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Logic Pro 9 Effects

1. Chapter9 Modulation Effects e Sync button Synchronizes the delay to the project tempo You can choose musical note values with the Time knob Level knob and field Sets the level of the delay added to the ring modulated or frequency shifted signal A Level value of 0 passes the effect signal directly to the output bypass Modulating the Ringshifter with the Envelope Follower The oscillator Frequency and Dry Wet parameters can be modulated with the internal envelope follower and the LFO see Modulating the Ringshifter with the LFO The oscillator frequency even allows modulation through the 0 Hz point thus changing the oscillation direction 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 Power button Turns the envelope follower on or off and enables the following parameters Sens itivity slider and field Determines how responsive the envelope follower is to the input signal At lower settings the envelope follower reacts only to the most dominant signal peaks At higher settings the envelope follower tracks the signal more closely but may react less dynamically Attack slider and field Sets the response time of the envelope follower Decay slider and field Controls the time it takes the envelope follower to return
2. You can also adjust the decibel scale of the graphic display by vertically dragging either the left or right edge of the display where the dB scale is shown when the Analyzer is not active When the Analyzer is active dragging the left edge adjusts the linear dB scale and dragging the right edge adjusts the Analyzer dB scale To increase the resolution of the EQ curve display in the most interesting area around the zero line drag the dB scale on the left side of the graphic display upward Drag downward to decrease the resolution Chapter5 Equalizers Using the Channel EQ Analyzer The Analyzer when active makes uses of a mathematical process called a Fast Fourier Transform FFT to provide a real time curve of all frequency components in the incoming signal This is superimposed over any EQ curves you have set The Analyzer curve uses the same scale as the EQ curves making it easy to recognize important frequencies in the incoming audio This also simplifies the task of setting EQ curves to raise or lower the levels of frequencies frequency ranges The bands derived from FFT analysis are divided in a logarithmic scale there are more bands in higher octaves than in lower ones As soon as the Analyzer is activated you can change the scaling with the Analyzer Top parameter on the right side of the graphic display The visible area represents a dynamic range of 60 dB Drag vertically to set the maximum value to anywhere between 20
3. Rate knob and field Defines the frequency the speed of the LFO Intensity slider and field Determines the modulation amount Mix slider and field Determines the balance between dry and wet signals Chapter9 Modulation Effects Microphaser The Microphaser is a simple plug in that allows you to quickly create swooshing phasing effects x Master Volume PF View ShowCS v Show Insert v Bypass lt gt default Microphaser LFO Rate 0 22Hz LFO Rate slider and field Defines the frequency the speed of the LFO Feedback slider and field Determines the amount of the effect signal that is routed back into the input This can change the tonal color and or make the sweeping effect more pronounced Intensity slider and field Determines the amount of modulation Modulation Delay The Modulation Delay is based on the same principles as the Flanger and Chorus effects but you can set the delay time allowing both chorus and flanging effects to be generated It can also be used without modulation to create resonator or doubling effects The modulation section consists of two LFOs with variable frequencies Chapter9 Modulation Effects 187 188 Although rich combined flanging and chorus effects are possible the Modulation Delay is capable of producing some extreme modulation effects These include emulations of tape speed fluctuations and metallic robot like modulations of incoming signals Feedback slide
4. or user To add notes to or remove notes from the chosen scale or chord Click unused keys on the small keyboard to add them to the scale or chord Click selected notes illuminated to remove them Tip Your last edit is remembered If you choose a new scale or chord but do not make any changes you can revert to the previously set scale by choosing user in the pop up menu EVOC 20 TrackOscillator Formant Filter Parameters The EVOC 20 TrackOscillator features two formant filter banks one for the Analysis In section and one for the Synthesis In section In essence the entire frequency spectrum of an incoming signal is analyzed Analysis section and equally divided into a number of frequency bands Each ls bank can control up to 20 of these frequency bands For more information see Hc ca der V s a Vocoder Work Chapter 6 Filter Effects 149 150 The Formant Filter display is divided in two by a horizontal line The upper half applies to the Analysis section and the lower half to the Synthesis section Parameter changes are instantly reflected in the Formant Filter display providing invaluable feedback on what is happening to the signal as it is routed through the two formant filter banks evoc 20 T0 highest QOH Formant Stretch Fonmant Shift Resonance High and Low Frequency parameters Determine the lowest and highest frequencies allowed to pass by the filter section Frequencies that fa
5. Decay knob and field Sets the decay time for the envelope Sustain knob and field Sets the sustain time for the envelope If the input signal falls below the threshold level before the envelope sustain phase the release phase is triggered Release knob and field Sets the release time for the envelope this is triggered as soon as the input signal falls below the threshold Dynamic knob and field Determines the input signal modulation amount You can modulate the peak value of the envelope section by varying this control Cutoff Mod slider and field Determines the impact of the envelope on the cutoff frequency Chapter 6 Filter Effects 133 134 AutoFilter LFO Parameters The LFO is used as a modulation source for filter cutoff Coarse Rate knob Fine Rate slider and field Used to set the speed of LFO modulation Drag the Coarse Rate knob to set the LFO frequency in Hertz Drag the Fine Rate slider the semicircular slider above the Coarse Rate knob to fine tune the frequency Note The labels shown for the Rate knob slider and field change when you activate Beat Sync Only the Rate knob and field is available Beat Sync button Activate to synchronize the LFO to the host application tempo You can choose from bar values triplet values and more These are determined by the Rate knob or field Phase knob and field Shifts the phase relationship between the LFO rate and the host application tempo when Beat Sync is
6. Sets the amount of distortion applied after the filter section AutoFilter Output Parameters The Output parameters are used to set the wet dry balance and overall level Dry Signal slider and field Sets the amount of the original dry signal added to the filtered signal e Main Out slider and field Sets the overall output level of the AutoFilter allowing you to compensate for higher levels caused by adding distortion or by the filtering process itself Chapter 6 Filter Effects EVOC 20 Filterbank The EVOC 20 Filterbank consists of two formant filter banks The input signal passes through the two filter banks in parallel Each bank features level faders for up to 20 frequency bands allowing independent level control of each band Setting a level fader to its minimum value completely suppresses the formants in that band You can control the position of the filter bands with the Formant Shift parameter You can also crossfade between the two filter banks A Short Primer on Formants A formant is a peak in the frequency spectrum of a sound When the term is used in relation to human voices formants are the key component that enables humans to distinguish between different vowel sounds based purely on the frequency of these sounds Formants in human speech and singing are produced by the vocal tract with most vowel sounds containing four or more formants Getting to Know the EVOC 20 Filterbank Interface The EVOC 20 Filterbank
7. Signals that exceed the threshold are reduced in level Attack knob and field Sets the amount of time it takes for the compressor to react when the signal exceeds the threshold Release knob and field Sets the amount of time it takes for the compressor to stop reducing the signal after the signal falls below the threshold Ratio slider and field Sets the ratio by which the signal is reduced when it exceeds the threshold Chapter 4 Dynamics Processors Silver Gate The Silver Gate is a simplified version of the Noise Gate for usage tips see Using the Noise Gate Lookahead slider and field Sets how far ahead the noise gate analyzes the incoming signal allowing the Silver Gate to respond more quickly to peak levels Threshold slider and field Sets the threshold level Signals that fall below the threshold will be reduced in level Attack knob and field Sets the amount of time it takes to fully open the gate after the signal exceeds the threshold 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 Surround Compressor The Surround Compressor based on the Compressor is specifically designed for compression of complete surround mixes It is commonly inserted in a surround output channel strip or in audio or aux channel strips busses
8. The main use of a limiter is to prevent clipping while preserving the maximum overall signal level Noise gates Noise gates alter the signal in a way that is opposite to that used by compressors or limiters Whereas a compressor lowers the level when the signal is louder than the threshold a noise gate lowers the signal level whenever it falls below the threshold Louder sounds pass through unchanged but softer sounds such as ambient noise or the decay of a sustained instrument are cut off Noise gates are often used to eliminate low level noise or hum from an audio signal Chapter 4 Dynamics Processors Adaptive Limiter The Adaptive Limiter is a versatile tool for controlling the perceived loudness of sounds It works by rounding and smoothing peaks in the signal producing an effect similar to an analog amplifier being driven hard Like an amplifier it can slightly color the sound of the signal You can use the Adaptive Limiter to achieve maximum gain without introducing generally unwanted distortion and clipping which can occur when the signal exceeds 0 dBFS The Adaptive Limiter is typically used on the final mix where it may be placed after a compressor such as the Multipressor and before a final gain control resulting in a mix of maximum loudness The Adaptive Limiter can produce a louder sounding mix than can be achieved by normalizing the signal Note Using the Adaptive Limiter adds latency when the Lookahead parameter is
9. also known as impulses 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 As a result percussive drum and instrumental rhythm parts such as basslines are suitable for tempo analysis Pad sounds are a poor choice 167 168 The LED shows the current analysis status If the LED is flashing a tempo measurement is taking place When the LED is continuously lit analysis is complete and the tempo is displayed The measurement ranges from 80 to 160 beats per minute The measured value is displayed with an accuracy of one decimal place Click the LED to reset the BPM Counter Note The BPM Counter also detects tempo variations in the signal and tries to analyze them accurately If the LED starts flashing during playback this indicates that the BPM Counter has detected a tempo that has deviated from the last received or set tempo As soon as a new constant tempo is recognized the LED is solidly lit and the new tempo displayed Correlation Meter The Correlation Meter displays the phase relationship of a stereo signal Acorrelation of 1 the far right position means that the left and right channels correlate 100 they are completely in phase A correlation of 0 the center position indicates the widest permissible left right divergence often audible as an extremely wide s
10. four and enables you to independently compress each band After compression is applied the bands are combined into a single output signal The advantage of compressing different frequency bands separately is that it allows you to apply more compression to the bands that need it without affecting other bands This avoids the pumping effect often associated with high amounts of compression As the Multipressor allows the use of higher compression ratios on specific frequency bands it can achieve a higher average volume without causing audible artifacts Raising the overall volume level can result in a corresponding increase in the existing noise floor Each frequency band features downward expansion which allows you to reduce or suppress this noise Downward expansion works as a counterpart to compression Whereas the compressor compresses the dynamic range of higher volume levels the downward expander expands the dynamic range of the lower volume levels With downward expansion the signal is reduced in level when it falls below the threshold level This works in a similar way to a noise gate but rather than abruptly cutting off the sound it smoothly fades the volume with an adjustable ratio Chapter 4 Dynamics Processors Multipressor Parameters The parameters in the Multipressor window are grouped into three main areas the graphic display in the upper section the set of controls for each frequency band in the lower section and th
