the grid resolution of the delay time in musical note durations—based on the project tempo. “T” values represent triplets, “.” values represent dotted notes.  Repeat: Determines how often the delay effect is repeated.  Color: Sets the harmonic content (color) of the delay signal.
Stereo Delay The Stereo Delay works much like the Tape Delay (see below), but allows you to set the Delay, Feedback, and Mix parameters separately for the left and right channel. The effect also features a Crossfeed knob for each stereo side. It determines the feedback intensity—or the level at which each signal is routed to the opposite stereo side. You can freely use the Stereo Delay on mono tracks or busses, when you want to create independent delays for the two stereo sides.
Tape Delay The Tape Delay simulates the warm sound of vintage tape echo machines, with the convenience of easy delay time synchronization to your project tempo. The Tape Delay is equipped with a highpass and lowpass filter in the feedback loop, making it easy to create authentic dub echo effects, and also includes an LFO for delay time modulation. The LFO produces a triangular wave, with adjustable speed and modulation intensity.
 Smooth: Evens out the LFO and flutter effect.  Dry and Wet: These individually control the amount of original and effect signal. Setting the Feedback When you set the Feedback slider to the lowest possible value, the Tape Delay generates a single echo. If Feedback is turned all the way up, the echoes are repeated ad infinitum. Note: The levels of the original signal and its taps (echo repeats) tend to accumulate, and may cause distortion.
3 Distortion 3 You can use Distortion effects to recreate the sound of analog or digital distortion, and to radically transform your audio. Distortion effects simulate the distortion created by vacuum tubes, transistors, or digital circuits. Vacuum tubes were used in audio amplifiers before the development of digital audio technology, and are still used in musical instrument amps today.
Bitcrusher The Bitcrusher is a low resolution digital distortion effect. You can use it to emulate the sound of early digital audio, create artificial aliasing by dividing the sample rate, or distort signals until they are unrecognizable. Bitcrusher Parameters  Drive slider and field: Sets the amount of gain (in decibels) applied to the input signal.  Resolution slider and field: Sets the bit rate (between 1 and 24 bits).
Clip Distortion Clip Distortion is a nonlinear distortion effect that produces unpredictable spectra. You can use it to simulate warm, overdriven tube sounds, and also to create drastic distortion. Clip Distortion features an unusual combination of serially connected filters. After being amplified by the Drive value, the signal passes through a highpass filter, and is then subjected to nonlinear distortion, as controlled by the Symmetry parameter.
 Input Gain field and slider (extended parameter): Sets the amount of gain applied to the input signal.  Output Gain field and slider (extended parameter): Sets the amount of gain applied to the output signal. Using Clip Distortion If you set the High Shelving Frequency to around 12 kHz, you can use it like the treble control on a mixer channel strip or a stereo hi-fi amplifier. Unlike those types of treble controls, however, you can boost or cut the signal by up to ±30 dB using the Gain parameter.
Distortion II Distortion II emulates the distortion effect section of a Hammond B3 organ. You can use it on musical instruments to recreate this classic effect, or use it creatively when designing new sounds. Distortion II Parameters . Â PreGain dial: Sets the amount of gain applied to the input signal. Â Drive dial: Sets the amount of saturation applied to the signal. Â Tone dial: Sets the frequency at which the signal is filtered.
Overdrive The Overdrive effect emulates the distortion produced by a field effect transistor (FET), which is commonly used in solid-state musical instrument amplifiers and hardware effects devices. When saturated, FETs generate a warmer sounding distortion than bipolar transistors. Overdrive Parameters  Drive slider and field: Sets the amount of saturation of the transistor.  Tone slider and field: Sets the cutoff frequency at which the signal is filtered.
Phase Distortion The Phase Distortion effect is based on a modulated delay line, similar to a chorus or flanger effect (for more information about these effects, see Chapter 9, “Modulation,” on page 99). Unlike these effects, however, in the Phase Distortion effect the delay time is not modulated by a low frequency oscillator (LFO), but rather by a lowpass-filtered version of the input signal itself. This means that the signal modulates its own phase position.
Using the Phase Distortion The input signal only passes the delay line and is not affected by any other process. The Mix parameter blends the effected signal with the original signal. The delay time is modulated by a side chain signal—namely, the input signal. The input signal passes through a resonant lowpass filter, with dedicated Cutoff frequency and Resonance controls. You can listen to the filtered side chain (instead of the Mix signal) by turning on the Monitor button.
4 Dynamics 4 You can use Dynamics effects to control the perceived loudness of your audio, add focus and punch to tracks and projects, and optimize the sound for playback in different situations. The dynamic range of an audio signal is the range between the softest and loudest parts of the signal (technically, between the lowest and the highest amplitude).
Some compressors, called multiband compressors, can divide the incoming signal into different frequency bands, and apply different compression settings to each band. This helps achieve the maximum level without introducing compression artifacts, and is typically used on an overall project mix. Expanders Expanders are similar to compressors, except that they raise, rather than lower, the signal when it exceeds the threshold. Expanders are used to enliven the audio signal.
Compressor The Compressor is designed to emulate the sound and response of a professional-level analog (hardware) compressor. It tightens up your audio by reducing sounds that exceed a certain threshold level, smoothing out the dynamics and increasing the overall volume—the perceived loudness. Compression helps bring the key parts of a track or a mix into focus while preventing softer parts from being inaudible. It is probably the most versatile and widely used sound-shaping tool used in mixing, next to EQ.
 Compression Threshold slider and field: Sets the threshold for the Compressor (the level above which the signal is reduced).  Peak/RMS buttons: Turn on one or the other to set whether the Compressor analyzes the signal using Peak or RMS method when using the Platinum Circuit Type.  Gain slider and field: Sets the amount of gain applied to the output signal.  Gain pop-up menu: Choose a value to raise the output level in order to compensate for volume reduction caused by compression.
Using the Compressor The following sections provide information on using each of the main Compressor parameters. Threshold and Ratio The most important Compressor parameters are Threshold and Ratio. The Threshold is the level (in decibels) above which the signal is reduced by the amount set as the Ratio. Because the Ratio is a percentage of the overall level, the more the signal exceeds the threshold, the more it is reduced.
Other Parameters Because the Compressor works by reducing levels, the overall volume of its output is typically lower than the input signal. You can adjust the output level using the Gain slider. You can use the Auto Gain parameter to compensate for the reduction in gain produced by compression, referenced to either –12 dB or 0 dB. Auto Gain sets the level of gain (amplification) to a value of T—(T/R), where T = the Threshold and R = the Ratio.
DeEsser The DeEsser is a frequency-specific compressor, designed to compress only a particular frequency band within a complex audio signal. It is used to eliminate hiss (also called sibilance) from the signal. The advantage of using the DeEsser instead of an EQ effect to cut high frequencies is that it compresses the signal dynamically rather than statically. This prevents the sound from becoming darker when no sibilance is present in the signal.
Suppressor Section  Suppressor Frequency knob: Sets the frequency band that is reduced when the Detector frequency sensitivity threshold is exceeded.  Strength knob: Sets the amount of gain reduction around the Suppressor frequency. Center Section  Detector and Suppressor frequency displays: The upper display shows the Detector frequency range, and the lower display shows the Suppressor frequency range (in Hz).  Smoothing slider: Sets the reaction speed of the gain reduction start and end phases.
Ducker Ducking is a common technique used in radio and television broadcasting: when the DJ/announcer speaks while music is playing, the music level is automatically reduced. When the announcement has finished, the music is automatically raised to its original volume level. The Ducker plug-in provides a simple means of performing this process. It can even reduce the music level before the speaker starts (but this introduces a small amount of latency).
 Release: Controls how quickly the volume returns to the original level. Set to a high value if you want the music mix to slowly fade up after the announcement. Using the Ducker For technical reasons, the Ducker plug-in can only be inserted in output and aux channels. To use the Ducker plug-in: 1 Insert the Ducker plug-in into an audio or aux channel strip. 2 Assign all track outputs that are supposed to “duck” (dynamically lower the volume of the mix) to a bus (using one of the Sends).
Enveloper The Enveloper is an unusual effect that lets you shape transients—the attack and release phases of a signal. This gives it a unique capability to shape the signal, and can be used to achieve impressive results different than any other dynamics effect. Enveloper Parameters The Gain and Time controls on the left apply to the attack portion of the signal, while the Gain and Time controls on the right apply to the release portion.
Emphasizing the release also boosts any reverb applied to the affected track. Conversely, toning down the release phase makes tracks originally drenched in reverb sound drier. This is particularly useful when working with drum loops, but it has many other applications as well. Let your imagination be your guide. When using the Enveloper, set the Threshold to the minimum value and leave it there.
Expander The Expander is similar to a compressor except that it increases, rather than reduces, the dynamic range above the Threshold level. You can use the Expander to add liveliness and freshness to your audio, specifically by emphasizing the transients of highly compressed signals. Expander Parameters  Threshold slider and field: Sets the level above which the Expander expands the signal.  Ratio slider and field: Sets the ratio by which the signal is expanded when it exceeds the threshold.
When using the Expander with Auto Gain active, the signal will sound softer even when the peak level remains the same; in other words, the expander decreases loudness. If you dramatically change the dynamics of a signal (by setting higher Threshold and Ratio values), you may find that you need to reduce the output level using the Gain slider to avoid distortion. In most cases, turning on Auto Gain will adjust the signal to the correct level.
The Lookahead parameter allows the Limiter to look forward in the audio so that it can react earlier to peak volumes by adjusting the amount of reduction. Using Lookahead causes latency, but this latency has no perceptible effect when you use the Limiter as a mastering effect, on previously recorded material. Set Lookahead to higher values if you want the limiting effect to take place before the maximum level is reached, creating a smoother transition.
 Hold knob and field: Sets the amount of time the gate is kept open after the signal falls below the threshold.  Release knob and field: Sets the amount of time it takes to fully close the gate after the signal falls below the threshold.  Hysteresis slider and field: Sets the difference (in decibels) between the threshold values that open and close the gate, to prevent it rapidly opening and closing when the input signal is close to the threshold.
The Hysteresis slider provides another option for avoiding chattering, without needing to define a minimum Hold time. You use it to set the range between the threshold values that open and close the Noise Gate. This is useful when the signal level jitters around the Threshold, fluctuating slightly but rapidly around it. This causes the Noise Gate to switch on and off repeatedly, producing an undesirable chattering effect.
Preset Multipressor The Preset Multipressor is an easy-to-use variant of the Logic Pro Multipressor plug-in. A multi-band compressor splits the incoming signal into different frequency bands before applying compression. These frequency bands are then compressed independently. Following compression, the frequency bands are mixed back together, and sent out of the plug-in.
Silver Compressor The Silver Compressor is a simplified version of the Compressor. It has fewer parameters and requires less CPU power. Silver Compressor Parameters  Gain Reduction display: Shows the amount of compression applied as the audio plays.  Threshold slider and field: Sets the threshold for the Compressor (the level above which the signal is reduced.)  Attack knob and field: Sets the attack time (the amount of time it takes for the compressor to react when the signal exceeds the threshold).
Silver Gate The Silver Gate is a simplified version of the Noise Gate. It has fewer parameters and requires less CPU power. Silver Gate Parameters  Lookahead slider and field: Adjusts how far ahead (in milliseconds) the noise gate analyzes the signal.  Threshold slider and field: Sets the level (in decibels) below which the signal is reduced.  Attack knob and field: Sets the amount of time it takes to fully open the gate after the signal exceeds the threshold.
5 EQ 5 EQ (short for Equalization) lets you shape the sound of your audio by changing the level of specific frequency bands. EQ is one of the most commonly used audio effects, both for music projects and in post-production work for video. You can use EQ to shape the sound of an audio file, track, or project by adjusting specific frequencies or frequency ranges. Using EQ, you can create both subtle and extreme changes to the sound of your projects.
Multiband EQs Multiband EQs give you control over a set of filters which, together, cover a large part of the frequency spectrum. On multiband EQs, you can set the frequency, bandwidth, and Q of each band independently. Using a multiband EQ (such as the Channel EQ or Fat EQ), you can perform extensive tone-shaping on any audio source. Multiband EQs are equally useful for shaping the sound of an individual track or an overall project mix.
Channel EQ The Channel EQ is a highly versatile multiband EQ. It offers eight frequency bands, including low and highpass filters, low and high shelving filters, and four flexible parametric bands. It also features an integrated Fast Fourier Transform (FFT) Analyzer that you can use to view the frequency curve of the audio you want to modify, allowing you to see which parts of the frequency spectrum need to be boosted or cut.
 Band 7 is a high shelving filter.  Band 8 is a lowpass filter.  Graphic display: Shows the current curve of each EQ band. You can adjust the frequency of each band by dragging left or right in the section of the display for that band, and adjust the gain of each band (except bands 1 and 8) by dragging up or down in the band’s section. The display reflects your changes immediately.
Using the Channel EQ How you use the Channel EQ depends on your audio and what you intend to do, but a useful workflow for many situations is as follows: with the Channel EQ set to a flat response (no frequencies boosted or cut), turn on the Analyzer and play the audio, observing the graphic display to see which parts of the frequency spectrum have frequent peaks and which parts stay at a low level. Notice particular places where the signal distorts or clips.
Using the Analyzer When you turn on the Analyzer, the Channel EQ shows a real time curve of all frequency components of the signal as the audio plays, superimposed over the EQ curves you set, using a Fast Fourier Transformation (FFT). The Analyzer curve uses the same scale as the EQ curves, allowing you to easily recognize important frequencies in the audio and use the EQ curves to raise or lower them.
Fat EQ The Fat EQ effect is a versatile multiband EQ with up to five individual frequency bands. You can use Fat EQ for individual tracks or for overall mixes. The Fat EQ includes a graphic display of the EQ curves and a set of parameters for each band. Fat EQ Parameters The main area of the Fat EQ window includes a graphic display area and a set of strips with parameters for each frequency band. To the right of the parameter section are the Master Gain slider and field.
Parameter Section Below the graphic display area are controls that both show the settings for each band, and which you can use to adjust each band’s settings. Â Frequency fields: Sets the frequency for each band. Â Gain knobs: Sets the amount of gain for each band. Â Q/Order fields: Sets the Q or bandwidth for each band (the range of frequencies around the center frequency that are altered). For bands 1 and 5, this changes the slope of the filter.
High Pass and Low Pass Filter The High Pass Filter affects the frequency range below the set frequency. Higher frequencies pass through the filter. You can use the High Pass Filter to eliminate the bass below a selectable frequency. In contrast, the Low Pass Filter affects the frequency range above the selected frequency. Both filter plug-ins offer the following parameters: Â Frequency field and slider: Sets the cutoff frequency. Â Order field and slider: Sets the filter order.
Frequency Ranges Used With EQ All sounds can be thought of as falling into one of three basic frequency ranges: bass, midrange, or high (or treble). These can each be further divided to include low bass, low and high midrange, and low and high highs. The following table describes some of the sounds that fall into each range: Name Frequency range Description High High 8–20 kHz Includes cymbal sounds and highest harmonics of instruments.
6 Filter 6 In addition to the filters of the EQ effects, you can use filters to change the character of your audio in both familiar and unusual ways. The Filter sub-menu contains a variety of filter-based effects that you can use to creatively modify your audio, including autofilters, filter banks, vocoders, wah-wah effects, and a gate that uses frequency rather than the amplitude (volume) as the criteria for which part of the signal is allowed to pass through.
AutoFilter The AutoFilter is a versatile filter effect with several unique features. You can use it to create classic, analog-style synthesizer effects, or as a tool for creative sound design. The filter cutoff can be dynamically modulated using both a synthesizer-style ADSR envelope and an LFO (low frequency oscillator).
LFO Section  Coarse and Fine Rate knobs and field: Use together to set the frequency of the LFO. Drag the Coarse slider to set the LFO frequency in Hertz, then drag the Fine slider to fine tune the frequency in 1/000ths of a Hertz.  Beat Sync button: When selected, the LFO is synchronized to the sequencer’s tempo.  Phase knob: Lets you shift the phase relationship between the LFO and the sequencer when Beat Sync is active.
Using the AutoFilter The following section provides additional information on using the parameters in the AutoFilter window. Filter Parameters The most important parameters are located on the right side of the AutoFilter window. The Filter Cutoff knob determines the point where the filter kicks in. Higher frequencies are attenuated, while lower frequencies are allowed to pass through. The Resonance knob controls how much frequencies around the cutoff frequency are emphasized.
LFO Parameters You set the waveform of the LFO by clicking one of 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 select a waveform, you can shape the curve with the Pulsewidth slider. Use the Coarse and Fine Frequency knobs to set the LFO frequency. The Rate Mod. (Rate Modulation) knob controls modulation of the LFO frequency independent of the input signal level.
EVOC 20 Filterbank The EVOC 20 Filterbank consists of two formant filter banks, which are also used in the EVOC 20 PolySynth vocoder plug-in. The input signal passes through the two filter banks in parallel. Each bank features volume faders for ten frequency bands, allowing you to adjust the volume of each band independently. Setting a fader to its minimum value completely suppresses the formants in that band.