11. similar to a Leslie 760 model Single Simulates the sound of a Leslie with a single full range rotor The sound resembles the Leslie 825 model Split The bass rotor s signal is routed slightly to the left and the treble rotor s signal is routed more towards the right Wood amp Horn IR This setting uses an impulse response of a Leslie with a wooden enclosure Proline amp Horn IR This setting uses an impulse response of a Leslie with a more open enclosure Split amp Horn IR This setting uses an impulse response of a Leslie with the bass rotor signal routed slightly to the left and the treble rotor signal routed more to the right Advanced Rotor Speaker Parameters The Rotor Cabinet offers the following advanced rotor speaker parameters Horn Deflector button A Leslie cabinet contains a double horn with a deflector at the horn mouth This deflector makes the Leslie sound Some people remove the deflector to increase amplitude modulation and decrease frequency modulation You can emulate this by using the Horn Deflector button to switch the deflectors on and off e Motor Ctrl pop up menu You can set different speeds for the bass and treble rotors in the Motor Ctrl pop up menu Note If you choose Single Cabinet from the Cabinet menu the Motor Ctrl setting is irrelevant because there are no separate bass and treble rotors in a single cabinet Normal Both rotors use the speed determined by the rotor speed but
12. stage Master knob Sets the output volume of the amplifier going to the cabinet For tube amplifiers increasing the Master level typically produces a somewhat compressed and saturated sound resulting in a more distorted and powerful that is louder signal High Master settings can produce an extremely loud output that can damage your speakers or hearing so ramp this up slowly The final output level of Amp Designer is set with the Output slider at the lower right edge of the interface See Setting Amp Designer s Output Level Chapter 1 Amps and Pedals 23 24 Getting to Know Amp Designer s Effects Parameters The effects parameters include Tremolo Vibrato and Reverb which emulate the processors found on many amplifiers these controls are found in the center of the knobs section You can use the switch toward the right to select either Tremolo TREM which modulates the amplitude or volume of the sound or Vibrato VIB which modulates the pitch Reverb which is controlled by a switch in the middle can be added to either of these effects or used independently Note The Effects section is placed before the Presence and Master controls in the signal flow and receives the pre amplified pre Master signal Reverb Tremolo and Vibrato are described in the following sections e Using Amp Designer s Reverb Effect e Using Amp Designer s Tremolo and Vibrato Effects Using Amp Designer s Reverb Effect Reverb is always
13. the target Match EQ instance See below To match the EQ of a project mix to the EQ of a source audio file In the project you want to match to the source audio file instantiate a Match EQ typically on Output 1 2 Drag the source audio file onto the Template Learn button Return to the start of your mix click Current Material Learn then play your mix the current material from start to finish When you are done click Current Material Match this automatically disengages the Current Material Learn button To use the matched EQ on a channel strip Choose the channel strip that you want to match from the Sidechain menu of the Match EQ window Click the Template Learn button Play the entire source audio file from start to finish then click the Template Learn button again to stop the learn process Return to the start of your mix click Current Material Learn then play your mix the current material from start to finish Chapter5 Equalizers 5 When you are done click Current Material Match this automatically disengages the Current Material Learn button Match EQ creates a filter curve based on the differences between the spectrum of the template and the current material This curve automatically compensates for differences in gain between the template and the current material with the resulting EQ curve referenced to 0 dB A yellow filter response curve appears in the graphic display showing the average spectrum of you
14. 16 if it consists mostly of sixteenth notes Swing slider and field Determines the amount that even beats are delayed A value of 50 means no swing which is typical for most pop and rock music styles The higher the value the stronger the swing effect e Accent slider and field Raises or lowers the level of even beats accentuating them Such accents are typical of a variety of rhythmic styles such as swing or reggae Chapter 13 Specialized Effects and Utilities Speech Enhancer You can use Speech Enhancer to improve speech recordings made with your computer s internal microphone if applicable It combines denoising advanced microphone frequency remodeling and multiband compression Denoise Mic Correction Mic Model Voice Enhonce Enhance Mode 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 Settings towards 0 dB increasingly suppress background noise but also proportionately increase artifacts e Mic Correction buttons Activate the On button to improve the frequency response of recordings made with your built in microphone This creates the impression that an up market microphone was used Mic Model pop up menu Provides a choice of several microphone models that compensate for tonal characteristics of particular built in Macintosh microphones Note You can use
15. B C or no group indicated by Moving the Threshold or Output Level slider for any grouped channel will move the sliders for all channels assigned to that group Tip Press Command and Option while moving the Threshold or Output Level slider of a grouped channel to temporarily unlink the channel from the group This allows you to make independent threshold settings while maintaining the side chain detection link necessary for a stable surround image Byp Bypass buttons Independently bypasses the corresponding channel unless grouped If the channel belongs to a group all channels in the group are bypassed Detection pop up menu Determines if the Surround Compressor uses the maximum level of each signal Max or the summed level of all signals Sum to exceed or fall below the threshold If Max is chosen and any of the surround channels exceeds or falls below the Threshold that channel or grouped channels is compressed e If Sum is chosen the combined level of all channels must exceed the Threshold before compression occurs Surround Compressor Main Parameters The Surround Compressor s Main section provides the following parameters e Ratio knob and field Sets the ratio of signal reduction when the threshold is exceeded Knee knob and field Determines the ratio of compression at levels close to the threshold Attack knob and field Sets the amount of time it takes to reach full compression after the signal excee
16. Browser on the right side of the interface Each effect and utility is grouped into a category such as distortion modulation and so on For information about these types of stompboxes see Distortion Pedals Modulation Pedals Delay Pedals Filter Pedals Dynamics Pedals and Utility Pedals View pop up menu Import Mode button je rere GRINDER mo i A To hide or show the Pedal Browser Click the disclosure triangle in the lower right corner of the Pedal area To show only specific pedal groups in the Pedal Browser Open the View pop up menu and choose Distortion Modulation Delay Filter Dynamics or Utility The Pedal Browser shows only the stompboxes within the category you choose To show all the pedal groups choose Show All from the View pop up menu To add a stompbox to the Pedal area Do one of the following Drag the effect that you want to insert from the Pedal Browser to the appropriate Pedal area position This can be to the left to the right or in between existing pedals Double click an effect in the Pedal Browser to add it to the right of all existing stompboxes in the Pedal area Chapter 1 Amps and Pedals 37 38 Note Double clicking a stompbox in the Pedal Browser when a stompbox is selected in the Pedal area will replace the selected pedal Using Pedalboard s Import Mode Pedalboard has a feature you can use to import parameter settings for each type of pedal In contrast to the plug in window S
17. CPU demand than the other options Apply slider and field Determines the impact of the filter curve on the signal Values above 100 magnify the effect Values below 100 reduce it e Negative values 1 to 100 invert the peaks and troughs in the filter curve Chapter5 Equalizers 123 124 A value of 100 has no impact on the filter curve Smoothing slider and field Sets the amount of smoothing for the filter curve using a constant bandwidth set in semitone steps A value of 0 0 has no impact on the filter curve A value of 1 0 means a smoothing bandwidth of one semitone A value of 4 0 means a smoothing bandwidth of four semitones a major third A value of 12 0 means a smoothing bandwidth of one octave and so on Note Smoothing has no effect on any manual changes you make to the filter curve Fade Extremes checkbox Extended Parameters area Select to smooth the filter curve at the high and low extremes of the frequency spectrum Using the Match EQ Following is a common usage example that you can adapt to your own workflow In this example the frequency spectrum of a mix is matched with the spectrum of a source audio file To learn or create a Match EQ template Do one of the following Drag an audio file from the finder onto the Template Learn button and select the source channel strip as a sidechain See below Use the Match EQ on the source channel strip and save a setting Import this setting into
18. Formant Stretch and Formant Shift are also useful if the frequency spectrum of the synthesis signal does not complement the frequency spectrum of the analysis signal You could create a synthesis signal in the high frequency range from an analysis signal that mainly modulates the sound in a lower frequency range for example Note The use of the Formant Stretch and the Formant Shift parameters can result in the generation of unusual resonant frequencies when high Resonance settings are used EVOC 20 TrackOscillator Modulation Parameters The parameters in this section control the LFO which can be used to modulate either the frequency the pitch of the tracking oscillator thus creating a vibrato or the Formant Shift parameter of the synthesis filter bank Formant Stretch Formant Shift Resonance Wave C Intensity Rate J LFO z Shift Intensity slider Controls the amount of formant shift modulation by the LFO Pitch Intensity slider Controls the amount of pitch modulation vibrato by the LFO Chapter 6 Filter Effects 151 e Waveform buttons Set the waveform type used by the LFO You can choose between triangle falling and rising sawtooth square up and down around zero bipolar good for trills square up from zero unipolar good for changing between two definable pitches a random stepped waveform S amp H and a smoothed random waveform for each LFO LFO Rate knob and field Determines the speed of modulation Valu
19. In most mixes you should set a lower level for the low frequency reverb signal This enables you to boost the bass level of the incoming signal making it sound punchier This also helps to counteract bottom end masking effects High Cut slider and field Frequencies above the set value are filtered from the reverb signal Uneven or absorbent surfaces wallpaper wood paneling carpets and so on tend to reflect lower frequencies better than higher frequencies The High Cut filter replicates this effect If you set the High Cut filter so that it is wide open maximum value the reverb will sound as if it is reflecting off stone or glass Density slider and field Controls the density of the diffuse reverb tail Ordinarily you want the signal to be as dense as possible In rare instances however a high Density value can color the sound which you can fix by reducing the Density slider value Conversely if you select a Density value that is too low the reverb tail will sound grainy Chapter 11 Reverb Effects 223 Diffusion slider and field Sets the diffusion of the reverb tail High Diffusion values represent a regular density with few alterations in level times and panorama position over the course of the diffuse reverb signal Low Diffusion values result in the reflection density becoming irregular and grainy This also affects the stereo spectrum As with Density find the best balance for the signal Reverb Time slider and f
20. Spectral Gate Imaging Processors Binaural Post Processing Direction Mixer Stereo Spread Metering Tools BPM Counter Correlation Meter Level Meter Plug in MultiMeter Surround MultiMeter Tuner Modulation Effects Chorus Effect Ensemble Effect Flanger Effect Microphaser Modulation Delay Phaser Effect Ringshifter Contents Chapter 10 Chapter 11 Chapter 12 Chapter 13 Chapter 14 197 199 201 202 203 203 207 208 213 214 214 215 218 225 227 228 229 233 239 241 247 249 249 252 253 255 259 259 260 261 263 263 Rotor Cabinet Effect Scanner Vibrato Effect Spreader Tremolo Effect Pitch Effects Pitch Correction Effect Pitch Shifter II Vocal Transformer Reverb Effects Plates Digital Reverb Effects and Convolution Reverb AVerb EnVerb GoldVerb PlatinumVerb SilverVerb Space Designer Convolution Reverb Getting to Know the Space Designer Interface Working with Space Designer s Impulse Response Parameters Working with Space Designer s Envelope and EQ Parameters Working with Space Designer s Filter Working with Space Designer s Global Parameters Automating Space Designer Specialized Effects and Utilities Denoiser Enhance Timing Exciter Grooveshifter Speech Enhancer SubBass Utilities and Tools Down Mixer Gain Plug in 1 0 Utility Multichannel Gain Test Oscillator Contents An Introduction to the Logic Pro Effects Preface Logic Pro
21. Spring Resonant Spring Another 1960s style spring with a strong slightly distorted mid range emphasis Boutique Spring A modernized version of the classic Vintage Spring with a richer tone in the bass and mids Sweet Reverb A smooth modern reverb with rich lows and restrained highs Rich Reverb A bold well balanced modern reverb Warm Reverb A lush modern reverb with rich low mids and understated highs Using Amp Designer s Tremolo and Vibrato Effects Tremolo and vibrato are controlled by several switches and two knobs in the Effects section found toward the right of the knobs section Tremolo modulates the amplitude or volume of the sound and vibrato modulates the pitch EFFECTS On Off switch Enables or disables the tremolo or vibrato effect Trem Vib switch Choose either tremolo or vibrato Depth knob Sets the intensity of the modulation tremolo or vibrato Speed knob Sets the speed of the modulation in Hertz Lower settings produce a smooth floating sound Higher settings produce a rotor like effect Chapter 1 Amps and Pedals 25 26 Sync Free switch When the switch is set to Sync the modulation speed is synchronized with the host application tempo The Speed knob lets you select different bar beat and musical note values 1 8 1 16 and so on including triplet and dotted note values When the switch is set to Free the modulation speed can be set to any available value with the Speed knob Sett
22. active Usually it should be used for mixing and mastering previously recorded tracks not while recording Input meters to the left Show the input levels in real time as the file or project plays The Margin field shows the highest input level You can reset the Margin field by clicking it Input Scale knob and field Scales the input level Scaling is useful for handling very high level or low level input signals It essentially squeezes the higher and lower signal levels into a range that allows the Gain knob to work effectively In general the input level should never exceed 0 dBFS which can result in unwanted distortion Gain knob and field Sets the amount of gain after input scaling Chapter 4 Dynamics Processors 87 88 Out Ceiling knob and field Sets the maximum output level or ceiling The signal will not rise above this Output meters to the right Show output levels allowing you to see the results of the limiting process The Margin field shows the highest output level You can reset the Margin field by clicking it Mode buttons Extended Parameters area Choose the type of peak smoothing OptFit Limiting follows a linear curve which allows signal peaks above 0 dB NoOver Avoids distortion artifacts from the output hardware by ensuring that the signal does not exceed 0 cB Lookahead field and slider Extended Parameters area Adjusts how far ahead the Adaptive Limiter analyzes the file
23. an audible echo begin materializing Obviously these guidelines are intended to help you design realistic sounding spaces that are suitable for different signals If you want to create unnatural sound stages or otherworldly reverbs and echoes feel free to experiment with the Pre Dly parameter Using Space Designer s IR Start Parameter The IR Start parameter enables you to shift the playback start point of the impulse response which will effectively cut off the beginning of the impulse response Chapter 12 Space Designer Convolution Reverb This can be useful for eliminating level peaks at the beginning of the impulse response sample Its use also affords a number of creative options particularly when combined with the Reverse function See Using Space Designer s Button Bar Note The IR Start parameter is not available or required in Synthesized IR mode because by design the Length parameter provides identical functionality Using Space Designer s Spread Parameters The Spread and Xover knobs enhance the perceived width of the signal without losing the directional information of the input signal normally found in the higher frequency range Low frequencies are spread to the sides reducing the amount of low frequency content in the center allowing the reverb to nicely wrap around the mix The Spread and Xover knobs function only in Synthesized IR mode Note As these parameters adjust stereo or surround processing they have no i
24. and includes a set of parameters for each band The Fat EQ offers the following parameters Band Type buttons Located above the graphic display For bands 1 2 and 4 5 click one of the paired buttons to select the EQ type for the corresponding band Band 1 Click the highpass or low shelving button Band 2 Click the low shelving or parametric button Band 3 Always acts as a parametric EQ band Band 4 Click the parametric or high shelving button Band 5 Click the high shelving or lowpass button Graphic display Shows the EQ curve of each frequency band Frequency fields Sets the frequency for each band Gain knobs Set the amount of gain for each band 116 Chapter 5 Equalizers Q fields Sets the Q or bandwidth of each band the range of frequencies around the center frequency that are altered At low Q factor values the EQ covers a wider frequency range At high Q values the effect of the EQ band is limited to a narrow frequency range The Q value can significantly influence how audible your changes are if you re working with a narrow frequency band you ll generally need to cut or boost more drastically to notice the difference Note For bands 1 and 5 this changes the slope of the filter Band On Off buttons Enables disables the corresponding band Master Gain slider and field Sets the overall output level of the signal Use it after boosting or cutting individual frequency bands Linear
25. and use an external effects unit with the I O utility Connect an output or output pair of your audio interface with the input pair on your effects unit Connect the output or output pair of your effects unit 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 Click an Insert slot of an aux channel strip being used as a bus send return and choose Utility gt I O In the I O window choose both the Outputs and Inputs of your audio hardware that your effects unit is connected to Route the signals of any channel strips that you want to process to the bus aux channel strip chosen in step 3 and set appropriate Send levels Adjust the Input or Output volume as required in the I O window Click the Latency Detection Ping button if you want to detect and compensate for any delay between the selected output and input When you start playback the signals of any channel strips routed to the aux channel chosen in step 3 will be processed by the external effects unit Chapter 14 Utilities and Tools Multichannel Gain Multichannel Gain allows you to independently control the gain and phase of each channel in a surround mix e Master slider and field Sets the master gain for the combined channel output Channel gain sliders and fields Set the amount of gain for the respective channel e Phase In
26. at the beginning and end of the sine sweep Sweep Mode pop up menu Extended Parameters area Choose Linear or Logarithmic sweep curve Trigger button and pop up menu Click the Trigger button to trigger the sine sweep Choose the behavior of the Trigger button in the pop up menu Single Triggers the sweep once Continuous Triggers the sweep indefinitely Level slider and field Determines the overall output level of the Test Oscillator Chapter 14 Utilities and Tools
27. bands Each frequency band represents one third of an octave The Analyzer parameters are used to activate Analyzer mode and to customize the way that the incoming signal is shown in the main display aM ator sd Mono Goniometer Analyzer parameters Scale Analyzer button Switches the main display to Analyzer mode Left Right LRmax and Mono buttons Determine which channels are displayed in the Analyzer results in the main display Left or Right Displays the left or right channels LRmax Displays the maximum level of the stereo inputs e Mono Displays the spectrum of the mono sum of both stereo inputs View fields Alter the way that values are shown in the Analyzer by setting the maximum level displayed Top and the overall dynamic range Range Mode buttons Determine how levels are displayed You can choose from Peak Slow RMS or Fast RMS characteristics The two RMS modes show the effective signal average and provide a representative overview of perceived volume levels The Peak mode shows level peaks accurately Scale shown in main display Indicates the scale of levels Adjusting the scale is useful when analyzing highly compressed material as it makes it easier to identify small level differences Drag vertically on the scale to adjust Chapter 8 Metering Tools 171 172 Using the MultiMeter Goniometer A goniometer helps you to judge the coherence of the stereo image and dete
28. be 48 kHz 24 kHz and 12 kHz Chapter 12 Space Designer Convolution Reverb 231 232 If the project sample rate is 44 1 kHz the options will be 22 05 kHz 11 025 kHz and 5512 5 Hz Changing the sample rate upward increases or changing it downward decreases the frequency response and length of the impulse response and to a degree the overall sound quality of the reverb Upward sample rate changes are of benefit only if the original IR sample actually contains higher frequencies When you are reducing the sample rate use your ears to decide if the sonic quality meets your needs Note Natural room surfaces except concrete and tiles tend to have minimal reflections in the higher frequency ranges making the half rate and full rate IRs sound almost identical When you select half the sample rate the impulse response becomes twice as long The highest frequency that can be reverberated will be halved This results in a behavior that is much like doubling every dimension of a virtual room multiplying a room s volume by eight Another benefit of reducing the sample rate is that processing requirements drop significantly making half sample rate settings useful for large open spaces Activating the preserve length button preserves the length of the impulse response when the sample rate is changed Manipulating these two parameters as you see fit can lead to interesting results The lower sample rates can also be used
29. below the graphic display Template Displays the learned frequency curve template for the source file This is shown in red Current Material Displays the frequency curve for the audio learned as current material This is shown in green Filter Displays the filter curve created by matching the template and the current material This is shown in yellow View button Determines if separate curves are displayed by the Analyzer L amp R for stereo All Cha for surround or the summed maximum level is shown LR Max for stereo Cha Max for surround Note The View parameters are disabled when using the effect on a mono channel Select buttons Determines if changes made to the filter curve created by matching the template with the current material are applied to the left channel L the right channel R or both channels L R Note The Select parameters are disabled when using the effect on a mono channel Chapter 5 Equalizers Select menu Surround instances only The Select buttons are replaced by the Select menu enabling you to choose an individual channel or all channels Changes to the filter curve will affect the chosen channel when a single channel is selected Channel Link slider and field Refines the settings made with the Select buttons or Select menu When set to 100 all channels L and R for stereo or all surround channels are represented by a common EQ curve When set to 0 a separate filter curve
30. contains a variety of advanced filter based effects that you can use to creatively modify your audio These effects are most often used to radically alter the frequency spectrum of a sound or mix Note Equalizers EQs are special types of filters Typically they are not used as effects per se but as tools to refine the frequency spectrum of a sound or mix See Equalizers This chapter covers the following e AutoFilter p 131 e EVOC 20 Filterbank p 137 e EVOC 20 TrackOscillator p 141 e Fuzz Wah p 153 e Spectral Gate p 157 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 effect works by analyzing incoming signal levels through use of a threshold parameter Any signal level that exceeds the threshold is used as a trigger for a synthesizer style ADSR envelope or an LFO low frequency oscillator These control sources are used to dynamically modulate the filter cutoff The AutoFilter allows you to choose between different filter types and slopes control the amount of resonance add distortion for more aggressive sounds and mix the original dry signal with the processed signal 131 132 Getting to Know the AutoFilter Interface The main areas of the AutoFilter window are the Threshold Envelope LFO Filter Distortion and Output parameter sections Threshold
31. determines the sample rate of the loaded impulse response Activate the preserve length button to preserve the length of the impulse response when changing the sample rate See Setting Space Designer s IR Sample Rate Length field Adjusts the length of the impulse response See Setting Impulse Response Lengths in Space Designer Synthesized IR button Click to activate Synthesized IR mode A new synthesized impulse response is generated This is derived from the values of the Length envelope Filter EQ and Spread parameters See Working in Space Designer s Synthesized IR Mode Note You may freely switch between a loaded impulse response sample and a synthesized impulse response without losing the settings of the other For more information see Working in Space Designer s Synthesized IR Mode Chapter 12 Space Designer Convolution Reverb 229 230 Important To convolve audio in real time Space Designer must first calculate any parameter adjustments to the impulse response This requires a moment or two following parameter edits and is indicated by a blue progress bar During this parameter edit processing time you can continue to adjust the parameter When calculation starts the blue bar is replaced by a red bar advising you that calculation is taking place nple rate iple rate preserve length preserve length Working in Space Designer s IR Sample Mode In IR Sample mode Space Designer loads and uses an impu