Formant Filter Section The parameters in this section control the frequency bands in the two filter banks: Filter Bank A and Filter Bank B. Â Frequency band faders: Set the volume of each frequency band in Filter Bank A using the upper (blue) faders, and set the volume of each frequency band in Filter Bank B using the lower (green) faders. You can easily create complex bar curves by dragging horizontally across either row of faders.
 Boost A knob: Sets the amount of boost (or cut) applied to the frequency bands in Filter Bank A. The range is ±20 dB. This allows you to compensate for the reduction in volume caused by lowering the level of one or more bands. 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.
Modulation Section The parameters in this section control the LFOs that modulate the Formant Shift and Fade A/B parameters in the Formant Filter section, respectively. The LFO Shift parameters on the left modulate the Formant Shift parameter of the filter bands, and the LFO Fade parameters on the right modulate the Fade AB parameter. Â LFO Shift Intensity slider: Sets the amount by which the LFO modulates the Formant Shift parameter.
Output Section The parameters in this section control the overall output of the EVOC 20 Filterbank. Â Overdrive button: Turns the overdrive circuit on or off. Note: To hear the Overdrive effect, you may need to boost the level of one or both filter banks. Â Level slider: Sets the level of the output signal. Stereo Mode pop-up menu: Sets the input/output mode of the EVOC 20 Filterbank. The choices are m/s (mono input to stereo output), and s/s (stereo input to stereo output).
EVOC 20 TrackOscillator The EVOC 20 TrackOscillator is a vocoder with a monophonic pitch tracking oscillator. The tracking oscillator allows the EVOC 20 TrackOscillator to track (follow) the pitch of a mono input signal. For example, if the input signal is a vocal melody, the individual pitches of the sung notes will be tracked and mirrored by the synthesis engine. The EVOC 20 TrackOscillator features two formant filter banks, an analysis and a synthesis filter bank.
EVOC 20 TrackOscillator Parameters The EVOC 20 TrackOscillator window is divided into the following sections, from left to right: Analysis In, Synthesis In, Tracking Oscillator, Formant Filter, LFO, U/V Detection and Output. Analysis In Section The parameters in section control various aspects of the analysis signal. Â Attack knob: Controls how quickly the envelope follower coupled to each analysis filter band reacts to rising signals.
 Freeze button: When selected, the current analysis sound spectrum is held indefinitely. This can capture a particular characteristic of the source signal, which is then imposed as a complex sustained filter shape on the Synthesis section. While Freeze is selected, the analysis filter bank ignores the input source, and the Attack and Release parameters have no effect.
Synthesis In Section The parameters in section control various aspects of the synthesis signal. Â Synthesis In pop-up menu: Sets the synthesis signal source. The choices are: Â Oscillator (Osc.): Sets the tracking oscillator as the synthesis source. The oscillator tracks the pitch of the analysis input signal. Choosing Osc. activates the other parameters in the Synthesis section. If Osc is not chosen, the FM Ratio, FM Int, and other parameters in this section have no effect.
 FM Ratio knob: Sets the ratio between Oscillators 1 and 2, which defines the basic character of the sound. Even-numbered values (or their multiples) produce harmonic sounds are produced, while odd-numbered values (or their multiples) produce inharmonic, metallic sounds.  An FM Ratio of 1.000 produces results resembling a sawtooth waveform.  An FM Ratio of 2.000 produces results resembling a square wave with a pulse width of 50%.  An FM Ratio of 3.
 Click the value below the word Scale to display the Root/Scale pop-up menu.  Choose the scale or chord to use as the basis for pitch correction from the pop-up menu.  Set the root key of the respective scale or chord by vertically dragging the Root parameter, or double-click and enter a root between C and B. The Root parameter is not available when the Root/Scale value is chromatic or user.
 Frequency bar and fields: The blue bar above the upper row of faders controls the upper and lower frequencies for both filter banks. Drag the bar to move both the upper and lower frequencies, drag the left end to move only the lower frequency (the value range is 75 to 750 Hz), or drag the right end to move only the upper frequency (the value range is 800 to 8000 Hz). You can also edit the numerical values above the bar directly (between 80 and 8000 Hz).
LFO Section The parameters in this section control the LFO that can be used to modulate either the frequency (Pitch) of the tracking oscillator (vibrato), or the Formant Shift (Shift) parameter of the synthesis filter bank. It allows synchronous/non-synchronous modulation in bar, beat (triplet) or free values. Â Wave buttons: Select the waveform used by the LFO.
U/V Detection Section The U/V Detection section detects the unvoiced portions of the sound in the analysis signal, improving speech intelligibility. Please refer to “Unvoiced/Voiced (U/V) Detection” on page 165, for a full explanation of the U/V Detection principle. More information about improving speech intelligibility can be found in “Tips for Better Speech Intelligibility” on page 169. Â Sensitivity knob: Sets the degree of responsiveness of U/V detection.
 Level slider: Controls the amount of the signal (Noise, Noise + Synth, or Blend) used to replace the unvoiced content of the input signal. Warning: Care should be taken with this control, particularly when a high Sensitivity value is used, to avoid internally overloading the EVOC 20 TrackOscillator. Output Section  Signal pop-up menu: Choose the signal to send to the EVOC 20 TrackOscillator’s main outputs. The choices are: Voc(oder), Syn(thesis), and Ana(lysis). To hear the vocoder effect, choose Voc.
 Stereo Width knob: Controls how the output signals of the filter bands are distributed in the stereo field.  At the left position, the output of all bands are centered.  At the centered position, the output of all bands ascends from left to right.  At the right position, the bands are output evenly on the left and the right channel. The stereo/stereo mode (s/s) uses one A/B filter bank per channel.
Wah Section  Wah Mode pop-up menu: Choose one of the six modes, which emulate various classic wah effects and filter types, or choose off.  Auto Gain button: The wah effect can cause the output level to vary widely. Turning Auto Gain On compensates for this tendency, and keeps the output signal within a more stable range.  Wah Level knob: Sets the amount of the wah-filtered signal.  Relative Q slider: Adjusts the sharpness of the wah sweep by raising or lowering the filter peak.
Using the Fuzz-Wah The following sections cover various aspects of the Fuzz-Wah parameters. Setting the Wah Level With Auto Gain The wah effect can cause the output level to vary widely. Turning Auto Gain on compensates for this tendency, and keeps the output signal within a more stable range. To hear the difference Auto Gain can make: 1 Switch Auto Gain to on. 2 Raise the effect level to a value just below the mixer’s clipping limit. 3 Make a sweep with a high relative Q setting.
Spectral Gate The Spectral Gate separates the signal above and below the Threshold level into two independent frequency ranges—that you can modulate separately. It can produce some unusual and rich filtering effects. Spectral Gate Parameters  Threshold slider and field: Sets the threshold level at which the frequency band defined by the Center Freq. and Bandwidth parameters is divided into upper and lower frequency ranges.
Using the Spectral Gate Using the Center Freq. and Bandwidth parameters, set the frequency band you want to process using the Spectral Gate. The graphic display visually indicates the band defined by these two parameters. Once the frequency band is defined, use the Threshold parameter to set the level above and below which the frequency band is divided into upper and lower ranges.
7 Imaging 7 You can use the Logic Express Imaging plug-ins to extend the stereo base of a recording, and to alter perceived signal positions. These effects enable you to make certain sounds, or the overall mix, seem wider and more spacious. You can also alter the phase of individual sounds within a mix, to enhance or suppress particular transients. The following sections describe the Imaging plug-ins included with Logic Express: Â “Direction Mixer” on page 91. Â “Stereo Spread” on page 94.
Using the Direction Mixer The Direction Mixer is a simple plug-in to use, as it only offers two parameters: Spread and Direction. Each alters the incoming signal differently when either the LR or MS Input buttons are active. Using the Spread Parameter on LR Input Signals At a neutral value of 1, the left side of the signal is positioned precisely on the left, and the right side precisely on the right. As you decrease the Spread value, the two sides move towards the center of the stereo image.
What Is MS? Relegated to obscurity for a good long while, MS stereo (middle-side as opposed to left-right) has recently enjoyed a renaissance of sorts. Making a Middle Side Recording Two microphones are positioned as closely together as possible (usually on a stand or hung from the studio ceiling). One is a cardioid (or omnidirectional) microphone which directly faces the sound source that you want to record—in a straight alignment.
Stereo Spread The Stereo Spread effect is typically used for mastering. There are several ways to extend the stereo base (or perception of space), including use of reverbs and other effects, and altering the signal’s phase. They can all sound great, but can also weaken the overall sound of your mix by ruining transient responses, for example.
8 Metering 8 You can use the Metering plug-ins of Logic Express to analyze audio in a variety of ways. Each Metering plug-in allows you to view different characteristics of an audio signal. As examples: The BPM Counter displays the tempo of an audio file, the Correlation Meter displays the phase relationship, and the Level Meter displays the level of an audio recording.
BPM Counter You can use the BPM Counter to analyze the tempo of an audio track. Insert the plug-in into a track, to analyze the dynamic events of the audio signal. The detection circuit looks for any transients in the input signal. Transients are very fast, non-periodic sound events in the attack portion of the signal. The more obvious this impulse is, the easier it is for the BPM Counter to detect the tempo.
Correlation Meter The Correlation Meter displays the phase relationship of a stereo signal. Â A correlation of +1 (plus one, the far right position) means that the left and right channels correlate 100% (they are completely in-phase). Â A correlation of 0 (zero, the center position) indicates the widest permissible left/ right divergence, often audible as an extremely wide stereo effect.
Tuner You can tune both acoustic and electric musical instruments connected to your system using the Tuner. Tuning your instruments ensures that your recordings will be in tune with any software instruments, existing samples, or existing recordings in your projects. Tuner Parameters  Graphic tuning display: As you play, the pitch of the note appears in the semicircular area, centered around the Keynote.
9 Modulation 9 Modulation effects are used to add motion and depth to your sound. Modulation effects include chorus, flanging, and phasing among others, which make sounds richer or more animated. This is often achieved through the use of an LFO, which is controlled with parameters such as speed or frequency, and depth (also called width, amount, or intensity). You can also control the ratio of the affected (wet) signal and the original (dry) signal.
Chorus The Chorus effect delays the original signal. The delay time is modulated with an LFO. The delayed, modulated signal is mixed with the original, dry signal. You can use the Chorus effect to enrich the sound and create the impression that it’s being played by multiple instruments or voices, in unison. The slight delay time variations generated by the LFO simulate the subtle pitch and timing differences heard when several people perform together.
 Phase knob and field: Controls the phase relationship between the individual voice modulations. The value that you choose here is dependent on the number of voices, which is why it is shown as a percentage value rather than degrees. The value 100 (or –100) is equal to the greatest possible distance between the modulation phases of all voices.  Spread slider and field: Used to distribute the voices across the stereo field. When you set a value of 200%, the stereo base is expanded artificially.
Microphaser The Microphaser is a simple phaser effect that allows you to quickly create swooshing, phasing effects with just three parameters: Â LFO Rate slider and field: Defines the frequency, and therefore the speed, of the LFO. Â Feedback slider and field: Determines the amount of the effect signal that is routed back into the input. Â Intensity slider and field: Determines the amount of modulation.
 LFO Left Right Link button (only available in stereo instances): Switch on to tie the modulation rates of the left and right stereo channels to each other.  LFO Phase knob and field (only available in stereo instances): Controls the phase relationship between the individual channel modulations. At 0°, the extreme values of the modulation are achieved simultaneously for all channels. 180° or –180° is equal to the greatest possible distance between the modulation phases of the channels.
 Ceiling and Floor slider and fields: Use the individual slider handles to determine the frequency range that will be affected by the LFO modulations.  Order slider and field: Allows you to choose between different phaser algorithms. The more orders a phaser has, the heavier the effect.  Env Follow slider and field (Sweep section): Determines how much the frequency range (as set with the Ceiling and Floor controls) is modulated by the level of the input signal.
Ringshifter The Ringshifter effect combines a ring modulator with a frequency shifter effect. Both effects were popular during the 1970s, and are currently experiencing something of a renaissance. Â The ring modulator modulates the amplitude of the input signal using either the internal oscillator or a side chain signal. The frequency spectrum of the resulting effect signal equals the sum and difference of the frequency content in the two original signals.
Modes The four mode buttons determine whether the Ringshifter operates as a frequency shifter or as a ring modulator. Â Single (Frequency Shifter) button: The frequency shifter generates a single, shifted effect signal. The oscillator Frequency control determines whether the signal is shifted up (positive value) or down (negative value). Â Dual (Frequency Shifter) button: The frequency shifting process produces one shifted effect signal for each stereo channel—one is shifted up, the other is shifted down.
 Frequency control: Sets the frequency of the sine oscillator.  Lin(ear) and Exp(onential) buttons: Use these buttons to switch the scaling of the Frequency control:  The exponential scaling offers extremely small increments around the 0 point, which is useful for programming slow moving phasing and tremolo effects.  In the Lin(ear) mode, the resolution of the scale is even across the entire control range.
Output  Feedback knob and field: Sets the amount of the signal that is routed back to the effect input.  Stereo Width knob and field: Determines the breadth of the effect signal in the stereo field. Stereo Width only affects the effect signal of the Ringshifter, not the dry input signal.  Dry/Wet knob and field: Set the mix ratio of the dry input signal and the wet effect signal.
Modulation Sources The oscillator Frequency and Dry/Wet parameters can be modulated via the internal envelope follower and LFO. The oscillator frequency even allows modulation through the 0 Hz point, thus changing the oscillation direction. Envelope Follower The envelope follower analyzes the amplitude (volume) of the input signal and uses this to create a continuously changing control signal—a dynamic volume envelope of the input signal. This control signal can be used for modulation purposes.
Rotor Cabinet The Rotor Cabinet effect emulates the rotating loudspeaker cabinet of a Hammond organ’s Leslie effect. It simulates both the rotating speaker cabinet, with and without deflectors, and the microphones which pick up the sound. Â Rotor speed buttons: These switch the rotor speed. Chorale switches to slow movement, Tremolo to fast movement, and Brake stops the rotor. Â Cabinet Type menu: Choose between various cabinet sizes, shapes, and material.
 Wood & Horn IR: This setting uses an impulse response (a recording) of a Leslie with a wooden enclosure.  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.
Scanner Vibrato The Scanner Vibrato effect simulates the scanner vibrato section of a Hammond organ. You can choose between three different vibrato and chorus types. The stereo version of the effect features two additional parameters: Stereo Phase and Rate Right. These allow you to set the modulation speed independently for the left and right channels. The stereo parameters of the mono version of the Scanner Vibrato are hidden behind a transparent cover.
Spreader You can use the Spreader effect to widen the stereo spectrum of a signal. The Spreader effect periodically shifts the frequency range of the original signal in a nonlinear way, changing the perceived width of the signal. Â Â Â Â Intensity slider and field: Determines the modulation amount. Speed knob and field: Defines the frequency, and therefore the speed, of the LFO. Channel Delay slider and field: Determines the delay time in samples.
Tremolo The Tremolo effect modulates the amplitude of a signal, resulting in periodic volume changes. You’ll recognize this effect from vintage guitar combo amps (where it is sometimes incorrectly referred to as vibrato). The graphic display shows all parameters, except Rate. Â Depth slider and field: Determines the modulation amount. Â Rate knob and field: Defines the frequency, and therefore the speed, of the LFO. Â Symmetry and Smoothing knobs and fields: Use these to set the shape of the modulation.
10 Pitch 10 You can use the Pitch effects of Logic Express to transpose or correct the pitch of audio tracks. These effects can also be used for creating unison or slightly thickened parts, or even the creation of harmony voices. Logic Express includes the following Pitch effects: Â “Pitch Correction” on page 115 Â “Pitch Shifter II” on page 119 Â “Vocal Transformer” on page 121 Pitch Correction You can use the Pitch Correction plug-in to correct the pitch of audio tracks.
Pitch Correction Parameters  Normal and Low buttons: These determine the pitch range that is scanned (for notes that need correction).  Use Global Tuning button: Enable to use the project’s Tuning settings for the pitch correction process. If this button is switched off, you can use the Ref. Pitch field to freely set the desired reference tuning, in cents.  Scale field: Click to choose different pitch quantization grids from the Scale menu.  Root field: Click to choose the root note of the scale.
Using the Pitch Correction Plug-in You can use the Normal and Low buttons to determine the pitch range that you want to scan for notes that need correction. 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 result in inaccurate pitch detection. These parameter have no affect on the sound, they are simply optimized tracking options for the chosen target pitch range.
Excluding Notes From Correction 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 Eb (E flat) and Bb (B flat), instead of the E and B.
Understanding the Correction Amount Display The amount of pitch change is indicated in the horizontal bar displayed below the keyboard. The red marker indicates the average correction amount over a longer time period. 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, and adjust the Response accordingly.
 Vocals retains the intonation of the original with no change. Hence Vocals is wellsuited for any signals that are inherently harmonic or melodious, such as string pads.  Speech provides a compromise between the two by attempting to retain both the rhythmic and harmonic aspects of the signal. This is suitable for complex signals such as spoken-word recordings, rap music, and other hybrid signals such as rhythm guitar.