32. distortion They are great for clean toned rock vintage R amp B surf music twangy country jazz or any other style where strong note definition is essential Model Description Large Blackface Combo A 4x 10 combo with a sweet well balanced tone favored by rock surf and R amp B players Great for lush reverb drenched chords or strident solos Silverface Combo A 2x 12 combo with a loud ultra clean tone Its percussive articulate attack is great for funk R amp B and intricate chord work It can be crunchy when overdriven but most players favor it for clean tones Mini Blackface Combo A1x10 combo that is bright and open sounding with a surprising amount of low end impact It excels at clean tones with just a hint of overdrive Small Brownface Combo A 1 x 12 combo that is smooth and rich sounding but retains a nice level of detail Blues Blaster Combo A 1 x 15 combo that has a clear top end with a tight defined low end This model is favored by blues and rock players Chapter 1 Amps and Pedals Tip While these amps tend toward a clean and tight sound you can use a Pedalboard distortion stompbox to attain hard edged crunch sounds with a biting treble and extended low end definition See Distortion Pedals and Pedalboard British Stacks The British Stack models are based on the 50 and 100 watt amplifier heads that have largely defined the sound of heavy rock especially when paired with their sign
33. effect pedals are moved to the left and are inserted into Bus A Note A Mixer pedal cannot be moved to a position directly after or to the left of a corresponding split point or Splitter utility To change a Splitter utility position in the Pedal area Drag the Splitter utility to a new position either to the right or the left When moved to the left The split between Bus A and B will occur at the earlier insertion point Relevant effect pedals are moved to the right and are inserted into Bus A When moved to the right The split between Bus A and B will occur at the later insertion point Relevant effect pedals are moved to the left and are inserted into Bus A Note A Splitter pedal cannot be moved to a position directly preceding or to the right of a corresponding Mixer utility To replace a pedal in the Pedal area Do one of the following Drag the stompbox from the Pedal Browser directly over the pedal you want to replace in the Pedal area Chapter 1 Amps and Pedals Click to select the stompbox you want to replace in the Pedal area then double click the appropriate pedal in the Pedal Browser Note You can only replace effect pedals not the Mixer or Splitter utilities Bus routings if active are not changed when an effect pedal is replaced To remove a pedal from the Pedal area Do one of the following Drag the pedal out of the Pedal area Click the pedal to select it and press the Delete key Using
34. has an extensive range of digital signal processing DSP effects and processors that are used to color or tonally shape existing audio recordings software instruments and external audio sources in real time These cover almost every audio processing and manipulation need you will encounter in your day to day work The most common processing options include EQs dynamic processors modulations distortions reverbs and delays Less common are simulations of amplifiers and speaker cabinets which enable you to play your instruments or other signals through a range of vintage and modern sound reproduction systems Guitarists and keyboard players will also benefit from a number of classic pedal effect emulations Further advanced features include precise signal meters and analyzers a test tone generator noise reduction pitch correction imaging bass enhancement and time altering processors and utilities As you can see many of the included processors and utilities don t really fall into the effects category but they may prove to be invaluable in your studio This preface covers the following e About the Logic Pro Effects p 7 e About the Logic Pro Documentation p 8 e Additional Resources p 8 About the Logic Pro Effects All effects processors and utilities provide an intuitive interface that simplifies operation enabling you to work quickly Outstanding audio quality is assured when needed or at the o
35. internal filter circuit that allows more low Lo or high Hi frequency content to be heard You can also disable this filter circuit by choosing Off Spring Box A spring reverb pedal Time sets the length of the reverberation to short medium or long values Tone controls the cutoff frequency making the effect brighter or darker Style switches between algorithms each with different characteristics You can choose from Boutique Simple Vintage Bright and Resonant Mix sets the ratio between the source and effect signals Tru Tape Delay A vintage tape delay effect The Norm Reverse switch changes the delay playback direction Reverse mode is indicated by a blue LED and Normal mode is indicated by a red LED Hi Cut and Lo Cut activate a fixed frequency filter Dirt sets the amount of input signal gain which can introduce an overdriven saturated quality Flutter emulates speed fluctuations in the tape transport mechanism Time sets the modulation speed and can run freely or be synchronized with the host application tempo by enabling the Sync button When synchronized you can specify bar beat and note values including triplets and dotted notes Feedback determines the amount of the effect signal that is routed back into the input The buildup of repeating signals can be used creatively for dub delay and other effects by adjusting Feedback in real time Mix sets the balance between the source and effect signals Filter Pedals Th
36. is activated you can change the scaling with the Analyzer Top parameter on the right side of the graphic display The visible area represents a dynamic range of 60 dB Drag vertically to set the maximum value to anywhere between 20 dB and 40 dB The Analyzer display is always dB linear Note When choosing a resolution be aware that higher resolutions require significantly more processing power High resolution is necessary when trying to obtain accurate analysis of very low bass frequencies for example It is recommended that you disable the Analyzer or close the Linear Phase EQ window after setting the appropriate EQ parameters This will free up CPU resources for other tasks Match EQ The Match EQ allows you to store the average frequency spectrum of an audio file as a template and apply the template to another audio signal so that it matches the spectrum of the original file This is also known as a fingerprint EQ where one sonic fingerprint is applied to another signal The Match EQ enables you to acoustically match the tonal quality or overall sound of different songs you plan to include on an album for example or to impart the color of any source recording to your own projects Match EQ is a learning equalizer that analyzes the frequency spectrum of an audio signal such as an audio file a channel strip input signal or a template The average frequency spectrum of the source file the template and of the current material thi
37. is displayed for each channel Use the Select buttons or Select menu to choose each channel Settings between 0 and 100 allow you to blend these values with your filter curve changes for each channel This results in a hybrid curve Note The Channel Link parameters are disabled when using the effect on a mono channel LFE Handling buttons Extended Parameters area In surround instances allow you to process or bypass the LFE channel Match EQ Display Learn and Match Parameters Graphic display Displays the filter curve created by oe the o to the current material You can edit the filter curve see Editing the Match EQ Filte ve Template Learn button Click to start the process of learning the frequency spectrum of the source file Click again to stop the learning process Current Material Learn button Click to start the process of learning the frequency spectrum of the project you want to match the source file Click again to stop the learning process Current Material Match button Matches the frequency spectrum of the current material to that of the template source file Match EQ Processing Parameters e Phase pop up menu Switches the operational principle of the filter curve Linear prevents processing from altering the signal phase but the latency of the plug in is increased e Minimal alters the signal phase minimally but latency is reduced e Minimal Zero Latency adds no latency but has a higher
38. is shown in both directions A vertical red line to the left of the correlation indicator shows the maximum negative phase deviation value You can reset this line by clicking on it during playback The LFE Correlation Meter s scale values indicate the following A 1 correlation value indicates that the signal is balanced Correlation values in the blue zone between 1 and the middle position indicate that the signal is mono compatible The middle position indicates the highest allowable amount of channel divergence When the meter moves into the red area to the left of the center position out of balance material is present Surround MultiMeter Peak Parameters The Surround MultiMeter offers the following Peak parameters Hold button Activates peak hold for all metering tools in the Surround MultiMeter as follows Analyzer A small yellow segment above each level bar indicates the most recent peak level Goniometer All illuminated pixels are held during a peak hold Level Meter A small yellow segment above each level bar indicates the most recent peak level Balance Correlation Meter The horizontal area around the white correlation indicator denotes phase correlation deviations in real time in both directions Chapter 8 Metering Tools 179 180 Note This meter must be manually opened by clicking on the Balance Correlation button Hold Time pop up menu When peak hold is active sets the
39. of a room s reverb characteristics or to be more precise a recording of all reflections in a given room following an initial signal spike The actual impulse response file is a standard audio file To understand how this works imagine a situation where Space Designer is used ona vocal track An IR recorded in an actual opera house is loaded into Space Designer This IR is convolved with your vocal track placing the singer inside the opera house Convolution can be used to place your audio signal inside any space including a speaker cabinet a plastic toy a cardboard box and so on All you need is an IR recording of the space In addition to loading impulse responses Space Designer includes an on board impulse response synthesis facility This enables you to create completely unique effects particularly when the synthesized IR doesn t represent a real space You can also record and edit impulse responses with Impulse Response Utility which is accessed from Space Designer s IR Sample menu Space Designer also offers features such as envelopes filters EQ and stereo surround balance controls which provide precise control over the dynamics timbre and length of the reverberation Space Designer can operate as a mono stereo true stereo meaning each channel is processed discretely or surround effect This chapter covers the following Getting to Know the Space Designer Interface p 228 e Working with Space Designer
40. on using the Amp Cabinet and Mic pop up menus located on the black bar at the bottom of the interface The EQ pop up menu is accessed by clicking the word EQ or Custom EQ toward the left of the knobs section Note If you create your own hybrid amp combo you can use the Settings menu to save it as a setting file which also includes any parameter changes you may have made EQ pop up menu Amp pop up menu Cabinet pop up menu Model pop up menu Mic pop up menu Building an Amp Designer model is described in the following sections e Choosing an Amp Designer Amplifier e Choosing an Amp Designer Cabinet e Using Amp Designer s Equalizer e Setting Amp Designer Microphone Parameters Choosing an Amp Designer Amplifier You can choose an amplifier model from the Amp pop up menu on the black bar at the bottom of the Amp Designer interface See the following sections for details on the characteristics of each amplifier in these categories Tweed Combos Classic American Combos e British Stacks British Combos e British Alternatives e Metal Stacks Chapter 1 Amps and Pedals Choosing an Amp Designer Cabinet Cabinets have a en impact on the character of a guitar sound see Amp Designer Cabinet Reference Table While certain amplifier and cabinet pairings ve been popular for decades pa from them is an effective way to create fresh sounding tones For example most players automatically associate British heads with 4 x 12
41. panner allowing you to determine the surround position of each tap Note In the Tap display s Pan view mode you can only adjust the angle of taps You must use the surround panner on the Tap Parameter bar to adjust diversity flip x mute level HASH e J NAPS A M 4 2dB gt To easily move the surround position you can Command drag to adjust diversity Command Option drag to adjust the angle Option click the blue dot to reset the angle and diversity Note Delay Designer generates separate automation data for stereo pan and surround pan operations This means that when you use it in surround channels it will not react to existing stereo pan automation data and vice versa Chapter 2 Delay Effects 72 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 x Audio 1 p View v ShowCS Show insert Bypass lt gt default Echo Time pop up menu 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 slider and field Determines how often the delay effect is repeated Color slider and field Sets the harmonic content color of the delay signal Dry and Wet sliders and fields Control the amount of original and effect
42. parameter Envelope parameters Filter parameters Output parameters LFO parameters Distortion parameters Threshold parameter Sets an input level that if exceeded triggers the envelope or LFO which are used to dynamically modulate the filter cutoff frequency See AutoFilter Threshold Parameter Envelope parameters Define how the filter cutoff frequency is modulated over time See AutoFilter Envelope Parameters LFO parameters Define how the filter cutoff frequency is modulated by the LFO See AutoFilter LFO Parameters Filter parameters Control the tonal color of the filtered sound See AutoFilter Filter Parameters Distortion parameters Distort the signal both before and after the filter See AutoFilter Distortion Parameters Output parameters Set the level of both the dry and effect signal See AutoFilter Output Parameters Chapter 6 Filter Effects AutoFilter Threshold Parameter The Threshold parameter analyzes the level of the input signal If the input signal level exceeds the set threshold level the envelope and LFO are retriggered this applies only if the Retrigger button is active The envelope and LFO can be used to modulate the filter cutoff frequency AutoFilter Envelope Parameters The envelope is used to shape the filter cutoff over time When the input signal exceeds the set threshold level the envelope is triggered Attack knob and field Sets the attack time for the envelope
43. ratio between the original dry and effect signals Formant knob and field Shifts the formants of the input signal See Setting Vocal Transformer Pitch and Formant Parameters Glide slider and field Extended Parameters area Determines the amount of time the vocal transformation takes allowing sliding transitions to the set Pitch value Grain Size slider and field Extended Parameters area The Vocal Transformer effect algorithm is based on granular synthesis The Grain Size parameter allows you to set the size of the grains and thus affect the precision of the process Experiment to find the best setting Try Auto first Formants pop up menu Extended Parameters area Determines whether the Vocal Transformer processes all formants Process always setting or only the voiced ones Keep Unvoiced Formants setting The Keep Unvoiced Formants option leaves sibilant sounds in a vocal performance untouched This setting will produce a more natural sounding transformation effect with some signals Detune slider and field Extended Parameters area Detunes the input signal by the set value This parameter is of particular benefit when automated Chapter 10 Pitch Effects 209 210 Setting Vocal Transformer Pitch and Formant Parameters Use the Vocal Transformer s Pitch parameter to transpose the pitch of the signal upward or downward Adjustments are made in semitone steps Incoming pitches are indicated by a v
44. release settings and because the maximum volume is reached more swiftly In addition compression can make a project sound better when played back in different audio environments For example the speakers of a television set or in a car typically have a narrower dynamic range than the sound system in a cinema Compressing the overall mix can help make the sound fuller and clearer in lower fidelity playback situations Compressors are typically used on vocal tracks to make the singing prominent in an overall mix They are also commonly used on music and sound effect tracks but they are rarely used on ambience tracks Some compressors multiband compressors can divide the incoming signal into different frequency bands and apply different compression settings to each band This helps to achieve the maximum level without introducing compression artifacts Multiband compression is typically used on an overall 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 add life to audio signals Limiters Limiters also called peak limiters work in a similar way to compressors in that they reduce the audio signal when it exceeds a set threshold The difference is that whereas a compressor gradually lowers signal levels that exceed the threshold a limiter quickly reduces any signal that is louder than the threshold to the threshold level
45. s M 1008 Delay Tap pads Tap parameter bar Main display Provides a graphic representation of all taps You can see and edit the parameters of each tap in this area See Getting to Know Delay Designer s Main Display Chapter 2 Delay Effects Tap parameter bar Offers a numeric overview of the current parameter settings for the selected tap You can view and edit the parameters of each tap in this area See Editing Taps in Delay Designer s Tap Parameter Bar Tap pads You can use these two pads to create taps in Delay Designer See Creating Taps in Delay Designer e Sync section You can set all Delay Designer synchronization and quantization parameters in this section See Synchronizing Taps in Delay Designer Master section This area contains the global Mix and Feedback parameters See Using Delay Designer s Master Section Getting to Know Delay Designer s Main Display Delay Designer s main display is used to view and edit tap parameters You can freely determine the parameter shown and quickly zoom or navigate through all taps Toggle buttons Tap display View buttons Autozoom button Identification bar Overview display View buttons Determine the parameter or parameters represented in the Tap display See Using Delay Designer s View Buttons Autozoom button Zooms the Tap display out making all taps visible Turn Autozoom off if you want to zoom the display in
46. s Impulse Response Parameters p 229 e Working with Space Designer s Envelope and EQ Parameters p 233 227 228 e Working with Space Designer s Filter p 239 e Working with Space Designer s Global Parameters p 241 e Automating Space Designer p 247 Getting to Know the Space Designer Interface The Space Designer interface consists of the following main sections Impulse response parameters Envelope and EQ Parameters Main display Button bar Input IR Sample sample rate Saris ix Global parameters i om vatcompensation lt Gefinition 100 20s a ess Global parameters Reverb Spread Uolume Envelope Init laval attack time decay time anp lin vendileval er s 002s 1988s o o Filter parameters Parameter bar Impulse response parameters Used to load save or manipulate recorded or synthesized impulse response files The chosen IR file determines what Space Designer will use to convolve with your audio signal See Working with Space Designer s Impulse Response Parameters Envelope and EQ parameters Use the view buttons in the button bar to switch the main display and parameter bar between envelope and EQ views Use the main display to edit the displayed parameters graphically and use the parameter bar to edit them numerically See Working with Space Designer s Envelope and EQ Parameters Filter parameters Used to modify the timbre of the Space Designer rev
47. signal Sample Delay Sample Delay is more a utility than an effect you can use it to delay a channel by single sample values When used in conjunction with the phase inversion capabilities of the Gain effect Sample Delay is useful for correction of timing problems that may occur with multichannel microphones It can also be used creatively to emulate stereo microphone channel separation Chapter 2 Delay Effects Every sample at a frequency of 44 1 kHz is equivalent to the time taken for a sound wave to travel 7 76 millimeters If you delay one channel of a stereo microphone by 13 samples this will emulate an acoustic microphone separation of 10 centimeters tx Inst 1 g View Show Channel Strip Show insert v Bypass lt gt default Sample Delay Delay L 0 sam 0 sam LinkL amp R m Delay slider and field L and R in stereo version Determines the number of samples that the incoming signal will be delayed by Link L amp R button only in stereo version Ensures that the number of samples is identical for both channels Adjusting one channel value will adjust the other Stereo Delay The Stereo Delay works much like the Tape Delay see Tape Delay but allows you to set the Delay Feedback and Mix parameters separately for the left and right channels The Crossfeed knob for each stereo side determines the feedback intensity or the level at which each signal is routed to the opposite stereo side You can f
48. signal If the signal clips the Limiter reduces the level before clipping can occur The Limiter cannot however fix audio that is clipped during recording Gain reduction meter Shows the amount of limiting in real time Gain slider and field Sets the amount of gain applied to the input signal Lookahead slider and field Adjusts how far ahead in milliseconds the Limiter analyzes the audio signal This enables it to react earlier to peak volumes by adjusting the amount of reduction Note Lookahead causes latency but this has no perceptible effect when you use the Limiter as a mastering effect on prerecorded material Set it to higher values if you want the limiting effect to occur before the maximum level is reached thus creating a smoother transition Release slider and field Sets the amount of time after the signal falls below the threshold level before the Limiter stops processing Output Level knob and field Sets the output level of the signal Softknee button When active the signal is limited only when it reaches the threshold The transition to full limiting is nonlinear producing a softer less abrupt effect and reducing distortion artifacts that can be produced by hard limiting Chapter 4 Dynamics Processors 100 Multipressor The Multipressor an abbreviation for multiband compressor is an extremely versatile audio mastering tool It splits the incoming signal into different frequency bands up to
49. signal These added harmonics contain frequencies at least one octave above the threshold of the highpass filter The distorted signal is then mixed with the original dry signal Chapter 13 Specialized Effects and Utilities You can use the Exciter to add life to recordings It is especially well suited to audio tracks with a weak treble frequency range The Exciter is also useful as a general tool for enhancing guitar tracks Frequency display Shows the frequency range used as the source signal for the excite process Frequency slider and field Sets the cutoff frequency in Hertz of the highpass filter The input signal passes through the filter before harmonic distortion is introduced Input button When the Input button is active the original pre effect signal is mixed with the effect signal If you disable Input only the effect signal is heard Harmonics knob and field Sets the ratio between the effect and original signals expressed as a percentage If the Input button is turned off this parameter has no effect Note In most cases higher Frequency and Harmonics values are preferable because human ears cannot easily distinguish between the artificial and original high frequencies Color 1 and Color 2 buttons Color 1 generates a less dense harmonic distortion spectrum Color 2 generates a more intense harmonic distortion Color 2 also introduces more unwanted intermodulation distortions Grooveshifter
50. signal falls below the threshold Release knob and field Sets the amount of time it takes to reach maximum attenuation 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 This prevents the gate from rapidly opening and closing when the input signal is close to the threshold Lookahead slider and field Sets how far ahead the Noise Gate analyzes the incoming signal allowing the effect to respond more quickly to peak levels Monitor button Enable to hear the side chain signal including the effect of the High Cut and Low Cut filters High Cut slider and field Sets the upper cutoff frequency for the side chain signal Low Cut slider and field Sets the lower cutoff frequency for the side chain signal Note When no external side chain is selected the input signal is used as the side chain Chapter 4 Dynamics Processors Using the Noise Gate In most situations setting the Reduction slider to the lowest possible value ensures that sounds below the Threshold value are completely suppressed Setting Reduction to a higher value attenuates low level sounds but still allows them to pass You can also use Reduction to boost the signal by up to 20 dB which is useful for ducking effects The Attack Hold and Release knobs modify the dynamic response of the Noise Gate If you want the gate to open extremely quickly for percussive
51. signals such as drums set the Attack knob to a lower value For sounds with a slow attack phase such as string pads set Attack to a higher value Similarly when working with signals that fade out gradually or that have longer reverb tails set a higher Release knob value that allows the signal to fade out naturally The Hold knob determines the minimum amount of time that the gate stays open You can use the Hold knob to prevent abrupt level changes known as chattering caused by rapid opening or closing of the gate The Hysteresis slider provides another option for preventing chattering without needing to define a minimum Hold time Use it to set the range between the threshold values that open and close the Noise Gate This is useful when the signal level hovers around the Threshold level causing the Noise Gate to switch on and off repeatedly producing the undesirable chattering effect The Hysteresis slider essentially sets the Noise Gate to open at the Threshold level and remain open until the level drops below another lower level As long as the difference between these two values is large enough to accommodate the fluctuating level of the incoming signal the Noise Gate can function without creating chatter This value is always negative Generally 6 dB is a good place to start In some situations you may find that the level of the signal you want to keep and the level of the noise signal are close making it difficult to separa
52. softer tone e Type pop up menu Choose the type of distortion you want to apply Growl Emulates a two stage tube amplifier similar to the type found in a Leslie 122 speaker cabinet which is often used with the Hammond B3 organ Bity Emulates the sound of a bluesy overdriven guitar amp Nasty Produces hard distortion suitable for creating very aggressive sounds Overdrive Overdrive 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 such as those emulated by the Distortion effect Drive slider and field Sets the saturation amount for the simulated transistor Chapter 3 Distortion Effects 81 82 Display Shows the impact of parameters on the signal Tone knob and field Sets the frequency for the high cut filter Filtering the harmonically rich distorted signal produces a softer tone Output slider and field Sets the output level This allows you to compensate for increases in loudness caused by using Overdrive Phase Distortion The Phase Distortion effect is based on a modulated delay line similar to a chorus or flanger effect see Modulation Effects Unlike these effects however the delay time is not modulated by a low frequency oscillator LFO but rather by a lowpass filtered version of the input signal