Vocal Transformer The Vocal Transformer allows you to manipulate vocal tracks in many different ways. You can use it to transpose the pitch of a vocal line, to augment or diminish the range of the melody, or even to reduce it to a single note—to mirror the pitches of a melody. No matter how you change the pitches of the melody, formants remain the same. You can shift the formants independently, which means that you can turn a vocal track into a Mickey Mouse voice, while maintaining the original pitch.
Setting the Pitch and Formant Parameters 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 below the Pitch Base field. Transpositions of a fifth upward (Pitch = +7), a fourth downward (Pitch = –5), or by an octave (Pitch = ±12) are the most useful, harmonically. As you alter the Pitch parameter, you might notice that the formants don’t change.
11 11 Reverb You can use Reverb effects to simulate the sound of acoustic environments such as rooms, concert halls, caverns, or the sound of infinite space. Sounds bounce off the surfaces of any space, or off objects within a space, repeatedly, gradually dying out until they are inaudible. The bouncing soundwaves result in a reflection pattern, more commonly known as a reverberation (or reverb).
Plates and Digital Reverb Effects The first form of reverb used in music production was actually a special room with hard surfaces (called an echo chamber). It was used to add echoes to the signal. Mechanical devices, including plates and springs, were used to add reverberation to the output of musical instruments and microphones. Digital recording introduced digital reverb effects, which consist of thousands of delays of varying lengths and intensities.
 Density/Time: Determines both the density and duration of the reverb.  Mix: Determines the balance between the effected (wet) and direct (dry) signals. EnVerb The EnVerb is a versatile reverb effect with a unique feature: It allows you to freely adjust the envelope of the diffuse reverb tail. The interface can be broken down into three areas:  Time parameters: These determine the delay time of the original signal and reverb tail, and change the reverb tail over time.
Sound Parameters  Density: Sets the reverb density.  Spread: Controls the stereo image of the reverb. At 0%, the effect generates a monaural reverb. At 200%, the stereo base is artificially expanded.  High Cut: Frequencies above the set value are filtered out of the reverb tail.  Crossover: Defines the frequency at which the input signal is split into two frequency bands, for separate processing.  Low Freq Level: Determines the relative reverb level of frequencies below the crossover frequency.
Early Reflection Parameters  Predelay: Determines the amount of time between the start of the original signal, and the arrival of the early reflections.  Room Shape: Defines the geometric form of the room. The numeric value (3 to 7) represents the number of corners in the room. The graphic display visually represents this setting.  Room Size: Determines the dimensions of the room. The numeric value indicates the length of its walls—the distance between two corners.
Setting Density and Diffusion Ordinarily, you want the signal to be as dense as possible. However, use of a lower Density value means the effect eats up less computing power. Beyond this, in rare instances, a high Density value can color the sound, which you can fix by simply reducing the Density knob value. Conversely, if you select a Density value that is too low, the reverb tail will sound grainy.
PlatinumVerb The PlatinumVerb allows you to edit both the early reflections and diffuse reverb tail separately, making it easier to precisely emulate real rooms. Its dual-band Reverb section splits the incoming signal into two bands, each of which is processed (and can be edited) separately. The interface can be broken down into four parameter groups: Â Early Reflections parameters: Emulates the original signal’s first reflections as they bounce off the walls, ceiling, and floor of a natural room.
 Stereo Base (only available in stereo instances): Defines the distance between the two virtual microphones that you are using in the simulated room. Spacing the microphones slightly further apart than the distance between two human ears generally delivers the best results. More realistic results can be obtained if you choose to use the distance between two ears located on opposite sides of the same head.
If you’re going for a natural-sounding, harmonic reverb, the transition between the early reflections and the reverb tail should be as smooth and seamless as possible. Set the Initial Delay so that it is as long as possible, without a noticeable gap between the early reflections and the reverb tail. Setting Density and Diffusion Ordinarily, you want the signal to be as dense as possible. However, use of a lower Density value means the effect eats up less computing power.
The Low Freq Level slider allows you to boost or attenuate the level of the low frequency band. 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. SilverVerb The SilverVerb is similar to the AVerb, but provides an additional LFO that you can use to modulate the reverberated signal.
12 Specialized 12 Logic Express includes a bundle of specialized plug-ins designed to address tasks often encountered during audio production. You should have a look at these specialized effects if you want to do one of the following: Â Eliminate or reduce noise below a threshold level (see “Denoiser” on page 134). Â Enhance the timing of audio recordings (see “Enhance Timing” on page 136). Â Add life to digital recordings by adding additional high frequency components (see “Exciter” on page 137).
Denoiser The Denoiser eliminates or reduces any noise below a threshold volume level. Denoiser Parameters  Threshold slider and field: Sets the volume level (the threshold) below which the DeNoiser reduces the signal.  Reduce slider and field: Sets the amount of noise reduction applied to sounds below the threshold. When reducing noise, remember that each 6 dB reduction is equivalent to halving the volume level (and each 6 dB increase equals a doubling of the volume level).
 Smoothing Transition knob: Adjusts how smoothing is applied to neighboring volume levels. If the Denoiser recognizes that only noise is present in a certain volume range, the higher you set the Transition Smoothing parameter, the more it also changes similar level values to avoid glass noise.  Graphic display: Shows how the lowest volume levels of your audio material (which should be mostly or entirely noise) are reduced.
Enhance Timing The Enhance Timing effect non-destructively enhances the timing of audio recordings. Enhance Timing Parameters  Intensity slider and field: Determines the amount of timing enhancement. Audio transients that don’t fall on the grid divisions (determined by the value chosen in the Grid menu) are corrected.  Grid menu: Allows you to choose between several grid divisions. As described above, the grid divisions serve as reference points for the timing correction process.
Exciter The Exciter generates high frequency components that are not part of the original signal, using a nonlinear distortion process that resembles overdrive and distortion effects. Unlike those effects, however, the Exciter passes the input signal through a highpass filter before feeding it into the harmonics (distortion) generator. This results in the artificial harmonics added to the signal having frequencies at least one octave above the threshold of the highpass filter.
Grooveshifter The Grooveshifter effect allows you to rhythmically vary recordings, imparting a swing feel to the track. Imagine a guitar solo played in straight eighth or sixteenth notes. The Grooveshifter can make this straightforward solo swing. The reference tempo is the project tempo. The Grooveshifter will automatically follow all changes to the project tempo. Note: The Grooveshifter is reliant on perfect matching of the project tempo with the tempo of the treated recording.
Speech Enhancer You can use the Speech Enhancer effect to improve speech recordings made with your computer’s internal microphone (if applicable). It combines denoising, advanced microphone frequency remodeling, and multiband compression. Â Denoise slider and field: Determines (your estimation of ) the noise floor in your recording, and therefore, how much noise should be eliminated. Settings towards 100 dB allow more noise to pass.
SubBass The SubBass plug-in generates frequencies below those of the original signal—in other words, an artificial bass. The simplest use for the SubBass is as an octave divider, similar to Octaver effect pedals for electric bass guitars. Where such pedals can only process a monophonic input sound source of clearly defined pitch, SubBass can be used with complex summed signals as well. SubBass creates two bass signals, derived from two separate portions of the incoming signal.
 Dry slider and field: Sets the amount of dry (non-effected) signal.  Wet slider and field: Sets the amount of wet (effected) signal. Using the SubBass Unlike a pitch shifter, the waveform of the signal generated by the SubBass is not based on the waveform of the input signal, but is sinusoidal (it uses a sine wave). Given that pure sine waves rarely sit well in complex arrangements, you can control the amount of (and balance between) the generated and original signals using the Dry and Wet sliders.
13 Utility 13 The Utility plug-ins are handy tools that can help you with routine tasks and situations that you may encounter when producing music. This includes the following tasks: Â Adjusting the level or phase of input signals (see “Gain” on page 144). Â Integrating external audio effects into Logic Express (see “I/O” on page 145). Â Generating a static frequency or sine sweep (see “Test Oscillator” on page 146).
Gain Gain lets you amplify (or reduce) the signal by a specific decibel amount. It is very useful when you are working with automated tracks during post-processing and want to quickly adjust levels. As examples: when you have inserted another effect that doesn’t have its own gain control, or when you want to change the level of a track for a remix version. Gain Parameters  Gain slider and field: Sets the amount of gain.
I/O The I/O plug-in allows you to use external audio effect units in a similar way to using the internal Logic Express effects. This only makes sense if you are using an audio interface which provides discrete inputs and outputs (analog or digital), used to send signals to and from the external audio effect unit. I/O Parameters  Output Volume field and slider: Adjusts the volume of the output signal.  Output menu: Assigns the respective output (or output pair) of your audio hardware to the plug-in.
Test Oscillator The Test Oscillator generates a static frequency or a sine sweep. The latter is a userdefined frequency spectrum tone sweep. Test Oscillator Parameters  Waveform buttons: Select the type of waveform to be used for test tone generation.  The Square Wave and Needle Pulse waveforms are available as either aliased or anti aliased versions. The latter when used in conjunction with the Anti Aliased button.  Needle Pulse is a single needle impulse waveform.
Using the Test Oscillator If you insert the Test Oscillator into an Insert slot of an audio channel, you must pass audio through this channel, in order to generate a signal. To use the Test Oscillator in an audio channel’s Insert slot: 1 Place any audio region onto a track. 2 Insert the Test Oscillator into this track’s channel, and start playback. You can also insert the Test Oscillator plug-in into the Instrument slot of instrument channel strips.
14 EVOC 20 PolySynth 14 The EVOC 20 PolySynth combines a vocoder with a polyphonic synthesizer, and can be played in real time. The EVOC 20 PolySynth is a sophisticated vocoder, equipped with a polyphonic synthesizer, and capable of receiving MIDI note input. This allows the EVOC 20 PolySynth to be played, resulting in classic vocoder choir sounds, for example. Single notes and chords played with the polyphonic EVOC 20 PolySynth will sing with the articulation of the analysis audio source.
Vocoder Basics If you are new to vocoders you should read this section. 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. A vocoder analyses and transfers the sonic character of the audio signal arriving at its analysis input to the audio signal present at its synthesis input.
An envelope follower is coupled to each filter band. The envelope follower of each band tracks (follows) any volume changes in the portion of the audio source allowed to pass by the associated bandpass filter. In this way, the envelope follower of each band generates dynamic control signals. These control signals are then sent to the synthesis filter bank where they control the levels of the corresponding synthesis filter bands. This is done via VCAs—Voltage Controlled Amplifiers.
Using the EVOC 20 PolySynth To make use of the EVOC 20 PolySynth, you need to insert the EVOC 20 PolySynth into an instrument channel’s Instrument slot, and you also need to provide an audio signal as the analysis audio source. You can do this by following these steps: 1 Select or create a new audio track in the Arrange window. 2 Insert (or record) an audio file—use a vocal part to start with—onto this audio track.
EVOC 20 PolySynth Parameters The EVOC 20 PolySynth interface is divided into six main sections. Sidechain Analysis section Formant Filter section Output section Synthesis section Modulation section U/V Detection section  Synthesis section: Controls the polyphonic synthesizer of the EVOC 20 PolySynth. See “Synthesis Parameters” on page 154.  Sidechain Analysis section: The parameters in this section define how the EVOC 20 PolySynth reacts to the analysis signal.
Synthesis Parameters The EVOC 20 PolySynth is equipped with a polyphonic synthesizer. It is capable of accepting MIDI note input. The parameters of the Synthesis section are described below. Mode Buttons These buttons determine the number of voices used by the EVOC 20 PolySynth: Â When Poly is selected, the maximum number of voices is set via the numeric field alongside the Poly button. Note: Increasing the number of voices also increases processor overhead.
Oscillator Section The EVOC 20 PolySynth is equipped with a two oscillator digital synthesizer which features a number of waveforms, and FM (Frequency Modulation). Further to these sound-generators in the Synthesis section is an independent noise generator. Click here to switch between Dual and FM mode There are two oscillator modes. Â Dual: Two oscillators make use of single-cycle digital waveforms to provide the synthesis sound source(s).
To switch between waveforms, do one of the following: m Click-hold on the numerical waveform field and drag up or down. When the desired waveform number is visible, release the mouse button. m Double-click the numerical field and input the desired value. Note: When in FM mode, the waveform of Wave 1 is a fixed sine wave. The waveform parameter of Wave 1 does not have an effect in this mode. Wave 2 Parameters The numerical value beside the Wave 2 label indicates the currently selected waveform type.
Dual Mode Parameters The parameters specific to Dual mode are found in the Wave 2 section, and the Balance slider to the right. Â Semi parameter: Adjusts the tuning of the second oscillator (Wave 2) in semitone steps. Â Detune parameter: Fine-tunes Wave 1 and Wave 2 in cents. 100 cents equals a semitone step. Doing so will detune Wave 1 in conjunction with Wave 2 around the tuning zero point. Â Balance slider: Allows you to blend the two oscillator signals (Wave 1 and Wave 2).
Tuning and Pitch Parameters  Analog knob: 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.  Tune: Defines the range of detuning.  Glide: Glide determines the time it takes for the pitch to slide from one note to another (portamento).  Bend Range: Determines the pitch bend modulation range in semitones.
Envelope Parameters The EVOC 20 PolySynth features an Attack/Release envelope generator used for level control of the Oscillator section. Â Attack slider: Determines the amount of time that it takes for the oscillators of the Synthesis section to reach their maximum level. Â Release slider: Determines the amount of time that it takes for the oscillators of the Synthesis section to reach their minimum level.
Release The Release parameter determines how quickly each envelope follower (coupled to each analysis filter band) reacts to falling signals. Longer Release times cause the analysis input signal transients to sustain longer at the vocoder’s output. Note: A long Release time on percussive input signals (a spoken word or hi-hat part, for example) will translate into a less articulate vocoder effect. Note that Release times that are too short result in rough, grainy vocoder sounds.
Formant Filter Parameters The Formant Filter display 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 and Low Frequency parameters, the Bands parameter, or the Formant Stretch and Shift parameters will result in visual changes to the Formant Filter display. This provides you with invaluable feedback on what is happening to the signal as it is routed through the two formant filter banks.
Lowest and Highest These parameters can be found in the two small fields on either side of the Formant Filter display. These switches determine whether the lowest and highest filter bands act as bandpass filters (like all of 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. Â In the Bandpass setting, the frequencies below/above the lowest/highest bands are ignored for both analysis and synthesis.
Resonance 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. Note: 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.
Intensity and Int via Whl The Intensity slider controls the amount of Formant Shift modulation by the Shift LFO. The Int via Whl slider for the Pitch LFO features a multi-part slider. The intensity of LFO pitch modulation can be controlled by the modulation wheel of an attached MIDI keyboard. The upper half of the slider determines the intensity when the modulation wheel is set to its maximum value, and the lower half when set to its minimum value.
Unvoiced/Voiced (U/V) Detection 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.
Sensitivity This parameter determines how responsive U/V detection is. By turning this knob to the right, more of the individual unvoiced portions of the input signal are recognized. 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.
Output Parameters This section covers the various parameters available in the EVOC 20 PolySynth output section. Signal This menu offers the choice of Voc(oder), Syn(thesis), and Ana(lysis). These settings allow you to determine the signal that you wish to send to the EVOC 20 PolySynth 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 buttons switch the ensemble effects on or off.
Block Diagram This block diagram illustrates the signal path in the EVOC 20 TrackOscillator (see “EVOC 20 TrackOscillator” on page 75) and EVOC 20 PolySynth. Analysis source Analysis section Legend Track -----------Side chain Audio signal Control signal R L Stereo to mono Parameter control Sensitivity U/V detection Frequency range between highest/lowest 1-5 Filter bank with five bands (example) TO: pitch analysis A Envelope follower 1-5 Freeze B Synthesis section TO: Max/Quant.
Tips for Better Speech Intelligibility The classic vocoder effect is very demanding, with regard to the quality of both the analysis and synthesis signals. Furthermore, the vocoder parameters need to be set carefully. Following, are some tips on both topics. Editing the Analysis and Synthesis Signals The following section outlines how you can edit the analysis and synthesis signals to achieve better speech intelligibility.
Avoiding Sonic Artifacts A common problem with vocoder sounds are sudden signal interruptions (ripping, breaking sounds) and rapidly triggered noises, during speech pauses. Release Parameter in the Analysis Section The Release parameter defines the speed that a given synthesis frequency band can decrease in level, if the signal level of the respective analysis band decreases abruptly. The sound is smoother when the band levels decrease slowly.
Achieving the Best Analysis and Synthesis Signals For good speech intelligibility, please keep these points in mind: Â 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. Â The synthesis signal must be constantly sustained, without breaks. The track should be played legato, as breaks in the synthesis signal will stop the vocoders output.
Vocoder 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. It was a composite device consisting of an analyzer and an artificial voice synthesizer.
Peter Zinovieff’s London-based company “EMS” developed a standalone—and altogether more portable—vocoder. EMS are probably best known for the “Synthi AKS” and VCS3 synthesizers. The EMS Studio Vocoder was the world’s first commercially available machine, released in 1976. It was later renamed the EMS 5000. Among its users were Stevie Wonder and Kraftwerk. Stockhausen, the German “Elektronische Musik” pioneer, also used an EMS vocoder.