53. sound As always rely on your ears I O Utility The I O utility enables the use of external audio effects units similar to the use of effects included with Logic Pro Note In practical terms this makes sense only if you are using an audio interface that provides discrete inputs and outputs analog or digital that are used to send signals to and from the external audio effects unit x Aux 1 p View Show Channel Strip Show insert v Bypass Output Volume 0 dB Output Input Input Volume Latency Detection Latency Oftset Output Volume field and slider Adjusts the level of the output signal Output pop up menu Assigns the respective output or output pair of your audio hardware Input pop up menu Assigns the respective input or input pair of your audio hardware Note The Input pop up menu is only visible when an audio interface with multiple inputs is active Input Volume field and slider Adjusts the level of the input signal Latency Detection Ping button Detects the delay between the selected output and input and compensates the delay accordingly Chapter 14 Utilities and Tools 261 262 Note Bypassing any latency inducing plug ins on the track will provide you with the most accurate reading Latency Offset field and slider Displays the value for the detected latency between the selected output and input Also allows you to offset the latency manually To integrate
54. than most classic Wah Wah pedals The sweep range of the Wah Wah filter is set with the Pedal Range parameters The highest and lowest possible values reached by a MIDI foot pedal are graphically represented by a gray bracket around the Pedal Position slider the slider represents the current position of the Wah Wah pedal Chapter 6 Filter Effects You can set the upper and lower limits of the range independently by dragging the left and right handles of the slider bracket You can move the entire range by dragging the center section of the slider bracket Spectral Gate The Spectral Gate is an unusual filter effect that can be used as a tool for creative sound design It works by dividing the incoming signal into two frequency ranges above and below acentral frequency band that you specify with the Center Freq and Bandwidth parameters The signal ranges above and below the defined band can be individually processed with the Low Level and High Level parameters and the Super Energy and Sub Energy parameters See Using the Spectral Gate Spectral Gate Parameters The Spectral Gate panel includes the following parameters Threshold slider Speed CF Modulation and BW Modulation Super Energy and High sliders Graphic display Level controls Gain slider Center Freq and Sub Energy and Low Bandwidth knobs Level controls Threshold slider and field Sets the threshold level for division of frequency ranges When the threshold is exc
55. that carry multichannel audio You can adjust the compression ratio knee attack and release for the main side surround and LFE channels depending on the chosen surround format All channels include an integrated limiter and provide independent threshold and output level controls Chapter 4 Dynamics Processors 107 108 You can link channels by assigning them to one of three groups When you adjust the threshold or output parameter of any grouped channel the parameter adjustment is mirrored by all channels assigned to the group Link section Main section LFE section The Surround Compressor is divided into three sections e The Link section at the top contains a series of menus where you assign each channel to a group See Surround Compressor Link Parameters The Main section includes controls common to all the main channels and the threshold and output controls for each channel See Surround Compressor Main Parameters The LFE section on the lower right includes separate controls for the LFE channel See Surround Compressor LFE Parameters Surround Compressor Link Parameters The Surround Compressor s Link section provides the following parameters Circuit Type pop up menu Choose the type of circuit emulated by the Compressor The choices are Platinum Classic A_R Classic A_U VCA FET and Opto optical Chapter 4 Dynamics Processors Grp Group pop up menus Set group membership for each channel A
56. the Speech Enhancer effect with other microphones but microphone correction models are offered only for built in Macintosh microphones and iSight Should a non Apple microphone be used you will achieve the best results if Mic Correction is set to Generic Voice Enhance button and Enhance Mode pop up menu This button turns on the Speech Enhancer multiband compression circuit When it is active you can choose from four settings that make the recorded voice louder and more intelligible Choose the setting that best matches your recording situation Female or Male Solo Use when the recorded signal consists of a vocal only Female or Male Voice Over Use when the recorded signal contains both a vocal performance and a musical or atmospheric bed Chapter 13 Specialized Effects and Utilities 255 256 SubBass The SubBass plug in generates frequencies below those of the original signal resulting in artificial bass content The simplest use for the SubBass is as an octave divider similar to octaver effect pedals for electric bass guitars Whereas such pedals can only process a monophonic input sound source of clearly defined pitch SubBass can be used with complex summed signals as well See Using SubBass SubBass creates two bass signals derived from two separate portions of the incoming signal These are defined with the High and Low parameters See SubBass Parameters Warning Using SubBass can produce extremely loud out
57. used Working in Space Designer s Synthesized IR Mode In Synthesized IR mode Space Designer generates a synthesized impulse response based on the values of the Length envelope Filter EQ and Spread parameters To switch to this mode click the Synthesized IR button in the Impulse Response Parameters section Repeated clicks of the activated Synthesized IR button will randomly generate new impulse responses with slightly different reflection patterns The current impulse response state including parameter and other values that represent the reflection patterns and characteristics of the synthetic IR is saved with the setting file Note Clicking the Synthesized IR button while you are in IR Sample mode will switch you back to the synthesized IR stored with the setting Setting Space Designer s IR Sample Rate The sample rate slider determines the sample rate of an impulse response eo IR Sample sample rate Orig Space Designer uses the current project sample rate When loading an impulse response Space Designer automatically converts the sample rate of the impulse response to match the current project sample rate if necessary For example this allows you to load a 44 1 kHz impulse response into a project running at 96 kHz and vice versa 2 4 8 These settings are half divisions of the preceding value one half one quarter one eighth For example If the project sample rate is 96 kHz the options will
58. 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 Distribution pop up menu Available only in surround instances Defines how the phase offsets between the individual channels are distributed in the surround field You can choose from circular leftesright fronteorear random and new random distributions Note When you load a setting that uses the random option the saved phase offset value is recalled If you want to randomize the phase setting again choose new random in the Distribution pop up menu Phaser Output Section Output Mix slider and field Determines the balance of dry and wet signals Negative values result in a phase inverted mix of the effect and direct dry signal Warmth button Enables a distortion circuit suitable for warm overdrive effects 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 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 Its sound is often described as metallic or clangoro
59. 2 It has a richer mid range and is more assertive in the treble range 20 Chapter 1 Amps and Pedals Cabinet Description Sunshine 4 x 12 A 4x 12 closed back cabinet with a thick rich mid range Sunshine 1 x 12 A single 12 open back combo amp cabinet with a bright lively sound that has sweet highs and transparent mids Stadium 4 x 12 A tight bright closed back British cabinet with bold upper mid peaks Stadium 2 x 12 A nicely balanced modern British open back cabinet Tonally it is a compromise between the fatness of the Blackface 4 x 10 and the brilliance of the British 2 x 12 Boutique Retro 2 x 12 A 2x 12 cabinet based on the British 2 x 12 It has a rich open mid range and is more assertive in the treble range High Octane 4 x 12 A modern closed back European cabinet with strong lows and highs and scooped mids appropriate for metal and heavy rock Turbo 4 x 12 A modern closed back European cabinet with strong lows very strong highs and deeply scooped mids appropriate for metal and heavy rock Pawnshop 1 x 8 Single 8 speaker cabinet that has excellent low end punch Direct This option bypasses the speaker emulation section Tip For creative sound design select the Direct option place Space Designer in the Insert slot after Amp Designer and then load one of Space Designer s warped speaker impulse responses Using Amp Designer s Equalizer Hardware amplifier tone controls v
60. BP mode the filter cutoff determines the center frequency of the frequency band that is allowed to pass Resonance knob and field Boosts or cuts the signals in the frequency band that surrounds the cutoff frequency Use of very high Resonance values causes the filter to begin oscillating at the cutoff frequency This self oscillation occurs before you reach the maximum Resonance value Fatness slider and field Boosts the level of low frequency content When you set Fatness to its maximum value adjusting Resonance has no effect on frequencies below the cutoff frequency This parameter is used to compensate for a weak or brittle sound caused by high resonance values when in the lowpass filter mode e State Variable Filter buttons Switch the filter between highpass HP bandpass BP or lowpass LP modes e 4 Pole Lowpass Filter buttons Set the slope of the filter to 6 12 18 or 24 dB per octave when the lowpass LP filter is chosen as the State Variable Filter Chapter 6 Filter Effects 135 136 AutoFilter Distortion Parameters The Distortion parameters can be used to overdrive the filter input or filter output The distortion input and output modules are identical but their respective positions in the signal chain before and after the filter respectively result in remarkably different sounds e Input knob and field Sets the amount of distortion applied before the filter section Output knob and field
61. 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 knob and field Frequencies above the set value are filtered from the reverb signal Uneven or absorbent surfaces wallpaper wood paneling carpets and so on tend to reflect lower frequencies better than higher frequencies The High Cut filter mimics this effect If you set the High Cut filter so that it is wide open maximum value the reverb will sound as if it is reflecting off stone or glass Density knob and field Controls the density of the diffuse reverb tail Ordinarily you want the signal to be as dense as possible In rare instances however a high Density value can color the sound which you can fix by reducing the Density knob value Conversely if you select a Density value that is too low the reverb tail will sound grainy Reverb Time knob and field Time it takes for the reverb level to drop by 60 dB often indicated as RT60 Most natural rooms have a reverb time somewhere in the range of 1 to 3 seconds This time is reduced by absorbent surfaces such as carpet and curtains and soft or dense furnishings such as sofas armchairs cupboards and tables Large empty halls or churches have reverb times of up to 8 seconds with some cavernous or cathedral like venues extending beyond that Diffusion slider and field Extended Parameters area Sets the diffusion of the
62. Enable or disable individual EQ bands Frequency fields Set the frequency for the selected EQ band Gain fields Adjust the gain cut or boost for the selected EQ band Q fields Set the Q factor for the two parametric bands The Q factor can be adjusted from 0 1 very narrow to 10 very wide To graphically edit an EQ curve in Space Designer 1 Enable the EQ and one or more bands with the EQ On Off and EQ band buttons in the top row of the parameter bar 238 Chapter 12 Space Designer Convolution Reverb 2 Drag the cursor horizontally over the main display When the cursor is in the access area of a band the corresponding curve and parameter area is automatically highlighted and a pivot point is displayed 3 Drag horizontally to adjust the frequency of the band 4 Drag vertically to increase or decrease the Gain of the band 5 Vertically drag the illuminated pivot point of a parametric EQ band to raise or lower the Q value Working with Space Designer s Filter Space Designer s filter provides control over the timbre of the reverb You can select from several filter types and also have envelope control over the filter cutoff which is independent from the volume envelope Changes to filter settings result in a recalculation of the impulse response rather than a straight change to the sound as it plays through the reverb Chapter 12 Space Designer Convolution Reverb 239 240 Using Space Designer s Main Filter Parame
63. Loads of resonance in the low frequency range therefore well suited for Combos crunchy sounds are also possible with low Bass control settings Chapter 1 Amps and Pedals 31 32 UK 4x 12 closed slanted when used in combination with off center miking you will get an interesting mid frequency range therefore this model works well when combined with High Gain amps US 1 x 10 open back Not much resonance in the low frequency range Suitable for use with blues harmonicas US 1 x 12 open back 1 Open enclosure of an American lead combo with a single 12 speaker US 1 x 12 open back 2 Open enclosure of an American clean crunch combo with a single 12 speaker US 1 x 12 open back 3 Open enclosure of another American clean crunch combo with a single 12 speaker US broad range Simulation of a classic electric piano speaker Analog simulation Internal speaker simulation of a well known British tube preamplifier UK 1 x 12 GarageBand A British Class A tube open back with a single 12 speaker UK 4x 12 GarageBand Classic closed enclosure with four 12 speakers black series suitable for rock US 1 x 12 open back GarageBand Open enclosure of an American lead combo with a single 12 speaker US 1x 12 bass reflex GarageBand Closed bass reflex cabinet with a single 12 speaker DI Box This option allows you to bypass the speaker simulation section Amp Spea