15 EFM1 15 The 16-voice EFM1 is a simple, but powerful, frequency modulation synthesizer. It produces the typically rich bell and digital sounds that frequency modulation (FM) synthesis has become synonymous with. At the core of the EFM1 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.
 The bottom section houses the Output section, and features the Sub Osc Level and Stereo Detune parameters, plus the volume envelope, Main Level, and Velocity controls. A Randomize field is shown to the lower right.  The extended parameters panel (accessed by clicking the disclosure triangle at the lower left) allows you to assign MIDI controllers to the FM Amount (FM depth, in other words) and Vibrato parameters.
Modulator and Carrier The modulator and carrier parameters are outlined below. Harmonic In FM synthesis, the basic overtone structure is determined by the tuning relationship of the modulator and volume envelope. This is often expressed as a tuning ratio. In the EFM1, 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 volume envelope to any of the first 32 harmonics.
FM Parameters These parameters affect the frequency modulation aspects of the EFM 1. FM (Intensity) The modulator oscillator modulates the volume envelope frequency, resulting in newly generated sidebands that add new overtones. Turning up the FM (Intensity) control (the large dial in the center) produces increasing numbers of overtones—and the sound becomes brighter. The FM (Intensity) parameter is sometimes called the FM Index.
LFO The LFO (Low Frequency Oscillator) serves as a cyclic modulation source for FM Intensity or Vibrato. Turning the LFO control clockwise increases the effect of the LFO on FM Intensity. Turning it counter clockwise introduces a vibrato. In the center (0) position the LFO does not have an effect. You can easily center the LFO dial by clicking on the 0. Rate The speed/rate of the LFO cycles is set with the Rate parameter. The Output Section The EFM1 provides several level controls, as discussed below.
Velocity The EFM1 is able to respond to MIDI velocity, and reacts with dynamic sound and volume changes—harder playing will result in a brighter and louder sound. The sensitivity of the EFM1 in response to incoming velocity information is determined by the Velocity parameter. Set the Velocity control all the way to the left (counter-clockwise) if you don’t want the EFM1 to respond to velocity.
16 ES E 16 This chapter discusses the eight-voice polyphonic ES E synthesizer. The ES E (ES Ensemble) is designed for pad and ensemble sounds. It is great for adding atmospheric beds to your music, with minimal CPU overhead. All ES E parameters are discussed in the following section. Â 4, 8, 16 buttons: Determine the ES E’s octave transposition.
 Resonance knob: Sets the resonance of the ES E’s dynamic lowpass filter.  AR Int knob: The ES E features one simple envelope generator per voice—offering an Attack and a Release parameter. The AR Int parameter defines the amount of cutoff frequency modulation (applied by the envelope generator).  Velo Filter knob: Sets the velocity sensitivity of the cutoff frequency modulation (applied by the envelope generator). This parameter has no effect if AR Int is set to 0.
17 ES M 17 The monophonic ES M (ES Mono) is a good starting point if you’re looking for bass sounds that punch through your mix. The ES M compact synthesizer features an automatic fingered portamento mode, making bass slides easy. It also features an automatic filter compensation circuit that delivers rich, creamy basses, even when using higher resonance values. All ES M parameters are discussed in the following section. Â 8, 16, 32 buttons: Set the ES M’s octave transposition.
 Decay (Filter) knob: Sets the decay time of the filter envelope. It is only effective if Int is not set to 0.  Velo (Filter) knob: Determines the velocity sensitivity of the filter envelope. This parameter is only effective if Int is not set to 0.  Decay (Volume) knob: Sets the decay time of the dynamic stage. The attack, release, and sustain times of the synthesizer are internally set to 0.  Velo (Volume) knob: Determines the velocity sensitivity of the dynamic stage.
18 ES P 18 This chapter introduces you to the 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.
 Speed knob: Controls the rate of the oscillator frequency or cutoff frequency modulation.  Frequency knob: Sets the cutoff frequency of the resonance-capable dynamic lowpass filter.  Resonance knob: Sets the resonance of the dynamic lowpass filter. Increasing the Resonance value results in a rejection of bass (low frequency energy) when using lowpass filters. The ES P compensates for this side-effect internally, resulting in a more bassy sound.
19 ES1 19 This chapter introduces the 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' Buttons 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.
Filter Parameters This section outlines the filter parameters available to the ES1. Drive This is an input level control for the lowpass filter, which allows you to overdrive the filter. Its use changes the behavior of the Resonance parameter, and the waveform may sound distorted. 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.
ADSR via Vel The main envelope generator (ADSR) modulates the cutoff frequency over the duration of a note. The intensity of this modulation can be set to positive or negative values, and can respond to velocity information. If you play pianissimo (Velocity = 1), the modulation will take place as indicated by the lower arrow. If you strike with the hardest fortissimo (Velocity = 127), the modulation will take place as indicated by the upper arrow.
All ADSR parameters will always remain active for the filter (ADSR via Vel). A stands for attack time, R for release time, while Gate is the name of a control signal used in analog synthesizers, which tells an envelope generator that a key is pressed. As long as an analog synth key is pressed, the gate signal maintains a constant voltage. Used as a modulation source in the voltage controlled amplifier (instead of the envelope itself ), it creates an organ type envelope without any attack, decay, or release.
Router The router defines the modulation target for LFO modulation and the modulation envelope. Only one target can be set for the LFO, and another one can be set for the modulation envelope.
Mod Envelope The modulation envelope itself only has one parameter. You can set a percussive type of decay envelope (low values), or attack type envelopes (high values). A full setting of the modulation envelope delivers a constant, full level.
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.
MIDI Controller List Controller Number Parameter Name 12 Oscillator pitch buttons 13 Oscillator waveform 14 Mix slider 15 Waveform of sub oscillator 16 Drive slider 17 Cutoff slider 18 Resonance slider 19 Slope buttons 20 ADSR via Vel: lower slider 21 ADSR via Vel: upper slider 22 Attack slider 23 Decay slider 24 Sustain slider 25 Release slider 26 Key slider 27 Amplifier Envelope Selector buttons 28 Level via Velocity: lower slider 29 Level via Velocity: upper slider 3
20 ES2 20 The ES2 synthesizer combines a powerful tone generation system with extensive modulation facilities. The ES2 provides three oscillators, which can be synchronized and ring-modulated. Pulse-width modulation is also possible. Oscillator 1 can be modulated in frequency by Oscillator 2, and is capable of producing FM-style synthesizer sounds. In addition to the classic analog synthesizer waveforms, the ES2 oscillators also provide 100 single-cycle waveforms, known as Digiwaves.
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 keyboard note is pressed.
Global Parameters These parameters impact on the overall instrument sound produced by the ES2. You can find the global parameters to the left of the oscillators, and above the filter section. Global parameters Global parameters Tune Tune sets the pitch of the ES2 in cents. 100 cents equals a semitone step. 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.
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.
Note: If you switch to Legato, you need to play legato to actually hear the Glide parameter taking effect. Note: On several monophonic synthesizers, the behavior in Legato mode is referred to as single trigger, while Mono mode is referred to as multi trigger. Voices The maximum number of notes that can be played simultaneously is determined by the Voices parameter. Maximum value for Voices is 32.
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). Â 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.
Oscillator Parameters The following section describes the parameters that you can set individually for each oscillator. You can find these parameters in the silver area to the right of the ES2 interface. Muting Oscillators By clicking on the green numbers to the right of the oscillators, you can mute and unmute them independently. This saves processor power. Frequency Knobs The Frequency knobs set the pitch in semitone steps over a range of ±3 octaves.
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.
Given the ES2’s ability to modulate Digiwaves, you can produce sounds reminiscent of classic wavetable synthesizers from PPG and Waldorf (and Korg’s Wavestation). 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 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 the “Pulse Width Modulation” section, on page 206).
Sync The rectangular and sawtooth waveforms also feature a Sync option. In this mode, the frequency of Oscillator 2 (or 3, respectively) is synchronized to the frequency of Oscillator 1. This does not mean that their frequency controls are simply disabled. They still oscillate at their selected frequencies, but every time that Oscillator 1 starts a new oscillation phase, the synchronized oscillator is also forced to restart its phase from the beginning.
White and Colored Noise (Oscillator 3 Only) Unlike Oscillator 2, Oscillator 3 is not capable of producing ring modulated signals or 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.
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. Read more about this in the “Vector Mode Menu” section, on page 241, and “The Vector Envelope” section, on page 242.
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). Filter button The Filter button bypasses (switches off ) the entire filter section of the ES2.
The mono output signal of Filter 2 is then fed into the input of the dynamic stage (the equivalent of a VCA in an analog synthesizer), where it can be panned in the stereo spectrum, and then fed into the effects processor. In the graphic to the right, 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.
Filter Blend and Signal Flow No matter whether parallel or series filter configurations are chosen, a Filter Blend setting of –1 results in only Filter 1 being audible. A Filter Blend setting of +1 will limit audibility to Filter 2. This is reflected in the user interface. In conjunction with the overdrive/distortion circuit (Drive) and a series cabling configuration, the ES2’s signal flow is far from commonplace.
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 Series mode  Filter Blend set to 0 (center position)  Filter 1 set to Peak Filter mode  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.
∏ Tip: If you are new to synthesizers, experiment with a simple saw wave, using Oscillator 1, and Filter 2 (lowpass filter, Filter Blend = +1) on its own. Experiment with the Cutoff Frequency and Resonance parameters. You’ll quickly learn how to emulate a number of recognizable sounds, and will pick up the basic principles of subtractive synthesis intuitively.
Filter Slope A filter can not completely suppress the signal portion outside the frequency range defined by the Cutoff Frequency parameter. The slope of the filter curve expresses the amount of rejection applied by the filter (beneath the cutoff frequency) in dB per octave. Filter 2 offers three different slopes: 12 dB, 18 dB and 24 dB per octave. Put another way, the steeper the curve, the more severely the level of signals below the cutoff frequency are affected in each octave.
 The abbreviation BP stands for bandpass. In this mode, only the frequency band directly surrounding the cutoff frequency can pass. All other frequencies are cut. The Resonance parameter controls the width of the frequency band that can pass. The bandpass filter is a two-pole filter with a slope of 6 dB/octave on each side of the band. 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.
Handling Processing Power Economically The ES2 is designed to make the most efficient use of computer processing power. Modules and functions that are not in use don’t use processing power. This principle is maintained by all elements of the ES2. As examples: If only one of the three oscillators is in use, and the others are muted, less processing power is required. If you do not modulate Digiwaves, or if you disengage the filters, processing power is saved.
Router Modulation Target: Amp The dynamic stage can be modulated by any modulation source in the Router. The modulation target is called AMP in the Router. Note: If you select AMP as the target, LFO1 as the source, and leave via set to Off in the Router, the level will change periodically, based on the current Rate of the LFO—and you’ll hear a tremolo.
The Router The ES2 features a modulation matrix, called the Router. If the vector envelope is displayed, click the Router button to view the Router. Any modulation source can be connected to any modulation target—much like an oldfashioned 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.
You can grab the area between the two slider halves with the mouse and drag both halves simultaneously. If this area is to small to be grabbed with the mouse, just click on a free part of the slider track and move the mouse up or down to move the area. 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.
Modulation Targets The following targets are available for real-time modulation. Note: These modulation targets are also available for the X and Y axes of the X/Y modulator (the Square). See “The Square” section, on page 241. Pitch 123 This target allows the parallel modulation of the frequencies (pitch) of all three oscillators. If you select an LFO as the source, this target leads to siren or vibrato sounds.
This leads to the following rules of thumb for modulation intensity settings. Â Modulation intensity of 8 equals a pitch shift of 10 cents. Â Modulation intensity of 20 equals a pitch shift of 50 cents, or one quarter tone. Â Modulation intensity of 28 equals a pitch shift of 100 cents, or one semitone. Â Modulation intensity of 36 equals a pitch shift of 200 cents, or two semitones. Â Modulation intensity of 76 equals a pitch shift of 1,200 cents, or one octave.
OscWaveB The transitions between Digiwaves during a wavetable modulation are always smooth. Depending on the modulation intensity, an additional OscWaveB target can be used to continuously modulate the shape of the transitions from smooth to hard. It applies to all Oscillators. Osc1WaveB If wavetable modulation is active for a Digiwave using Osc1Wav, 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 214. 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 lowpass filter).
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. Nevertheless, pan can still be modulated, with positions being moved in parallel. Lfo1Asym Lfo1Asym (Lfo1 Asymmetry) can modulate the selected waveform of LFO 1. In the case of a square wave, it changes its pulse width.
Env2Rel Env2Rel (Envelope 2 Release) modulates the Release time of the second envelope generator. Env2Time Env2Time (Envelope 2 All Times) modulates all of ENV2’s time parameters: Attack time, Decay time, Sustain time, and Release time. Env3Atck Env3Atck (Envelope 3Atck) modulates the Attack time of the third envelope generator. Env3Dec Env3Dec (Envelope 3 Decay) modulates the Decay time of the third envelope generator.
ENV3 Envelope Generator 3 is described in “The Envelopes (ENV 1 to ENV 3)” section, on page 236. 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. The cursor can be moved to any position in the Square, either manually or controlled by the vector envelope. See “The Square” on page 241 and “The Vector Envelope” on page 242.
Touch Aftertouch serves as modulation source. The ES2 reacts to poly pressure (polyphonic aftertouch). It uses the sum of channel pressure and the note-specific poly pressure value. Note: If you set the target to Cut 1+2, the cutoff frequencies will rise and fall, dependent on how firmly you press a key on your touch-sensitive MIDI keyboard after the initial keystrike. Whl+To The modulation wheel and aftertouch serve as modulation sources.
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. The (random) note-on modulation remains constant throughout the duration of the note until the next note-on trigger. Note: There is no value change when playing legato, while in legato mode.
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. If you select Pitch 123 as the target, modulate it with the LFO1 source, and select Keyboard as the via value, the vibrato depth will change, dependent on key position. Put another way, the vibrato depth will be different for notes higher or lower than the defined Keyboard position.
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.
The LFOs LFO is the abbreviated form of Low Frequency Oscillator. In an analog synthesizer, LFOs deliver modulation signals below the audio frequency range—in the bandwidth that falls between 0.1 and 20 Hz, and sometimes as high as 50 Hz. LFOs serve as modulation sources for periodic, cyclic modulation effects. If you slightly modulate the pitch of an audio oscillator at a rate (speed, LFO frequency) of, say, 3–8 Hz, you’ll hear a vibrato.
EG (LFO1) At its center position—which can be accessed by clicking the middle mark—the modulation intensity is static: it won’t be faded in or out at all. At positive values, it is faded in. The higher the value, the longer the delay time is. At negative values, it is faded out. The lower (onscreen) the slider is positioned, the shorter the fade out time is. The function is abbreviated as EG because the fading in or out is internally performed by an ultra-simple envelope generator.
Note: An interesting effect you may wish to try out is achieved by modulating Pitch123 with a suitable modulation intensity that leads to an interval of a fifth. Choose the upper rectangular wave to do so. Sample & Hold The two lower waveform settings of the LFOs output random values. A random value is selected at regular intervals, defined by the LFO rate. The upper waveform delivers exact steps of randomization. At its lower setting, the random wave is smoothed out, resulting in fluid changes to values.
The Envelopes (ENV 1 to ENV 3) In addition to the complex vector envelope, described in “The Vector Envelope” section, on page 242, the ES2 also features three envelope generators per voice. On both the front panel and as a source in the Router, they are abbreviated as ENV 1, ENV 2, and ENV 3, respectively. Note: The roots of the term envelope generator and its basic functionality are described in the “Envelopes” section, on page 423. The feature sets of ENV 2 and ENV3 are identical.
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. Decay Release button Trigger Modes menu Attack via Velocity slider 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. Â 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.
The Parameters of ENV 2 and ENV 3 The feature sets of ENV 2 and ENV 3 are identical, but it is always the task of ENV 3 to define the level of each note—to modulate the dynamic stage. ENV 3 is available for simultaneous use as a source in the Router as well. The envelope’s time parameters can also be used as modulation targets in the Router. Note: See the “Envelopes” section, on page 423 for information on the basic functionality and meaning of envelope generators.
Sustain and Sustain Time When the Sustain Time (rise) parameter is set to its center value—which can be achieved by clicking the center symbol shown above—the Sustain level behaves like the Sustain parameter of any synthesizer ADSR envelope. In this position, the Sustain level (abbreviated as S) defines the level that is sustained for as long as the key remains depressed, following the completion of the Attack time and Decay time phases.
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.
Note: Like all of the ES2’s parameters, the movements of the cursors in the Triangle and Square can be recorded and automated by Logic Express. This automation data can be edited and looped in Logic Express. This is completely independent of the cyclic modulations of the vector envelope. Vector modulation of the Square and Triangle should be disabled for this type of use (Vector mode = off ).
Envelope Points, Times, and Loops The vector envelope consists of up to 16 points on the time axis. Each point can control the position of both the Triangle and Square’s cursors. The points are numbered sequentially. Point 1 is the starting point. In order to edit a point, simply select it—by clicking on it. Note: A number of vector envelope editing commands can be quickly accessed in a shortcut menu. Control-click anywhere on the vector envelope to open it.
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. In order to define a point as the loop point, click on the turquoise strip below the desired point. A loop point is indicated by an L in the strip below.
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 Triangle 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 The Solo Point button 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.
Loop Rate Just as every LFO has a speed (or rate) parameter, the loop can be set to cycle at a defined Loop Rate. And just like an LFO, the vector envelope Loop Rate can be synchronized to the project tempo automatically. Â If you switch the Loop Rate to as set, the duration of the loop cycle is equal to the sum of the times between the sustain and loop points. Click on the field labeled as set (below the Rate slider) to select.
Time Scaling You can stretch and compress the entire vector envelope. As an example, to double the vector envelope’s speed, it’s not necessary to halve the time values of every point. All you need to do is set Time Scaling to 50%. Â The range for the Time Scaling parameter is from 10% to 1000%. It is scaled logarithmically. Â If the Loop Rate is “as set,” the scaling also affects the loop. If not (Loop Rate = free or sync), the setting will not be affected by Time Scaling.
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. Despite the inclusion of this integrated effects processor, please feel free to process the ES2 with the other effect plug-ins included in Logic Express. The sound and parameter set of the integrated effects unit is reminiscent of classic pedal effects, designed for the electric guitar.
Using Controls and Assigning Controllers The section at the bottom of the ES2 interface provides three modes, accessed by clicking the buttons on the left: Â Macro: Shows a number of macro parameters which affect groups of other parameters. Â MIDI: Allows you to assign MIDI controllers to particular Router channels (see “MIDI Controllers A–F” on page 229). Â Macro Only: Replaces the ES2 interface with a dedicated (and much smaller) view that is limited to the macro parameters.
Random Sound Variations The ES2 offers a unique feature that allows you to vary the sound parameters randomly. You can define the amount of random variation, and can restrict the variations to specific sonic elements. The random sound variation feature will inspire and aid (or occasionally amuse) you when creating new sounds. Clicking the RND button randomly alters the sound. The process is triggered by a single click and can be repeated as often as you like.
RND Destination Some aspects of your sound may already be ideal for the sound you had in mind. As such, it may not be desirable to alter them. Say your sound setting has a nice percussiveness, and you’d like to try a few sonic color variations while retaining this percussive feel. To avoid the random variation of any attack times, you can restrict the variation to the oscillator or filter parameters, with the envelope parameters excluded from the variation process.
LFOs All LFO parameters of all LFOs are varied. Router All Router parameters of all Router channels are varied with all Intensities, Target, via, and Source parameters. FX All effects parameters are varied. Vector Envelope All vector envelope parameters are varied, including the XY routing of the Square. Vector Env Mix Pad The Oscillator mix levels (Triangle cursor positions) of the vector envelope points are altered.
Tutorials You will find the settings for these tutorials in the Tutorial Settings folder in the Settings menu (in the header of the ES2 window). Sound Workshop The Sound Workshop will guide you—from scratch—through the creation of commonly used sounds. The following tutorial section will also guide you through the sound creation process, but starts you off with a number of templates.
Three Detuned Sawtooth Oscillators and Unison Mode “Fat” synthesizer sounds have always been popular, and are likely to remain so, given their use in modern trance, techno, R n’ B, and other styles. The Analog Saw 3 Osc setting features three detuned oscillators, and sounds fat as it is. The following will introduce you to some additional tools to fatten the sound even more. Â Check out the 3-oscillator basic sound with different filter and envelope settings.
Clean Bass Settings With One Oscillator Only Not every sound needs to comprise of several oscillators. There are numerous simple, effective, sounds which make use of a single oscillator. This is especially true of synthesizer bass sounds, which can be created quickly and easily with the basic “Analog Bass clean” setting. The basic sound is a rectangular wave, transposed down by one octave. The sound is filtered by Filter 2. What’s special about this sound is its combination of Legato and Glide (portamento).
FM Intensity and Frequency The FM Start setting is great for familiarizing yourself with linear Frequency Modulation (FM) synthesis. You’ll 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. IN other words, Oscillator 2 modulates Oscillator 1.
 Following such drastic augmentations to the modulation range, the sound will become uneven across the keyboard. In the lower and middle ranges, it sounds nice, but in the upper key range the FM intensity appears to be too severe. You can compensate for this effect by modulating the Osc 1 Wave target by keyboard position (kybd)—in modulation channels 5 and 6. This results in a keyboard scaling of the FM intensity.
Distorted FM in Monophonic Unison The FM Megafat setting is well-suited for distorted basses and guitar-like sounds. This sound gets rather “rude” in the upper key range. This cannot be compensated for with key scaling, but not every sound has to be “nice” across the entire keyboard range! Â Check out extreme detunings by adjusting the Analog parameter. Â Check out the Flanger with this sound. Â Engage the filter envelope by lowering the Cutoff Frequency of Filter 2 down to 0.
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. Â Choose the PWM Start setting, and move the Wave control slowly back and forth between the rectangular and the pulse wave symbols. Both are green. What you will hear is a (manual) pulse width modulation. Â Choose the PWM Slow setting. Here, LFO 1 controls the pulse width modulation source, not your manual movements.
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.
Oscillator Synchronization If you select the synced square and sawtooth waveforms for Oscillators 2 and 3, they will be synchronized with Oscillator 1. In the Sync Start setting, only Oscillator 2 is audible and Oscillator 3 is switched off. Typical sync sounds feature dynamic frequency sweeps over wide frequency ranges. These frequency modulations (the sweeps) can be applied in various ways. Â Try the pre-programmed pitch modulation, assigned to the modulation wheel, first.
First Steps in Vector Synthesis This tutorial section provides some useful hints for vector envelope programming. In the Vector Start setting, the “mix” of the oscillators is controlled by the vector envelope. Each oscillator has been set to a different waveform. Â Switch from the Router view to Vector view. Â In its basic (default) setting, the vector envelope has 3 envelope points. Point 1 is the start point, Point 2 the sustain point, and Point 3 is the target in the release phase.
Vector Synthesis—XY Pad The vector envelope example starts where the first one left off. You have a simple vector envelope consisting of 4 points, which is set to modulate the oscillator mix (the Triangle). In this example, the vector envelope will be used to control two additional parameters: The Cutoff Frequency of Filter 2 and Panorama. These are pre-set as the X and Y targets in the Square. Both have a value of 0.50.
Vector Synthesis Loops The basic sound of the Vector Loop setting (without the vector envelope) consists of three elements: Â Oscillator 1 delivers a metallic FM spectrum, modulated by Oscillator 2’s wavetable. Â Oscillator 2 outputs cross-faded Digiwaves (a wavetable), modulated by LFO 2. Â Oscillator 3 plays a PWM sound at the well-balanced, and keyboard-scaled, speed of LFO 1. Unison and Analog make the sound fat and wide. These heterogenic sound colors will be used as sound sources for the vector loop.
Bass Drum With Self-Oscillating Filter and Vector Envelope Electronic kick drum sounds are often created with modulated, self-oscillating filters. This approach can also be taken with the ES2, particularly when the vector envelope is used for filter modulation. An advantage of the vector envelope, in comparison with conventional ADSR envelopes, is its ability to define/provide two independent decay phases.
Templates for the ES2 Welcome to a brief programming tour of the ES2! While working on the factory preset programming for the ES2, a number of testers, sound programmers and other people involved in the project indicated that it would be nice to start their programming work from templates, rather than entirely from scratch. Needless to say, creating templates which cover all sound genres is something of a mission impossible.
So set up Envelope 1’s Decay time for this short push, moving the wave selectors for all Oscillators on the attack. (… actually it makes no sense on the synced sawtooth Oscillator, No. 2—it just works this way …) So you can vary the punchiness of the content between: Â Envelope 1’s contribution to the overall attack noise, changing decay speed (a slow one gives you a peak, a long one gives you a growl, as it is reading a couple of waves from the wavetable).
The Big Twirl, Basically (Wheelrocker) This quite ordinary organ patch doesn’t hold any deep, high-end sound design secrets: it is just a combination of three oscillators—with mixed wave levels. You’ll probably find a different combination, which more closely matches your vision of what an organ sounds like. Check out the Digiwaves. Focus your attention on the mod wheel’s response: hold a chord, and bring the wheel in by moving it slowly upwards, until you reach the top (maximum).
You can also bring in LFO 2 to increase the pitch diffusion against LFO 1’s pitch and pan movements. Just exchange it for LFO 1 on modulation 2 and 3—but note that there will be no modulation source for the Leslie acceleration—so you’ll need to use it in a static way, just fading it in. Alternately, you’ll need to sacrifice one of the other modulations in favor of a second twirl.
Oscillator 3 generates a Digiwave, which is “brassy” enough, within the overall wave mix. As an alternative to the Digiwave, another modulated pulse wave could be used to support the ensemble, or another sawtooth wave—to achieve a “fatter” sound, when detuning it with Oscillator 1’s sawtooth wave. The primary aim, however, is to have a little bit of “growl,” achieved through a short wavetable push, as described for the Stratocaster patch, on page 268.
MW-Pad-Creator 3 This is an attempt to create a patch which is able to automatically generate new patches. The Basics Again, Oscillator 2 is used for a pulse width modulation—which creates a strong ensemble component (please refer to “Something Horny (Crescendo Brass)” section, on page 271, for further information). Oscillators 1 and 3 are set to an initial start wave combination within their respective Digiwave tables.
Another Approach to Crybaby (Wheelsyncer) Never obsolete—and undergoing a renaissance in new popular electronic music: Sync Sounds The technical aspects of forcing an Oscillator to sync are described in “Sync” on page 207. Here’s the practical side of the playground. Wheelsyncer is a single-oscillator lead sound, all others are switched off. Although Oscillator 2 is the only one actively making any sound, it is directly dependent on Oscillator 1.
21 EXS24 mkII 21 The EXS24 mkII is a software sampler. This means that instead of having a built-in sound set, it plays back audio files (called samples) that you load into it. These samples are combined into tuned, organized collections of samples called sampler instruments. You can use the EXS24 mkII to play, edit, and create sampler instruments. You can assign the samples (in sampler instruments) to particular key and velocity ranges, and process them with the EXS24 mkII filters and modulators.
The EXS24 mkII interface consists of two windows: Â Parameters window: Offers numerous sample processing and synthesis options, enabling you to tailor EXS instrument sounds to meet your needs. Â Instrument Editor: Used to create and edit sampler instruments.
Using the EXS24 mkII typically involves the following steps: 1 Load or import a sampler instrument. 2 Change the overall sound of the sampler instrument in the EXS24 mkII Parameters window by turning the knobs, pressing switches, and moving sliders. You can also automate these controls, enabling dynamic changes over time. 3 Edit specific samples in the Instrument Editor. Advanced users may also create an instrument from scratch, in which case they will start with this step, then move on to step 2 above.
You can, however, choose to store the current Parameters window settings into a sampler instrument (see “Working With Sampler Instrument Settings” on page 281 for further information). This overrides the settings currently saved in the sampler instrument. The EXS24 mkII is compatible with all audio file formats supported by Logic Express: AIFF, WAV, SDII, CAF. Each audio file is loaded into the EXS24 mkII as a separate sample.
Loading Sampler Instruments The EXS24 mkII ships with a ready-to-play sampler instrument library. To load an instrument: 1 Click the Sampler instrument field directly above the Cutoff knob in the EXS24 mkII Parameters window. This opens the Sampler Instruments menu. 2 Browse to, and select, the desired sampler instrument.
To browse to the next or previous instrument of your sampler instrument library, do one of the following: m Click the plus or minus button to the left and right of the Sampler Instruments menu. m Choose Next Instrument or Previous Instrument in the Sampler Instruments menu (or use the Next EXS Instrument and Previous EXS Instrument key commands).
To copy sampler instruments to your hard drives: 1 Copy the sampler instrument file into the ~/Library/Application Support/Logic/Sampler Instruments folder. 2 Copy the associated samples into a folder named Samples in the same directory as the Sampler Instruments folder. Working With Sampler Instrument Settings Do not mistake plug-in settings, loaded and saved in the plug-in header, with sampler instruments.
Managing Sampler Instruments As your sample library grows, the list of sampler instruments will also expand. To aid you in keeping the list of sampler instruments manageable, the EXS24 mkII features a simple, but sophisticated file management method. To organize your sampler instruments into a preferred hierarchy: 1 Create a folder in the Finder—Basses for example—and drag it into the desired Sampler Instruments folder. 2 Drag the desired EXS24 mkII sampler instruments into this newly created folder.
Searching for Sampler Instruments In order to minimize the number of sampler instruments displayed in the Sampler Instruments menu, you can make use of the Find function. This will limit the Sampler Instruments menu to only display sampler instrument names that contain the search word. To search for sampler instruments: 1 Click the Sampler Instrument field directly above the Cutoff knob in the EXS24 mkII Parameters window, then choose Find in the Sampler Instruments menu.
Importing Sampler Instruments The EXS24 mkII is compatible with the AKAI S1000 and S3000, SampleCell, ReCycle, Gigasampler, DLS, and SoundFont2 sample formats, as well as the Vienna Library. Importing SoundFont2, SampleCell, DLS, and Gigasampler Files The EXS24 mkII automatically recognizes SoundFont2, SampleCell, DLS, and Gigasampler files placed inside the Sampler Instruments folder and converts them into sampler instruments.
The procedure outlined above can also be used to import SoundFont2 and SampleCell Bank files, which contain multiple sounds, in addition to single instrument files. If you load a SoundFont2 or SampleCell Bank into the EXS24 mkII, it creates a Bank and a Samples folder, named after the SoundFont2/SampleCell Bank file. The word Bank or Samples is appended to each folder name. All sounds contained in the bank will automatically have an EXS sampler instrument file created, and placed into the new Bank folder.
Once conversion is complete, the original SoundFont2, SampleCell, or Gigasampler source files can be freely deleted from the hard disks. Note: You can store your imported sampler instruments in any folder on any of your computer’s hard drives. To make sure that these instruments are displayed in the Sampler Instruments menu, you must create an alias pointing to this folder within the ~/Library/Application Support/Logic/Sampler Instruments folder.
Generating a Zone for Each Slice The Extract MIDI Region and Make New Instrument command creates a new EXS24 instrument from a ReCycle file, and generates an independent zone for each slice. To create a new EXS instrument and assign each slice to a zone: 1 Choose Instrument > ReCycle Convert > Extract MIDI Region and Make New Instrument in the Instrument Editor. 2 Browse to, and select the desired ReCycle file in the file selector, then click Open. 3 Enter a velocity factor in the Create MIDI Region window.
Assigning the Complete ReCycle Loop to a Zone The Instrument > ReCycle Convert > Slice Loop and Make New Instrument command creates an EXS instrument from a ReCycle loop, in which each zone plays back the ReCycle loop to its very end (at the current project tempo), starting with the slice points originally assigned to the respective zones. In other words, the lowest zone will play back the entire loop, and the highest zone will only play the last slice of the loop.
Converting AKAI Files The EXS24 mkII can import samples 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) To convert AKAI files: 1 Click the Options button in the Parameters window, then choose AKAI Convert from the menu. This will launch the AKAI Convert window, with the text Waiting for AKAI CD spread across the four columns.
4 To continue through the architecture of the CD-ROM, click on the Volume entries to view any Programs contained therein, and on the Program entries, to view the raw audio files (samples). You can use the Prelisten button below the Audio File column to individually audition AKAI audio files before deciding whether or not to import them. 5 If desired, set any of the additional AKAI Convert parameters at the bottom of the window (see “Additional AKAI Convert Parameters” below).
 Any audio files imported will be stored within a folder which matches the name of the Volume. This folder is created within the ~/Library/Application Support/Logic/ AKAI Samples folder.  The sampler instruments created by the import procedure match the Program names. They are placed inside the ~/Library/Application Support/Logic/Sampler Instruments folder, or the sub-folder determined by the “Save converted instrument file(s) into sub folder” parameter.
Default instrument output volume (head room) Sustained pad sounds and polyphonic instruments in AKAI format often tend to have a higher output than a drum groove, for example. This can result in the output levels of some converted AKAI instruments being much higher than the rest of your EXS24 mkII sampler instrument library—occasionally, converted programs may be so loud that they clip.
Create interleaved stereo files whenever possible This option should always be left enabled, as interleaved files offer better performance within the EXS24 mkII. When converting AKAI format samples, some audio files are created as split stereo and as interleaved stereo files. The detection (of when it is possible to build an interleaved file) is based on information stored with both the AKAI Program and audio files.
Parameters Window The EXS24 mkII Parameters window contains settings that determine how the EXS24 mkII processes the entire loaded sampler instrument.
General Parameters The following section describes the general parameters of the EXS24 mkII. Legato/Mono/Poly Buttons These buttons determine the number of voices used by the EXS24 mkII (in other words, how many notes can be played simultaneously): Â When Poly is selected, the maximum number of voices is set via the numeric field alongside the Poly button. To change the value, drag up or down to increase or decrease polyphony.