64. Logic Pro 9 Copyright 2011 Apple Inc All rights reserved Your rights to the software are governed by the accompanying software license agreement The owner or authorized user of a valid copy of Logic Pro software may reproduce this publication for the purpose of learning to use such software No part of this publication may be reproduced or transmitted for commercial purposes such as selling copies of this publication or for providing paid for support services The Apple logo is a trademark of Apple Inc registered in the U S and other countries Use of the keyboard Apple logo Shift Option K for commercial purposes without the prior written consent of Apple may constitute trademark infringement and unfair competition in violation of federal and state laws Every effort has been made to ensure that the information in this manual is accurate Apple is not responsible for printing or clerical errors Note Because Apple frequently releases new versions and updates to its system software applications and Internet sites images shown in this manual may be slightly different from what you see on your screen Apple 1 Infinite Loop Cupertino CA 95014 408 996 1010 www ole com Apple the Apple logo Finder GarageBand Logic Macintosh and MainStage are trademarks of Apple Inc registered in the U S and other countries Other company and product names mentioned herein are trademarks of their respective companie
65. MS While Peak is more technically accurate RMS provides a better indication of how people perceive the signal loudness Chapter 4 Dynamics Processors 91 92 Note If you activate Auto Gain and RMS simultaneously the signal may become over saturated If you hear any distortion switch Auto Gain off and adjust the Gain slider until the distortion is inaudible Using a Side Chain with the Compressor Use of a side chain with a compressor is common This allows you to use the dynamics level changes of another channel strip as a control source for compression For example the dynamics of a drum groove can be used to rhythmically change the compression and therefore dynamics of a guitar part Important The side chain signal is used only as a detector or trigger in this situation The side chain source is used to control the Compressor but the audio of the side chain signal is not actually routed through the Compressor To use a side chain with the Compressor Insert the Compressor into a channel strip Select the channel strip that carries the desired signal side chain source in the Side Chain menu of the Compressor plug in Choose the desired analysis method Max or Sum from the Side Chain Detection pop up menu Adjust the Compressor parameters DeEsser The DeEsser is a frequency specific compressor designed to compress a particular frequency band within a complex audio signal It is used to eliminate hiss also called sib
66. PU intensive synthesized IRs Zoom to fit a o definition 100 195s Natural reverbs contain most of their spatial information in the first few milliseconds Toward the end of the reverb the pattern of reflections signals bouncing off walls and so on becomes more diffuse In other words the reflected signals become quieter and increasingly nondirectional containing far less spatial information To emulate this phenomenon as well as to conserve CPU power you can configure Space Designer to use the full IR resolution only at the onset of the reverb and to use a reduced IR resolution toward the end of the reverb The Definition parameter defines the crossover point where the switch to the reduced IR resolution occurs It is displayed in both milliseconds indicating when the crossover occurs and as a percentage 100 is equal to the length of the full resolution IR Chapter 12 Space Designer Convolution Reverb Using Space Designer s Rev Vol Compensation Rev Vol Compensation Reverb Volume Compensation attempts to match the perceived not actual volume differences between impulse response files re rev vol compensation It is enabled by default and should generally be left in this mode although you may find that it isn t successful with all types of impulse responses If this is the case turn it off and adjust input and output levels accordingly Using Space Designer s Output Sliders The output parameters
67. Pedalboard s Routing Area Pedalboard has two discrete signal buses Bus A and Bus B that are found in the Routing area above the Pedal area These busses provide a great deal of flexibility when you are setting up signal processing chains All stompboxes that you drag into the Pedal area are inserted into Bus A by default Note The Routing area appears when you move your pointer to a position immediately above the Pedal area and it disappears when you move the pointer away When you create a second bus routing the Routing area remains open even when your pointer is not over it You can close the Routing area by clicking the small latch button at the top and then the Routing area will open or close automatically when you move your pointer over it To create a second bus routing Do one of the following Move your pointer immediately above the Pedal area to open the Routing area and click the name of a stompbox in the Routing area The pedal name moves upward and the chosen stompbox is routed to Bus B Two gray lines appear in the Routing area which represent Bus A and Bus B A Mixer utility pedal is automatically added to the end of the signal chain Drag a Splitter utility pedal into the Pedal area when more than one pedal is inserted This also inserts a Mixer at the end of the signal chain if one doesn t already exist To remove the second bus routing Do one of the following Remove the Mixer and Splitter utility pedals
68. Phase EQ The high quality Linear Phase EQ effect is similar to the Channel EQ sharing the same parameters and eight band layout The Linear Phase EQ uses a different underlying technology however which perfectly preserves the phase of the audio signal This phase coherency is assured even when you apply the wildest EQ curves to the sharpest signal transients A further difference is that the Linear Phase EQ uses a fixed amount of CPU resources regardless of how many bands are active The Linear Phase EQ also introduces greater amounts of latency Therefore it is strongly recommended that you use it for mastering previously recorded audio and avoid using it when you are playing software instruments live for example Tip The parameters of the Channel EQ and Linear Phase EQ are identical enabling you to freely copy settings between them If you replace a Channel EQ with a Linear Phase EQ or vice versa in the same insert slot the current settings are automatically transferred to the new EQ Chapter5 Equalizers 117 118 Linear Phase EQ Parameters The left side of the Channel EQ window includes the Gain and Analyzer controls The central area of the window includes the graphical display and parameters for shaping each EQ band Linear Phase EQ Gain and Analyzer Controls Master Gain slider and field Sets the overall output level of the signal Use it after boosting or cutting individual frequency bands Analyzer button Turn
69. Pre Post EQ button Determines whether the Analyzer shows the frequency curve before or after EQ is applied when Analyzer mode is active Chapter 5 Equalizers Resolution pop up menu Sets the sample resolution for the Analyzer with the following menu items low 1024 points medium 2048 points and high 4096 points Channel EQ Graphic Display Section Band On Off buttons Click to turn the corresponding band on or off Each button icon indicates the filter type Band 1 is a highpass filter Band 2 is a low shelving filter Bands 3 through 6 are parametric bell filters Band 7 is a high shelving filter Band 8 is a lowpass filter Graphic display Shows the current curve of each EQ band Drag horizontally in the section of the display that encompasses each band to adjust the frequency of the band Drag vertically in the section of the display that encompasses each band to adjust the gain of each band except bands 1 and 8 The display reflects your changes immediately Drag the pivot point in each band to adjust the Q factor Q is shown beside the cursor when it is moved over a pivot point Channel EQ Parameter Section Frequency fields Adjust the frequency of each band Gain Slope fields Set the amount of gain for each band For bands 1 and 8 this changes the slope of the filter Q fields Adjust the Q factor or resonance for each band the range of frequencies around the center frequency that are affecte
70. The Grooveshifter allows you to rhythmically vary audio recordings imparting a swing feel to the input signal 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 automatically follows all changes to the project tempo Chapter 133 Specialized Effects and Utilities 253 254 Note The Grooveshifter relies on perfect matching of the project tempo with the tempo of the treated recording Any tempo variations deliver less precise results Grooveshifter Source Material Parameters Beat and Tonal buttons Switch between two algorithms each optimized for different types of audio material Beat algorithm Optimized for percussive input material The Grain Size slider has no effect when Beat is chosen Tonal algorithm Optimized for tonal input material Because this algorithm is based on granular synthesis it offers an additional Grain Size slider Grain Size slider and field Sets the size of the grains technically speaking this determines the analysis precision The default Auto setting automatically derives a suitable grain size value from the incoming signal Grooveshifter Swing Parameters Grid buttons Determine the beat division used as a timing reference by the algorithm when analyzing the audio material Choose 1 8 if the audio material contains primarily eighth notes and choose 1
71. ack circuit which means that the filtering effect increases in intensity with each delay repeat If you want an increasingly muddy and confused tone move the High Cut slider towards the left For ever thinner echoes move the Low Cut slider towards the right If you re unable to hear the effect even though you seem to have a suitable configuration be sure to check out both the Dry and Wet controls and the filter settings move the High Cut slider to the far right and the Low Cut slider to the far left Smooth slider and field Evens out the LFO and flutter effect LFO Rate knob and field Sets the frequency of the LFO LFO Depth knob and field Sets the amount of LFO modulation A value of 0 turns delay modulation off Flutter Rate and Intensity sliders and fields Simulate the speed irregularities of the tape transports used in analog tape delay units Flutter Rate Sets the speed variation Flutter Intensity Determines how pronounced the effect is Dry and Wet sliders and fields Independently control the amount of original and effect signal Distortion Level slider and field Extended Parameters area Determines the level of the distorted tape saturation signal Chapter 2 Delay Effects Distortion Effects 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 transis
72. active This parameter is grayed out when Beat Sync is disabled Decay Delay knob and field Sets the amount of time it takes for the LFO to go from 0 to its maximum value Rate Mod knob and field Sets the rate of modulation for the LFO frequency independent of the input signal level Typically when the input signal exceeds the threshold the modulation width of the LFO increases from 0 to the Rate Mod value This parameter allows you to override this behavior Stereo Phase knob and field In stereo instances of the AutoFilter sets the phase relationship of the LFO modulations between the two channels Cutoff Mod slider and field Determines the impact of the LFO on the cutoff frequency Retrigger button When the Retrigger button is active the waveform starts at 0 each time the threshold is exceeded Waveform buttons Click one of the following buttons to set the shape of the LFO waveform descending sawtooth ascending sawtooth triangle pulse wave or random Pulse Width slider and field Shapes the curve of the selected waveform Chapter 6 Filter Effects AutoFilter Filter Parameters The Filter parameters allow you to precisely tailor the tonal color e Cutoff knob and field Sets the cutoff frequency for the filter Higher frequencies are attenuated whereas lower frequencies are allowed to pass through in a lowpass filter The reverse is true in a highpass filter When the State Variable Filter is set to bandpass
73. ag it to the left to go forward in time or to the right to go backward in time This method also works when more than one tap is selected Note Editing the Delay Time parameter in the Tap Delay field of the Tap parameter bar also moves a tap in time For more details about the Tap Delay field and editing taps see Editing Taps in Delay Designer s Tap Parameter Bar To delete a tap Do one of the following Select it and press the Delete or Backspace key Chapter 2 Delay Effects 59 60 Select a tap letter in the Identification bar and drag it downward out of the Tap display This method also works when more than one tap is selected To delete all selected taps Control click or right click a tap and choose Delete tap s from the shortcut menu Using Delay Designer s Tap Toggle Buttons Each tap has its own toggle button in the Toggle bar These buttons offer you a quick way to graphically activate and deactivate parameters The specific parameter being toggled by the toggle buttons depends on the current view button selection e Cutoff view Toggle buttons turn the filter on or off Reso view Toggle buttons switch the filter slope between 6 dB and 12 cB Pitch view Toggle buttons switch pitch transposition on or off e Pan view Toggle buttons switch between the Flip modes Level view Toggle buttons mute or unmute the tap To temporarily switch the mute state of taps Command Option click a toggle