Unison Mode In Unison mode, multiple EXS24 mkII voices are played when a key is triggered: Â In Poly mode, two voices per note. Â In Mono or Legato mode, you can adjust the number of voices per note with the Voices parameter. The voices are equally distributed in the panorama field and are symmetrically detuned, dependent on the Random knob value. Note: The number of voices actually used per note increases with the number of layered sample zones.
 Save settings to instrument: Stores the current settings of the parameters window into the instrument file. When the instrument is reloaded, these settings are restored in the parameters window.  Delete settings from instrument: Removes the stored settings from the instrument.  Rename instrument: Opens a file dialog box which allows you to rename the current instrument. This will overwrite the existing instrument name.
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 MIDI controller number 64 (the MIDI standard controller number for hold functions). 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.
Amount refers to the range of velocity values in which the crossfade takes place. In other words, the crossfade is applied symmetrically around each layered zone, with the crossfade amount determining the overlap of the two zones. The Velocity 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 mkII will not fade smoothly between zones but simply switch from one zone to another.
Random This rotary knob controls the amount of random detuning which will apply to every played note. You can use Random (detune) to simulate the tuning drift of analog synthesizers. This parameter can also be effective in emulating a natural feel for some stringed instruments. Fine This parameter lets you tune the loaded sampler instrument in cent increments. You can use this to correct for samples that are slightly out of tune, or to create a thick chorus-like effect.
Glide The effect of this slider depends on the setting of the Pitcher slider: When Pitcher is centered, Glide determines the time it takes for the pitch to slide from one note to another (called portamento). When the Pitcher parameter is set to a value above its centered value, Glide determines the time it takes for the pitch to glide down from this higher value back to its normal value.
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 These parameters control the EXS24 mkII filter section.
Highpass (HP) Click the button under the HP label to engage the highpass filter. 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) Click the button under the BP label to engage the bandpass filter. The bandpass filter is a 2 pole (12 dB/Oct.
Key This knob defines the amount of filter cutoff frequency as determined by note number. When Key is fully turned to the left, the cutoff frequency is not affected by the note number, and is identical for all notes played. When Key is set fully right, the cutoff frequency follows the note number 1:1—if you play one octave higher, Cutoff is also shifted by one octave. This parameter is very useful in avoiding overly filtered high notes.
Volume The Volume knob is the main volume parameter for the EXS24. Move this knob to find the right balance between avoiding distortion and getting the best (highest) resolution in the channel fader and the Level via Vel slider. Key Scale This parameter modulates the sound’s level by note number (position on the keyboard). Negative values increase the level of lower notes. Positive values increase the level of higher notes.
LFO Parameters The EXS24 mkII includes three LFOs (low frequency oscillators) which can be used as modulation sources. This section explains their parameters. LFO 1 EG This knob allows LFO 1 to be faded out (when the knob is pointing inside the Decay area) or faded in (when the knob is pointing in the 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.
LFO 2 Rate The frequency rate of LFO 2 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 3 Rate There is a third LFO available—which always uses a triangular waveform. LFO 3 can oscillate freely between 0 and 35 Hz, or can be tempo synchronized in values between 32 bars and 1/128 triplets.
3 Set the modulation depth with the green triangular fader on the right side of each modulation path. In the example above, the LFO 1 speed is modulated by channel pressure (aftertouch) messages of a MIDI keyboard. You have the option of inserting another modulation source in the middle slot labeled via. The via modulation source doesn’t directly modulate the destination, but instead it modulates the source—in essence, modulating the modulator.
The following example shows an inverted via modulation source. You can see how the green and orange triangles have swapped positions. The orange triangle always marks the modulation depth for the maximum value of the via source, while the green triangle always marks the modulation depth if the via source is at its minimum value. They are reversed by inverting the modulation.
EXS24 mkI Modulation Paths Many of the hard-wired modulation paths that were available as sliders on the original EXS24 (mkI) are part of the modulation matrix. In order to reconstitute the modulation slider configuration of the mkI version, click on the “options” button in the upper-right corner and choose (Recall default EXS24 mkI settings) from the pop-up menu.
Modulation Destinations:  Sample Select  Sample Start  Glide Time  Pitch  Filter Drive, Filter Cutoff, and Filter Resonance  Volume  Pan  Relative Volume  LFO 1 Dcy./Dly (LFO 1 Decay/Delay)  LFO 1 Speed, LFO 2 Speed, LFO 3 Speed  Env 1 Attack, Env 1 Decay, Env 1 Release  Env 2 Attack, Env 2 Decay, Env 2 Release  Time  Hold Note: Controllers 7 and 10 are marked as (not available). Logic Express uses these controllers for volume and pan automation for the audio channel strips.
If you choose a continuous controller such as the modulation wheel to modulate the Sample Select destination, you can step through the velocity layers during playback. In this case, the crossfade (XFade) parameters are very important for creating smooth transitions between velocity split points. The Instrument Editor You can use the Instrument Editor of the EXS24 mkII to create and edit sampler instruments.
The screenshot below shows the Instrument Editor in Zones view. Parameters area Click to switch between Zones and Groups views Zone column Velocity area Zones/Groups area Keyboard  Zones column: Displays all zone groups of the instrument. The All Zones and Ungrouped Zones groups exist by default in every instrument. Click the desired group to display the associated zones in the Parameters area. You can also select and view multiple zone groups at once (but only if the All Zones group is not selected).
Click the Groups button in the upper left corner to switch to Groups view. Click the Zones button in the upper left corner to switch to Zones view. You can also use the Toggle Zones/Groups View key command to switch between views. Instrument Editor in Groups view The EXS24 button in the upper right of the Instrument Editor reopens a closed EXS24 mkII Parameters window. The button does not bring the Parameters window to the foreground if covered by other floating windows.
To create a zone and assign a sample to it: 1 Choose Zone > New Zone (or use the New Zone key command). A new zone entry appears in the Instrument Editor. 2 Do one of the following: Â Click the arrow in the Audio File column, then choose Load Audio Sample from the pop-up menu (or you can use the Load Audio Sample key command). Â Double-click the empty area in the audio file column. 3 Browse to, and select the desired audio file in the file selection dialog.
Creating Zones via Drag and Drop You can also create a new zone (and a new instrument, if none is currently displayed in the editor) by dragging a file onto one of the keys of the onscreen keyboard. The start key, end key, and root key are all set to the note that the file was dropped on. This drag and drop functionality works for audio files from the following sources: Browser, Audio Bin, and the Finder.
To load multiple samples in one operation: 1 Choose Zone > Load Multiple Samples in the Instrument Editor (or use the Load Multiple Samples key command). 2 Browse to the desired location, then use the Add, Add All, Remove, or Remove All buttons to select the desired samples. 3 Click the Done button when you are finished.
Groups Imagine a drum kit has been created, with a number of different samples being used in several zones, mapped across the keyboard. In many musical circumstances, you might want to adjust the sound editing parameters of each of the samples independently— to alter the decay of the snare, or to use a different cutoff setting for the hi-hat samples, for example. This scenario is where the EXS24 mkII’s groups feature comes in. Groups allow for very flexible organization of samples.
To assign a zone to a group, do one of the following: m Select the group in the zone’s Group menu. m Select a zone in the EXS Instrument Editor, Finder, Audio Bin, or Browser and drag it into a group displayed in the Zones column. m Drag a zone (or multiple selection of zones) out of one group into another group to change the group assignment to a new group. m Drag a zone (or multiple selection of zones) out of one group to Ungrouped Zones to change the group assignment to unassigned.
Selecting Zones and Groups There are a number of ways that you can select zones and groups for editing. First of all, the Edit menu also contains commands specifically relating to the selection of EXS24 mkII zones and groups: Â Select All: Selects all zones and groups of the loaded sampler instrument. Â Toggle Selection (also available as key command): Toggles the selection between the currently selected zones or groups and all currently unselected zones or groups.
Audio File This column displays a zone’s audio file. Mousing over an audio file reveals a help tag with additional information about the audio file, such as format, bit depth, sample rate, and so on. If you press Command before the help tag displays, the help tag will also include the full file path of the audio file. Clicking the arrow button opens a shortcut menu that offers the following commands: Â Load Audio Sample: Opens a file selection box in which you can select an audio file.
Output Parameters Volume adjusts the volume of the zone. Pan adjusts the pan position of the zone. This parameter only works when the EXS24 mkII is used in stereo. Negative Scale values make notes lower than the note position defined by the root key sound louder than higher ones; positive values have the opposite effect. Use this parameter for balancing the volume of a sample across the selected key range. Root key The Routing parameter determines the outputs used by the zone.
Loop Parameters The EXS24 mkII can loop playback of either all or a portion of an audio sample, when you trigger a sample to play. In other words, the sample will loop when sustained MIDI notes are received. Â Loop On checkbox: Activate this checkbox and the rest of the Loop parameters become available for editing. Â Loop Start, Loop End: You can define discrete loop start and end points in these fields, allowing you to loop a portion of the audio file.
Editing Samples in the Sample Editor The most intuitive way to adjust sample and loop start and end points is to have a visual representation of the waveform, which you can use as a guide to graphically move the points to where you want them to be on the waveform. This allows you to get much closer to the desired values much quicker than having to type in numerical values without any visual guide. The Instrument Editor does not offer a graphic representation of a sample’s waveform.
To use the Sample Editor Loop commands: 1 Choose either of the selection commands in the Edit menu of the Sample Editor: Â Sample Loop → Selection: The loop area (defined by the Loop Start and End points) is used to select a portion of the overall audio file. Â Selection → Sample Loop: The selected area is used to set the Loop Start and End points. 2 Once you’ve selected the desired area using either of the commands above, choose Edit > Write Sample Loop to Audio File.
Output Parameters  Volume: Adjusts the volume of the group, and therefore the volume of all assigned zones, simultaneously. This works much like a sub-group volume slider on a mixing console.  Pan: Adjusts the pan position of the group (stereo balance for stereo samples), and the pan position of all assigned zones simultaneously.  Routing: Determines the outputs used by the group.
Envelope 1 and Envelope 2 Offsets Parameter Use these parameters to offset the envelope settings from the Parameters window separately for each group. This is useful if you want the filter or volume envelopes to affect the samples in a group after the initial impact of the triggered sounds. Note: When the Trigger parameter is set to Key Release, the Decay Time parameter determines the volume decay, not Envelope 2 (the volume envelope).
Remapping of Pitch Bend and Modulation Wheel Events In order to create realistic-sounding performances in an easy and intuitive way, the Jam Pack 4 (Symphony Orchestra) instruments use the modulation wheel to switch between articulations (legato, staccato, and so on) and the pitch bend wheel to change expression (crescendo, diminuendo, and so on). Further information about this can be found in the Jam Pack 4 documentation.
Editing Zones and Groups Graphically You are not limited to editing zones and groups in the Parameters area. You can also edit some zone and group parameters graphically in the Zones Display or Groups Display area above the Instrument Editor keyboard. To move a zone or group: 1 Move the mouse cursor over an existing zone or group. It will change to a two-headed arrow. 2 Click-hold and drag the zone or group to the desired position.
To change the start or end note of a zone or group: 1 Move the mouse cursor to the beginning or end of a zone or group. The cursor will change to a two header arrow. 2 Click-hold and drag the start or end point of the zone or group to the desired position.
To edit the velocity range of a zone or group: 1 Click the Show Velocity button at the top right of the Instrument Editor (or use the Show/Hide Velocity key command). The Velocity Display area will open above the Zones or Views Display area. 2 Click on one or more zones or groups in the Display area. The velocity bars of the selected zones will be highlighted in the Velocity Display area. 3 Click-hold on either the High or Low value of the velocity bar of the zone or group you wish to edit.
Sorting Zones and Groups You can easily sort zones and groups in the EXS24 mkII Instrument Editor by clicking the sub-column heading that you wish to sort by. For example, if you want to sort your zones by name, click the Name sub-column heading under the Name column, and your zones will be sorted alphabetically by name.
Saving, Deleting, Renaming, and Exporting Instruments You can access all basic sampler instrument operations in the Instrument Editor’s Instrument menu. Save Saves the currently loaded sampler instrument. When you create a new instrument and save it for the first time, you will be asked to give it a name. If you have edited an existing sampler instrument and save it via this command, the existing file name is used and the old instrument is overwritten. You can also use the Save Instrument key command.
Setting Sampler Preferences The EXS24 mkII offers a separate Sampler Preferences window, allowing you to configure various operational and sample related preferences, such as sample rate conversion quality, velocity responsiveness, sample storage, search-related parameters, and so on. To open the Sampler Preferences window, do one of the following: m Click the Options button in the Parameters window, then choose Preferences in the pop-up menu. m Choose Edit > Preferences in the Instrument Editor.
Search Samples On Menu 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. Drives can be selected individually, or grouped as follows: Â Local Volumes internal storage media (hard disks and CD ROM mechanisms) attached to or installed in the computer directly. Â External Volumes storage media accessible 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).
Configuring Virtual Memory These days, many sample libraries contain many gigabytes of audio samples in order to create the most accurate sampler instruments possible. Often, these gigantic sample libraries are too large to fit into your computer hardware’s random access memory (RAM) all at once. To let you use these huge sample libraries, the EXS24 mkII can use a portion of your hard drive as virtual memory.
 Performance section: Shows the current disk I/O traffic and the data not read from disk in time. If these numbers start rising, the EXS24 mkII may glitch when trying to stream your samples from the disk in time with your performance. If you notice these values rising to high levels, you should change the general settings to free up additional RAM for virtual memory use.
22 External Instrument 22 You can use the External Instrument to route your external MIDI sound generators through the Logic Express Mixer, allowing you to process them with Logic Express effects. Ideally, you will use a multi input and output audio interface, to avoid constant repatching of devices. The External Instrument can be inserted in software instrument channels in place of a software instrument.
Using the External Instrument The following section outlines the steps required to route external MIDI sound generators through the Logic Express Mixer. To process external MIDI instruments with effects: 1 Connect the output (or output pair) of your MIDI module with an input (pair) on your audio interface. Note: These can be either analog or digital connections if your audio interface and effects unit are equipped with either, or both. 2 Create an instrument channel.
23 Klopfgeist 23 Klopfgeist is an instrument that is optimized to provide a metronome click in Logic Express. Klopfgeist is inserted on instrument channel 128 by default, and used to generate the MIDI metronome click. Theoretically, any other Logic Express or third-party instrument could be used as a metronome sound source on instrument channel 128. Similarly, Klopfgeist can be inserted on any other instrument channel for use as an instrument.
 Damp slider and field: Controls the release time. The shortest release time is reached when Damp is at its maximum (1.00) value.  Level Via Vel slider and fields: Determine the velocity sensitivity of Klopfgeist. The upper half of the two-part 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.
24 Ultrabeat 24 Ultrabeat is a rhythm synthesizer that incorporates a step sequencer. Ultrabeat’s synthesis engine is optimized for creating electronic and acoustic drum and percussion sounds. It includes an impressive diversity of synthesis engines: phase oscillators, sample playback, FM (frequency modulation), and physical modeling.
The Structure of Ultrabeat Most software synthesizers offer one synthesizer per plug-in instance. Ultrabeat, however, places 25 independent synthesizers at your disposal. These synthesizers— called drum voices in Ultrabeat—are optimized for the generation of drum and percussion 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.
Overview of Ultrabeat Ultrabeat’s user interface is divided into three functional sections. Synthesizer section Assignment section Step sequencer  Assignment section: Displays all drum sounds of a drum kit, allowing you to select, rename, and organize them. It also includes a small mixer, which you can use to adjust each sound’s volume and pan position.
Loading and Saving Sounds You can use the same methods to save and load settings in Ultrabeat as in all other Logic Express instruments. For more information, see the Logic Express 8 User Manual. An Ultrabeat setting contains: Â The drum kit, which consists of 25 sounds, inclusive of assignment and mixer settings. Â The complete settings of all parameters for all 25 sounds. Â The sequencer settings and all 24 patterns, including the step automation, trigger, velocity, and gate rows for all 25 sounds.
The Assignment Section The Assignment section displays all drum sounds of a drum kit, and also includes a small mixer. It allows you to:  Select, organize, and name drum sounds  Import drum sounds from other Ultrabeat settings or EXS instruments  Mix drum sounds Selecting Sounds The 25 sounds of an Ultrabeat drum kit are mapped to the onscreen keyboard found on the left hand side of the Ultrabeat window.
Naming and Organizing Sounds Double-clicking on the name of a drum sounds opens its text entry field, allowing you to rename it. Press Return or click anywhere outside the text entry field to complete the naming operation. Swapping and copying drum sounds within an Ultrabeat kit can be achieved via a drag and drop operation or via a shortcut menu. To swap or copy drum sounds using drag and drop: 1 Click-hold the drum sound in the Assignment section (not on a button or menu).
To swap or copy drum sounds via a shortcut menu: 1 Control-click or right-click the drum sound in the Assignment section. 2 Choose the desired command from the shortcut menu that appears: Â Copy (Voice & Seq): This command copies the selected sound—inclusive of Mixer settings and sequences—to the Clipboard. Â Paste Voice: This command replaces the selected sound with the sound from the Clipboard, without changing its sequences.
Importing Drum Sounds and EXS Instruments You can import drum sounds and sequences from other Ultrabeat settings into your currently active drum kit. You can also import sounds from EXS instruments: Ultrabeat reproduces the EXS layout as closely as possible. Layered EXS zones are set up as layered drum sounds, using the sample playback mode of Osc 2. To open an Ultrabeat setting or EXS instrument in the import list: 1 Click on the Import field in Ultrabeat’s upper left corner.
There are two methods to transfer sounds from the import list to the Assignment section. The simplest method is to drag sounds from the import list to the Mixer section. Holding Command while doing so includes all sequences. You can also use a shortcut menu. To transfer sounds via a shortcut menu: 1 Control-click (or right-click) the sound’s name in the import list. 2 Choose Copy (Voice & Seq) in the ensuing shortcut menu. This copies the selected sound and its sequences to the Clipboard.
The 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. Master volume Pan knob Output Selection menu Volume slider Mute button Solo button Volume The individual volumes of all sounds are indicated by blue bars, providing a complete overview of all levels within the kit.
The Synthesizer Section The Synthesizer section is the heart and soul of Ultrabeat. As noted above, each drum sound has its own Synthesizer section. Don’t be intimidated by how many parameters the Ultrabeat synthesizer squeezes into one plug-in window; in fact, its signal flow is quite easy to understand. The Signal Flow Ultrabeat’s synthesis engine is based on classic subtractive synthesis principles.
The filter receives its signal from the following sound sources: Oscillator 1, Oscillator 2, the noise generator and the ring modulator. Their output sections are displayed by four objects that sit adjacent to the filter (three round objects and the smaller, rectangular ring modulator to the right of the filter). One level down, you’ll find the control elements for these sound sources.
The pitch value is displayed 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 also be modulated by the sources found in the Mod and Via menus. You’ll find a signal flow switch between each of the oscillators and the Filter section that controls routing (filter bypass button).
 Saturation: Increasing Saturation values clip the waveform, gradually molding its shape towards a rectangular waveform. This results in a corresponding increase in odd numbered overtones.  Asym (Asymmetry): Tilts the waveform towards a sawtooth wave, making the sound more edgy. Asym 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. For more information, see “Modulation” on page 373.
Side Chain In Side Chain mode, Ultrabeat uses an external side chain input as its source for Oscillator 1. This means that you send any audio channel, aux, or live input channel through Ultrabeat’s filters, envelopes, LFO, and step sequencer. Once you select Side Chain mode for Oscillator 1, you need to select which channel you wish to use as your side chain input. You do this in Ultrabeat as you would with any plug-in with side chain input by choosing the desired channel in the Side Chain menu.
Unique Properties of Oscillator 2 Oscillator 2 can be switched between three different types of synthesis engines: Phase Oscillator, Sample, and Model. This can be done by clicking the appropriately labeled buttons at the lower edge of the Oscillator 2 section. Phase Oscillator Oscillator 2’s phase oscillator operates nearly identically to the phase oscillator of Oscillator 1.
To load an audio file: 1 Click the arrow in the upper left corner of the waveform display, then choose Load Sample from the pop-up menu. 2 In the Load Sample window, browse to, and select the desired audio file. A selection of multi-layer drum and percussion samples that were specially created for Ultrabeat and its function set are included with Ultrabeat. You can also load your own samples in AIFF, WAV, CAF, or SDII stereo interleaved format.
The Preview Sample in Ultrabeat Voice option temporarily replaces the sample files in the currently selected drum sound. The drum sound is not directly triggered by activating this option, but it can as usually be triggered via MIDI notes (played notes, MIDI region events, or Ultrabeat sequencer events) while the Load Sample window is open and different files are being selected. The selected sample can be heard as part of the current drum sound, inclusive of all synthesizer processing.
Stiffness controls the stiffness or rigidity of the string. In the real world, this depends on the material the strings are made of and their diameter (or, more precisely: their speed of vibration or response to being struck or plucked and so on). Rigid strings create an inharmonic vibration where the overtones do not represent whole number multiples of the fundamental frequency. These overtones are, in fact, slightly higher.
The Ring Modulator The ring modulator functions as its own sound source; its signal can bypass or be sent into the filter, independent of Oscillators 1 and 2. Its volume can also be regulated. Please note that both oscillators need to be switched on to use it. The sound of the ring modulator is largely dependent on both of the oscillators, as it modulates the output signals of both.
The Noise Generator The fourth synth engine is the noise generator. Noise contains—in a technical sense—all tonal frequencies; that’s why human hearing can’t recognize any tonality in a noise signal. Despite this (or as a direct result of it), noise is an indispensable ingredient when creating drum sounds. For this reason, Ultrabeat’s noise generator is outfitted with extensive features. To use the noise generator, you first need to turn it on. This can be done with the On/ Off button.
The Cut knob determines the cutoff frequency, defining 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. Cutoff can be modulated by the sources found in the Mod and Via menus. For more information, see “Modulation” on page 373. Increasing Resonance boosts frequencies that surround the cutoff frequency.
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. Multimode filter Distortion unit The order that sounds are passed through the filter and distortion unit is determined by the arrow found at the “equator” of the Filter section.
The names of the individual filters illustrate their function: A lowpass (LP) filter allows frequencies lower than the cutoff frequency to pass. It removes (cuts) the highs of a sound, making it darker and less bright. A highpass (HP) filter allows frequencies higher than the cutoff frequency to pass. The lows of the sound are cut. A bandpass (BP) filter allows a frequency band centered around the cutoff frequency to pass.
The Distortion Unit Depending on the order determined by the 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. If neither button is red the distortion unit is bypassed The distortion effect is modeled on an analog distortion unit, which distorts the sound by overdriving the level.
Output Section Depending on the status of each filter bypass button, the output signals of both oscillators, the ring modulator and the noise generator are routed either; directly or via the Filter section to the Output section of Ultrabeat. The Output section passes signals through both equalizers (EQ) and the Pan Modulation and Stereo Spread section (in a pre-configured order) before the final level is set, and the trigger behavior (of the signals) is adjusted.
The EQ Gain knob is bipolar. Positive values boost a certain frequency range as determined by the EQ type and Hz settings. Negative gain values lower the gain of the frequency range. If the Gain knob set to the mean value of 0, the EQ has no effect. Note: You can also return this knob to its neutral position by Option-clicking on it. Alternately, you can click on the tiny 0 above the EQ Gain knob. The frequency is set by click-dragging vertically on the Hz parameter field.
Pan Modulation and Stereo Spread The EQ’s output signal is passed along to the Pan Modulation and Stereo Spread section. In the Pan Modulation and Stereo 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).
Stereo Spread Stereo Spread broadens the stereo image, making it wider and more spacious. Low Frequency applies the (spreading) effect to the bass frequencies: the higher the value, the more prominent the effect becomes. Hi Frequency allows you to apply the effect to the high frequencies. Voice Volume This rotary knob adjusts the output volume of the individual drum sounds.
Clicking the button below the Group label opens the Group menu, allowing a choice between the Off and group 1 to 8 settings. If two different sounds are assigned to the same group, they will cut each other off. A typical use of this facility is when you’re programming hi-hat sounds: when playing a real hi-hat, the closed hi-hat note cuts off and mutes the ringing of the open hi-hat. This function is often referred to as hi-hat group mode.
Modulation Numerous sound parameters can be controlled dynamically (modulated) in Ultrabeat. Ultrabeat provides two powerful LFOs, four envelope generators, velocity, and four freely-definable MIDI controllers as modulation sources. The setting of modulation routings follows a universal principle that is explained in this chapter.
The Cut (Cutoff ) parameter has a mean (default) value of 0.50. It’s not being modulated yet as no modulation source has been selected in either the red Mod or blue Via menu (set to Off ). As soon as a modulation source is selected in the Mod menu (Env 1 in this example), the ring around the rotary knob is activated. Grabbing and moving this ring with the mouse allows you to set the value that this parameter will be increased to by the Mod source (0.70 in the example).
Back to the example; the frequency of the filter is set to the mean value of 0.50. When the Mod source Env 1 enters the equation, the Env 1 envelope generator drives the Cut value up from 0.50 to 0.70 (during the attack phase) and back down to 0.50 (during the decay phase). Note: You can view the exact values in the help tags that appear when you grab the individual handles of various parameters.
Setting the Modulation Routing Clicking on the Mod label opens the Mod menu. This is where you can choose one of the LFOs or envelope generators (Env) as a modulation source. The Off setting deactivates the Mod routing, and the Mod ring can no longer be adjusted. In this situation, no Via modulation can occur either (this is because Via no longer has a modulation target) and the Via slider disappears. Note: The Max setting produces a static modulation at maximum level.
MIDI Controllers A–D In the MIDI Controller Assignments area at the upper edge of the Ultrabeat window you can assign a standard MIDI controller to each of the four controller slots: Ctrl A, B, C, or D. Ctrl A, B, C, and D can be used as Via modulation sources within Ultrabeat. Use these assignments to set up your external MIDI controller hardware to operate with Ultrabeat. As examples: aftertouch or the modulation wheel of your MIDI keyboard.
The LFO section display shows the LFO waveform, the shape of which is governed by the Shape slider located underneath it. Dragging the slider from left to right causes the waveform to fluidly morph from a sine to a triangle, and then finally to a square wave (with variable pulse width), including all variations in-between. At the far right hand position of the Shape slider, the LFO produces random waveforms.
Env 1 to 4 Further modulation sources available to you in the Mod menu include four identically specified envelope generators. Envelope parameters are described in this section. Note: In addition to its potential use in the Mod menus of various sound parameters, Env 4 is permanently connected to the Voice Volume. In other words, Ultrabeat has a hard-wired volume envelope generator.
Envelope Parameters In order to edit the envelope parameters, first select an envelope by clicking on the desired 1 to 4 buttons. The parameters of the corresponding envelope can now be directly changed in the envelope display window. Attack Time Attack time defines the period of time the envelope needs to reach its maximum value. This is measured from the instant you press a key (note on). This period is called the attack phase.
Note: If the Sustain button is not activated, the envelope functions in one shot mode, and the note length (MIDI note off command) is disregarded. Zoom (to fit) When you select the Zoom button, the envelope is enlarged to fill the entire width of the display, making it easier to adjust junction points and curves. The graphic display is quickly redrawn after any change is made to the Attack or Decay values.
The Step Sequencer The integrated step sequencer allows all Ultrabeat sounds to be combined in sequences, based on patterns. Its design and use (step programming input) are based on analog predecessors.
Step Sequencing With Ultrabeat Ultrabeat’s step sequencer contains 24 sequences—each consisting of up to 32 steps. The sequencer is divided into three sections. Global parameters Pattern parameters Pattern parameters Step Grid  Global parameters: These parameters globally control the pattern and sounds, independent of the individual steps and patterns.  Pattern parameters: Control the currently selected pattern.  Step Grid: Here, the actual sequencing takes place.
 Step mode: In Step mode, you can automate a sound’s parameter from one step to the next. Values are offset, instead of set—all of your original drum sound settings remain unaltered by any parameter changes performed in Step mode. Step automation changes only operate on parameters when the sequencer is running. These parameter changes occur individually per step. This means that if the sequencer is turned off, you’ll hear the original sound. For more information see “Step Mode” on page 390.
Resolution This parameter determines the resolution of the pattern. It defines the metric unit of a measure that is represented by the individual steps. As an example: The 1/8 setting means that each step of the grid represents an eighth note. Given a pattern length of 32 steps, the pattern would run for 4 measures (32 ÷ 8). The Resolution setting applies to the entire grid, and therefore, equally to all sounds.
Step Grid In the step grid, the pattern is displayed in numerous rows and steps. The rows always correspond to the sound that is currently selected in the assignment area. Choosing a different sound switches the sequencer display to show the rows that correspond to the newly-selected sound. The step grid area contains two rows—each with 32 fields. Trigger row Velocity/Gate row  Trigger row: Click a button to activate or deactivate the sound on that corresponding beat.
 Add Every Downbeat: Adds triggers on every downbeat until the sequence is filled. The exact determination of which steps are downbeats depends on the grid resolution. For example, if the resolution is set to 1/16, Add Every Downbeat would create triggers on every 4th step. Starting with the initial downbeat at step 1, this would create trigger events on step 5, step 9, step 13, and so on. This command does not erase existing trigger events, it only adds trigger events.
 Create & Replace Some: Same as Create & Replace Randomly above, however it creates less trigger events. How many events are created depends on the grid resolution.  Create & Replace Many: Same as Create & Replace Randomly above, however it creates more trigger events, filling the sequencer with events. Velocity/Gate Row In this row, you set the length (gate time) and the velocity of the notes entered in the trigger row. Both parameters are displayed as a single graphical bar.
Switching the Step Grid to Full View Clicking the Full View button in Ultrabeat’s lower right corner switches the synthesizer controls to a large grid filled with trigger buttons. The large grid displays the 32 trigger buttons for each of the 25 drum sounds simultaneously—independent of the currently selected sound. As usual the selected sound is visible in the step sequencer area so that you can set the velocity and gate time for each step as well as offsets in Step mode.
Step Mode Setting the Edit Mode switch to Step mode engages Ultrabeat’s step automation feature. Step automation lets you program parameter changes per step for each drum sound. You can automate as many of the parameters available for automation per step as you want.
The Parameter Offset Row In this row you can view and enter offset values per step for any parameter in the Synthesizer section framed in yellow. Editing parameters themselves is done, as before, using the controls in the Synthesizer section. In addition, it is however possible to edit offset values directly in the offset row. Note: These have an effect only in relation to the actual parameter value.
The Parameter Menu of the Parameter Offset Row All parameters that have been changed in Step mode are automatically added to the Parameter menu of the offset row. You can also select other parameters in the Parameter menu to display their recorded offset values: You can change these values in two different ways: Â By simply drawing with the mouse you can change or add parameter offset values.
Copying and Reorganizing Patterns You can reorganize the 24 patterns of a sound in the Pattern menu, using a copy and paste operation. To copy a pattern using a shortcut menu: 1 Choose the desired pattern in the Pattern menu. 2 Control-click (or right-click) the Pattern menu, then choose Copy in the shortcut menu. 3 Choose the target pattern in the Pattern menu. 4 Control-click the Pattern menu, then choose Paste in the shortcut menu. You can also use a key command short cut to copy patterns.
Using MIDI to Control the Sequencer Pattern performance can be influenced by incoming MIDI notes. This allows you to spontaneously interact with the step sequencer, making Ultrabeat an excellent live performance instrument. The manner in which Ultrabeat reacts to MIDI control is determined by the pattern, playback, and Voice Mute mode. Pattern Mode If activated, you can switch and start patterns via incoming MIDI note commands.
Voice Mute Mode When you activate Voice Mute mode, playing a MIDI note starting at C1 and upwards mutes the corresponding sound in Ultrabeat’s mixer. A subsequent MIDI note of the same pitch un-mutes it. This is ideal for spontaneous arranging of pre-programmed patterns, and muting single elements of a pattern without deleting them. This is especially useful in a live performance—but not only there. Triggering the step sequencer via MIDI notes opens up a number of remixing possibilities.
Creating Drum Sounds in Ultrabeat The following section covers a few specific sound creation tips. Please take the time to explore the vast and complex possibilities 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. Note: In Ultrabeat’s Settings > Factory > Tutorial Settings folder, you will find a drum kit called Tutorial Kit.
To give the bass drum more kick by controlling the pitch with an envelope: 1 Ensure that Env 1 is chosen in the Mod menu of Oscillator 1’s Pitch parameter. 2 Set the degree of modulation by moving the blue Mod slider approximately 3 to 4 octaves above the original pitch. 3 Set the attack time in Env 1 to zero by sliding the leftmost of the two junction points that sit on the x-axis all the way to the left.
To reduce tonality using the 2 Band EQ: 1 For band 1, choose the Shelving mode at a frequency of about 80 Hz, a high Q value, and a negative Gain value. 2 For band 2, choose the Peak mode at a frequency of around 180 Hz, a medium Q value and also a negative Gain value. On the EQ graph, you can see how the frequencies around 80 Hz are boosted, while the surrounding frequencies are reduced.
4 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. 5 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). Choose a Mod amount for Res of about 0.80.
The Ultrabeat Kick You can use the extensive Ultrabeat features to create bass drums that are uniquely “Ultrabeat.” Try modulating pitch with an LFO, rather than an envelope, for example. To create an LFO modulated kick drum: 1 Start with the Standard Tutorial sound at a pitch of A#0 (Osc 1 Pitch), and choose LFO 1 as the Mod source in the Osc 1 Pitch section. 2 Set the degree of modulation by moving the blue Mod control to a value of A3.
Creating Snare Drums 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. To create a basic snare drum: 1 Load the Standard Tutorial setting. Deactivate Oscillator 1, and switch Oscillator 2 on (in phase oscillator mode).
To 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 by adding 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.
To increase the performance dynamics: 1 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: 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.
4 Set the additional control that appears as shown below, to control the character of the sound with velocity: 5 Repeat this with the other parameters of Oscillator 2, as well as pitch: 6 Modulate the noise generator as follows: Â Cut parameter: Choose Max as modulation source, then set the modulation control as shown below. Â Dirt parameter: Choose LFO 2 as modulation source, then set the modulation control as shown below.
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. To recreate the Kraftwerk snare sound with Ultrabeat: 1 Select the Snare 1 sound. 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). 4 Modulate the filter resonance with Env 2.
Creating Hi-Hats and Cymbals Electronic hi-hat sounds are very easy to create in Ultrabeat. To create a hi-hat in Ultrabeat: 1 Load the Standard Tutorial sound. 2 Switch off Oscillator 1 and turn on the noise generator. 3 Choose the following settings for the noise generator: In the screenshot 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.
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: 1 Load the Standard 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, realizing 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.
25 GarageBand Instruments 25 GarageBand Instruments are automatically installed with Logic Express. You can insert them as per other software 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 Express a trouble-free experience. GarageBand Instruments are actually less CPU and memory-intensive versions of equivalent Logic Express or Logic Pro instrument plug-ins.
This has two main benefits: Â As the GarageBand instrument plug-ins are less CPU and memory-intensive, they load faster than the equivalent Logic Express software instruments. Â Limitation to a few, powerful, parameters simplifies the use of the instruments. Play around with the parameters to see how easily spectacular sounds can be created! The macro parameter sliders available to each GarageBand Instrument are different.
Analog Mono This is a monophonic (one note can be played at a time) analog synthesizer lead sound. Unique macro parameters are: Â Glide: Determines the time it takes a note to change (slide) to another. Â Richness: Determines the complexity of the sound texture, making the sound fuller. Analog Pad The Analog Pad is based on the ES2. This is a warm analog synthesizer pad that is useful for a range of musical styles.
Digital Basic The Digital Basic instrument is based on the ES2. This is a basic digital synthesizer sound that is useful for a range of musical styles. Unique parameters are: Â Harmonics: Changes the sound dramatically as more harmonics (overtones) are added. The impact of this parameter is difficult to describe, so please experiment with it. Â Timbre: Changes the color of the sound from dark to bright. Digital Mono The Digital Mono instrument is based on the ES2.
Electric Clavinet The Electric Clavinet sound is based on the EVD6. It emulates the Hohner D6 clavinet. It offers the following unique parameter: Â Damper: Changes the tone of the clavinet, making it less sustained, and more woody sounding as you move towards the high setting. Electric Piano The Electric Piano sound is based on the EVP88. It sounds like the Fender Rhodes electric piano. Macro parameters are: Â Model: A more bell like tone is achieved when the Tines button is selected.
Piano, Sound Effects, and Strings The Piano, Sound Effects, and Strings sounds are sample-based. As with other GarageBand instruments, the Settings menu offers several variations. Tonewheel Organ The Tonewheel Organ sound is based on the EVB3. It emulates the Hammond B3 organ. Please try out the various Settings available, as this instrument is capable of generating a wide array of organ tones. Unique parameters are: Â Drawbars: Makes the sound a little thicker (more) or thinner (less).
Appendix Synthesizer Basics 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.
Subtractive Synthesis Subtractive synthesis is synthesis using filters. All analog and virtual analog synthesizers use subtractive synthesis to generate sound. In analog synthesizers, the audio signal of each voice is generated by the oscillator. The oscillator generates an alternating current, using a selection of waveforms which contain differing amounts of (more or fewer) harmonics.
The picture below shows a sawtooth wave with the filter half closed (24 dB/Fat). The effect of the filter is somewhat like a graphic equalizer, with a fader set to a given cutoff frequency (the highest frequency being fed through) pulled all the way down (full rejection), so that the highs are damped. With this setting, the edges of the sawtooth wave are rounded, making it resemble a sine wave. The wave length here is not really higher, but the zoom setting is.
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 the Sample Editor of Logic Express). 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.
Envelopes What does the term envelope mean in this context? In the image, you see an oscillogram of a percussive tone. It’s easy to see how the level rises immediately the top of its range, and how it decays. If you drew a line surrounding the upper half of the oscillogram, you could call it the envelope of the sound—a graphic displaying the level as a function of time. It’s the job of the envelope generator to set the shape of the envelope.
Glossary Glossary AAC Abbreviation for Advanced Audio Codec. A compression and decompression algorithm and file format for audio data. AAF Abbreviation for Advanced Authoring Format. A cross-platform project exchange file format that you can use to import multiple audio tracks, inclusive of references to tracks, time positions, and volume automation. accelerando A gradual increase in tempo (see tempo).
ALAC Abbreviation for Apple Lossless Audio Codec, an encoding/decoding algorithm that delivers lossless audio compression. alias A pointer to a MIDI region in the Arrange area. An alias does not contain any data. It simply points to the data of the original MIDI region. You can create an alias by Shift-Option-dragging the original MIDI region to a new location. An alias can not be edited directly. Any change to the original region will be reflected in the alias.
audio file Any digital recording of sound, stored on your hard drive. You can store audio files in the AIFF, WAV, Sound Designer II (SDII), and CAF formats in Logic Express. All recorded and bounced WAV files are in Broadcast Wave format. audio interface Device used to get sound into and out of your computer. An audio interface converts digital audio data, sent from your computer, into analog signals that speakers can broadcast.
bit depth The number of bits used by a digital recording or digital device. The number of bits in each sample determines the (theoretical) maximum dynamic range of the audio data, regardless of sample rate. bit rate Bit rate, when talking about MP3 files, refers to the transfer bit rate at which the files are encoded. Conversationally, the term is more often used to describe the relative quality of the file, with lower bit rates resulting in less defined audio. bit resolution Alternative term for bit depth.
chorus effect Effect achieved by layering two identical sounds with a delay, and slightly modulating the delay time of one, or both, of the sounds. This makes the audio signal routed through the effect sound thicker and richer, giving the illusion of multiple voices. click Metronome, or metronome sound. Clipboard The Clipboard is an invisible area of memory, into which you cut or copy selected data, using the Edit menu. Data stored in the Clipboard can be pasted to different positions.
cueing Monitoring (hearing playback) while fast-forwarding or rewinding. cutoff frequency Frequency at which the audio signal passing through a low or highpass filter is attenuated by 3 dB. cutting The act of reducing a level, or frequency, when using EQ or other filters. Also used to describe physically dividing and removing sections of files, regions, and so on (see boosting and attenuation).
Digital Full Scale See DFS. disclosure triangle A small triangle you click to show or hide details in the user interface. distortion The effect that occurs when the limit of what can be accurately reproduced in a digital signal is surpassed, resulting in a sharp, crackling sound. drag & drop Grabbing objects with the mouse, moving them, and releasing the mouse button. driver Drivers are software programs that enable various pieces of hardware and software to be recognized by computer applications.
event Individual MIDI command, such as a note on command. Continuous controller movements (modulation wheel, for example) produce a quick succession of individual events—each with an absolute value. expander An effect process that increases the dynamic range of an audio signal. It is the antithesis of the compression effect (see compressor). export To create a version of a file, such as a Logic Express project, in a different format that can be distributed and used by other applications.
GM Abbreviation for General MIDI. A standard for MIDI sound modules that specifies a uniform set of instrument sounds on the 128 program numbers, a standardized key assignment for drum and percussion sounds on MIDI channel 10, 16-part multi-timbral performance and at least 24 voice polyphony. The GM specification is designed to ensure compatibility between MIDI devices. A musical sequence generated by a GM instrument should play correctly on any other GM synthesizer or sound module.
instrument object An object in the Logic Express Environment designed to communicate with a single-channel MIDI device. An instrument object represents a physical or virtual device which handles MIDI information. Also see multi instrument object. interface 1) A hardware component such as a MIDI or audio device that allows your Logic Express applications to interface (connect) with the outside world. You need an audio or MIDI interface to get sound or MIDI into and out of your computer.
LFO Abbreviation for Low Frequency Oscillator. An oscillator that delivers modulation signals below the audio frequency range—in the bandwidth that falls between 0.1 and 20 Hz, and sometimes as high as 50 Hz or 400 Hz. low cut filter A low cut filter is essentially a highpass filter that offers no slope or resonance controls. It attenuates all frequencies below the defined cutoff.
MIDI Clock Short MIDI message for clock signals. It is used to provide a timing pulse between MIDI devices. It is accurate to 24 ppqn (pulses per quarter note), although some devices interpolate these pulse values, resulting in a more precise clock signal if each device is capable of interpreting this additional information correctly. Also see: SPP. MIDI message A message transmitted via MIDI that consists of one status byte and none, one, two, or many data bytes (with system exclusive commands). See event.
modulation path A modulation path determines which target parameter will be affected by a specific modulator (modulation source). modulation wheel A MIDI controller found on most MIDI keyboards. mono Short for monophonic sound reproduction. The process of mixing audio channels into a single track, using equal amounts of the left and right audio channel signals. Compare with stereo. movie See video. MP3 Abbreviation for MPEG-2 Audio Layer 3.
note number Pitch of a MIDI note, controlled by the first data byte of a MIDI note event. OpenTL Abbreviation for Open Track List. This file format, typically used for data exchange with Tascam hard disk recorders, such as the MX 2424, can be imported and exported by Logic Express. The OpenTL file format only supports the exchange of audio data (audio regions, inclusive of track position information). MIDI and automation data are ignored when using the Logic Express OpenTL export function.
post fader Sends in analog mixers are positioned either before (pre) or after (post) the fader. Post fader means positioned after the volume fader in the signal flow, with the level of a signal going to the send changing along with the fader movements. pre fader Sends in analog mixers are positioned either before (pre) or after (post) the fader.
resonance A term generally associated with filters, particularly those of synthesizers. Resonance emphasizes the frequency range surrounding the cutoff frequency. See cutoff frequency. reverb Reverb(eration) is the sound of a physical space. More specifically, the reflections of soundwaves within a space. As an example, a hand clap in a cathedral will reverberate for a long time as sound waves bounce off the stone surfaces within a very large space.
scale A group of related musical notes (or pitches) that forms the basis of the melody and harmony in a piece of music. The most common scales are the major scale and minor scale. scan code Each key on a computer keyboard has a scan code rather than an ASCII symbol associated with it. As an example: The plus and minus keys on the numeric keypad and the corresponding keys above the keyboard have a different scan code, but use the same ASCII symbol. scroll bar and scroller Gray beam at the edge of a window.
single trigger mode This term is associated with synthesizers such as the ES1. In this mode, envelopes are not retriggered when tied (legato) notes are played. SMF See Standard MIDI File. SMPTE Abbreviation for Society of Motion Picture and Television Engineers. The organization responsible for establishing a synchronization system that divides time into hours, minutes, seconds, frames, and subframes (SMPTE time code). SMPTE time code is also used for synchronizing different devices.
stereo Short for stereophonic sound reproduction of two different audio channels. Compare with mono. subframe A sub-division of a SMPTE frame, corresponding to the individual bits of a SMPTE frame. One frame consists of 80 bits. 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 Express.
time code A format (and signal) for assigning a unique, sequential time unit to each frame of video or project position. The SMPTE time code format, for example, is measured in hours : minutes : seconds : frames and subframes. timing Measure of the ability to play notes at the right time. Timing can also refer to synchronization between events, regions, and devices. toggle To switch between two states such as on or off (applies to windows, parameter values, and so on).
white noise Noise type that consists of all frequencies (an infinite number) sounding simultaneously, at the same intensity, in a given frequency band. Its name is analogous to white light, which consists of a mixture of all optical wavelengths (all rainbow colors). Sonically, white noise falls between the sound of the consonant F and breaking waves (surf ). Synthesis of wind and seashore noises, or electronic snare drum sounds, requires the use of white noise.
Index Index A B aftertouch event 425 AIFF file 425 AKAI file described 425 importing with EXS24 mkII 289–293 aliasing creating artificially 28 described 426 allpass filter 426 amplitude, described 426 amp modeling effects 13–19 Analog Basic (GarageBand Instrument) 412 Analog Mono (GarageBand Instrument) 413 Analog Pad (GarageBand Instrument) 413 analog signal 426 Analog Swirl (GarageBand Instrument) 413 Analog Sync (GarageBand Instrument) 413 analog synthesizer 417 attack phase (envelope) 423, 426 atten
Compressor 37–40 Auto Gain parameter 40 Circuit Type menu 40 parameter overview 37 using 39 compressor described 35, 429 frequency-specific 41 controller, described 429 Controls view (plug-in) 429 Core Audio 429 Core MIDI 429 Correlation Meter 97 cutoff frequency 55, 420 D decibel 430 DeEsser 41 Detector section 41 Suppressor section 42 delay effects 21–25 delaying channel by sample values 22 Denoiser 134–135 parameter overview 134 using 135 density (reverb effect) 124 destructive editing, described 430 di
equalizer.
Bender 228 ENV1 227 ENV2 227 ENV3 228 Kybd 228 LFO1 227 LFO2 227 Max 228 MIDI Controllers A–F 229 ModWhl 228 Pad-X 228 Pad-Y 228 RndNO1 230 RndNO2 230 SideCh 230 Touch 229 Velo 228 Whl+To 229 modulation target 222 Mono button 200 muting oscillators 203 noise 208 Osc1Levl modulation target 224 Osc1WaveB modulation target 224 Osc1Wave modulation target 223 Osc2Levl modulation target 224 Osc2WaveB modulation target 224 Osc2Wave modulation target 223 Osc3Levl modulation target 224 Osc3WaveB modulation target 22
Vel slider 240 via menu (Router) 220 via source 230 Bender 231 ENV1 230 ENV2 230 ENV3 230 Kybd 231 LFO1 230 LFO2 230 ModWhl 231 Pad-X 230 Pad-Y 230 RndNO1 232 RndNO2 232 SideCh 232 Touch 231 Velo 231 Whl+To 231 Voices parameter 201 Wave buttons 234 Wave knob 203 wavetable synthesis 205 ES E 181–182 ES M 183–184 ES P 185–186 EVOC 20 Filterbank 70–74 filter bank parameters 71 LFO parameters 73 Output section 74 EVOC 20 PolySynth 149–171 Analog knob 158 analysis bands 160 Attack slider (envelope) 159 Bands par
previewing 315 root note 321 supported formats 278 b/p button (modulation matrix) 309 bandpass filter 303 bypassing modulation path 309 chain symbol (filter) 303 Clear Find command 283 crossfading layered zones 298 Cutoff knob 303 deleting settings from instrument 281 Dest(ination) menu (modulation matrix) 307 disabling Search filter 283 DLS file 284 Drive knob 303 Edit button 296 EG knob (LFO 1) 306 Enable Find command 283 ENV 1 304 ENV 2 305 EXS24 mkI modulation paths 310 Extract MIDI Region and Add Sampl
SampleCell file 284 sampler instrument 277 backing up 282 copying to hard disk 281 creating 314 creating from ReCycle file 287–288 exporting with its audio files 333 importing 284–293 loading 279 managing 282 renaming 333 saving 333 searching 283 sampler instrument folders 279 sampler instrument settings 281 Sampler Instruments menu 279 organizing 282 refreshing 282 Sample Select modulation destination 311 Sample Start and End parameters 322 saving parameter settings into sampler instrument 281 Scale parame
Analog Sync 413 Bass 413 Digital Basic 414 Digital Mono 414 Digital Stepper 414 Drum Kits 414 Electric Clavinet 415 Electric Piano 415 Guitar 415 Horns 415 Hybrid Basic 415 Hybrid Morph 415 macro parameters 411 overview 411 Piano 416 Sound Effects 416 Strings 416 Tonewheel Organ 416 Tuned Percussion 416 universal parameters 412 Voice 416 Woodwind 416 General MIDI.
parameter overview 49 using 50 noise gates 36 O oscillator 438 Overdrive (effect plug-in) 32 P Parallel Bandpass Vocoder 172 Parametric EQ (effect plug-in) 63 peak, described 438 peak limiters 36 Phase Distortion (effect plug-in) 33 phase inversion 144 Phaser (effect plug-in) 103–104 phaser effect 250 phase relationship, analyzing 97 Piano (GarageBand Instrument) 416 pink noise 438 pitch correcting 115 described 438 shifting 119 Pitch Correction 115–119 automating 119 correction amount display 119 definin
stereo base, extending 91, 94 Stereo Delay 23 Stereo Spread 94 Strings (GarageBand Instrument) 416 SubBass 140–141 parameter overview 140 using 141 subtractive synthesis 419 swing feeling, applying to audio 138 synthesis 418 FM.
switching on/off 368 type 368 Filter section 365–367 arrow button 365 signal flow 365 slope buttons 366 type buttons 365 FM amount knob 356 FM mode (oscillator 1) 356 frequency modulation 356 Full View button 389 Gate button 372 Group menu 372 hi-hat 407 importing EXS instrument sounds 350–351 Ultrabeat sounds 350–351 increasing performance dynamics 404 Inner Loss parameter 360 interface 345 kick drum 396 Kraftwerk snare 406 learning MIDI controller assignment 377 Length menu 384 Level knob 367 LFO 377 cycl
soloing parameter offsets 392 step grid 386 Step mode. See step automation step sequencer 382–395 adding trigger on every down/upbeat 387 automation.