74. ameter that you want to control with a MIDI controller Macro A H Value sliders and fields Set and display the current value for the parameter chosen in the corresponding Macro Target pop up menu To assign a Macro A H Target Do one of the following Click any of the Macro A H Target pop up menus and choose the parameter that you want to control Each stompbox parameter is shown in the following way Slot number Pedal Name Parameter As examples Slot 1 Blue Echo Time or Slot 2 Roswell Ringer Feedback The slot number refers to the pedal position as they appear from left to right in the Pedal area Choose the Auto assign item in any Macro A H Target pop up menu then click the appropriate parameter in any inserted pedal Note The chosen parameter is displayed in the Macro A H Target pop up menu Distortion Pedals This section describes the distortion effects pedals Stompbox Description Candy Fuzz A bright nasty distortion effect Drive controls the input signal gain Level sets the effect volume Double Dragon A deluxe distortion effect It offers independent level controls for input Input and output Level Drive controls the amount of saturation applied to the input signal The Tone knob sets the cutoff frequency The Squash knob sets the threshold for the internal compression circuit Contour sets the amount of nonlinear distortion applied to the signal Mix s
75. and classic pedal effects You can play live or process recorded audio and software instrument parts through these amps and effects The amplifier models re create vintage and modern tube and solid state amps Built in effect units such as reverb tremolo or vibrato are also reproduced Accompanying the amplifiers are a variety of emulated speaker cabinets which can be used as a matching set or combined in different ways to create interesting hybrids Also emulated are a number of classic foot pedal effects or stompboxes that were and remain popular with guitarists and keyboardists As with their real world counterparts you can freely chain pedals in any order to create the perfect sound This chapter covers the following e Amp Designer p 11 Bass Amp p 28 e Guitar Amp Pro p 29 e Pedalboard p 35 Amp Designer Amp Designer emulates the sound of over 20 famous guitar amplifiers and the speaker cabinets used with them Each preconfigured model combines an amp cabinet and EQ that re creates a well known guitar amplifier sound You can process guitar signals directly which allows you to reproduce the sound of your guitar played through these amplification systems Amp Designer can also be used for experimental sound design and processing You are free to use it with other instruments applying the sonic character of a guitar amp to a trumpet or vocal part for example The amplifiers cabinets and EQs emula
76. and what you intend to do with it but a useful workflow for many situations is as follows Set the Channel EQ to a flat response no frequencies boosted or cut turn on the Analyzer and play the audio signal Keep an eye on the graphic display to see which parts of the frequency spectrum have frequent peaks and which parts of the spectrum stay at a low level Pay particular attention to sections where the signal distorts or clips Use the graphic display or parameter controls to adjust the frequency bands as desired You can reduce or eliminate unwanted frequencies and you can raise quieter frequencies to make them more pronounced You can adjust the center frequencies of bands 2 through 7 to affect a specific frequency either one you want to emphasize such as the root note of the music or one you want to eliminate such as hum or other noise While doing so change the Q parameter s so that only a narrow range of frequencies are affected or widen it to alter a broad area Each EQ band has a different color in the graphic display You can graphically adjust the frequency of a band by dragging horizontally Drag vertically to adjust the amount of gain for the band For bands 1 and 8 the slope values can be changed only in the parameter area below the graphic display Each band has a pivot point a small circle on the curve at the location of the band s frequency you can adjust the Q or width of the band by dragging the pivot point vertically
77. are found around the main display Input Reset F ilter Env ta Reverse Output Output sliders Input slider Latency Compensation Definition area Rev Vol Compensation button button Input slider Determines how Space Designer processes a stereo or surround input signal For more information see Using Space Designer s Input Slider Latency Compensation button Switches Space Designer s internal latency compensation feature on or off See Using Space Designer s Latency Compensation Feature Definition area Lets you switch to a less defined IR set in order to emulate reverb diffusion and save CPU resources See Using Space Designer s Definition Parameter Rev Vol Compensation button Engages Space Designer s internal IR volume matching function See Using Space Designer s Rev Vol Compensation Output sliders Adjust output levels See Using Space Designer s Output Sliders Space Designer Global Parameters Lower Section These parameters are found below the main display Ss definition 100 200s Reverb Spread Uolume Envelope REN 7 Fi p Pre Diy 5 id init level attack time decay time enp lin endlevel 69 s oos e iss S oo s Tane OANE Pre Dly knob Sets the reverb s predelay time or time between the original signal and the first reflections from the reverb See Working with Pre Dly Predelay in Space Designer IR Start knob Sets the playback start
78. ary between models and manufacturers There s a good chance for example that the treble knobs on two different models target different frequencies or provide different levels of cut or boost Some equalizer EQ sections amplify the guitar signal more than others affecting the way the amp distorts Amp Designer provides multiple EQ types to mirror these variations in hardware amplifiers No matter which EQ type you choose you ll see an identical set of controls Bass Mids and Treble Switching between EQ types can result in these controls behaving very differently Selecting an EQ type other than the one traditionally associated with a certain amplifier typically results in significant tonal changes although these may not necessarily be for the better As with hardware amplifiers Amp Designer s EQs are calibrated to perform well with particular amplifier sounds Choosing other EQ types can sometimes produce a thin or unpleasantly distorted tone See Amp Designer Equalizer Type Reference Table Chapter 1 Amps and Pedals 21 Despite these less pleasant sounding possibilities you should experiment with different amplifier and EQ combinations because many will sound great together EQ pop up menu a ize F ese 872 th 8 0 9 3 10 10 MIDS TREBLE 4 Bass Mids and Treble knobs _4 The EQ parameters include the EQ pop up menu and the Bass Mids and Treble knobs These parameters are found
79. atically enables the main filter Controls the Attack Time endpoint and Decay Time startpoint and Break Level parameters simultaneously Controls the Decay endpoint and End Level parameters simultaneously Filter Enuelope init level attack time break level decay time end level 170 2 Hz Init Level field Sets the initial cutoff frequency of the filter envelope Attack Time field Determines the time required to reach the Break Level see below Break Level field Sets the maximum filter cutoff frequency that the envelope reaches It also acts as the separation point between the attack and decay phases of the overall filter envelope In other words when this level has been reached after the attack phase the decay phase will begin You can create interesting filter sweeps by setting the Break Level to a value lower than the Init Level Decay Time field Determines the time required after the Break Level point to reach the End Level value End Level field Sets the cutoff frequency at the end of the filter envelope decay phase Working with Space Designer s Global Parameters Space Designer s global parameters affect the overall output or behavior of the effect The global parameters are divided into two sections those around the main display and those below the main display Chapter 12 Space Designer Convolution Reverb 241 242 Space Designer Global Parameters Upper Section These parameters
80. ature 4x 12 cabinets At medium gain settings these amps are great for chunky chords and riffs Raising the gain yields lyrical solo tones and powerful rhythm guitar parts Complex peaks and dips across the tonal spectrum keep the tones clear and appealing even when heavy distortion is used Model Description Vintage British Stack Captures the sound of a late 1960s 50 watt amp famed for its powerful smooth distortion Notes retain clarity even at maximum gain After four decades this remains a definitive rock tone Modern British Stack 1980s and 1990s descendants of the Vintage British amplifier head which were optimized for hard rock and metal styles of the time The tones are deeper on the bottom brighter on top and more scooped in the middle than the Vintage British amp Brown Stack Unique tones can be coaxed from a British head by running it at lower voltages than its designers intended The resulting brown sound often more distorted and loose than the standard tone can add interesting thickness to a guitar sound British Blues Combo This 2 x 12 combo has a loud aggressive tone that is cleaner than the British heads yet delivers fat distortion tones at high gain settings Tip You ll rarely go wrong combining a British head a 4 x 12 cabinet and a great riff at high levels But don t hesitate to break that mold These heads can sound stunning through small cabinets or at clean low gain setting
81. available in Amp Designer even when using a model that is based on an amplifier that provides no reverb function Reverb is controlled by an On Off switch and a Level knob in the middle above which is the Reverb pop up menu Reverb can be added to either the Tremolo or Vibrato effect or used independently On Off switch Enables or disables the reverb effect Reverb pop up menu Click the word Reverb to choose one of the following reverb types from the pop up menu Vintage Spring Simple Spring Mellow Spring Bright Spring Dark Spring Resonant Spring Boutique Spring Sweet Reverb Rich Reverb and Warm Reverb See Amp Designer Reverb Type Reference Table for information on these reverb types Chapter 1 Amps and Pedals Level knob Sets the amount of reverb applied to the pre amplified signal Amp Designer Reverb Type Reference Table You can choose a reverb type by clicking the Reverb label in the center of the Amp section The table below covers the properties of each reverb type available in Amp Designer Reverb type Description Vintage Spring This bright splashy sound has largely defined combo amp reverb since the early 1960s Simple Spring A darker subtler spring sound Mellow Spring An even darker somewhat low fidelity spring sound Bright Spring Has some of the brilliance of Vintage Spring but with less surf style splash Dark Spring A moody sounding spring More restrained than Mellow
82. ay not be fed back into the circuit Delay Designer provides up to 26 individual taps These taps are all fed from the source signal and can be freely edited to create delay effects that have never been heard before 51 Delay Designer provides control over the following aspects of each tap e Level and pan position Highpass and lowpass filtering e Pitch transposition up or down Further effect wide parameters include synchronization quantization and feedback As the name implies Delay Designer offers significant sound design potential You can use it for everything from a basic echo effect to an audio pattern sequencer You can create complex evolving moving rhythms by synchronizing the placement of taps This leads to further musical possibilities when coupled with judicious use of transposition and filtering Alternatively you can set up numerous taps as repeats of other taps much as you would use the feedback control of a simple delay but with individual control over each repeat You can use Delay Designer on channel strips with mono stereo or surround inputs and or outputs See Working with Delay Designer in Surround for details on using it in surround channel strips Getting to Know the Delay Designer Interface The Delay Designer interface consists of five main sections Sync section Main display Master section Feedback pitch transp flip pan spread mute leve 2 1000 0ms 2 os 0 s Be DA cemers i00
83. ayback You can however use the integrated conditioning of the Binaural Panner to ensure a neutral sound for speaker or headphone playback 161 162 Note When using multiple Binaural Panners on several channels you should turn the integrated conditioning off and route the output of all binaurally panned signals to an aux channel Insert a Binaural Post Processing plug in into this aux channel and apply diffuse field compensation to all Binaural Panner outputs at once This approach is simpler to manage better sonically and reduces computer processing requirements x Out 1 2 P View Show CS v Show insert Y Bypass Compare gt Binaural Post Processing Compensation Speaker CTC Cross Talk CTC Speaker 70 a Compensation pop up menu Determines the type of processing applied for different playback systems You can choose from Headphone FF optimized for front direction Setting for headphone playback utilizing free field compensation In this mode sound sources placed in front of the listening position will have neutral sound characteristics Headphone HB optimized for horizontal directions Setting for headphone playback Optimized to deliver the most neutral sound for sources placed on or close to the horizontal plane Headphone DF averaged over all directions Setting for headphone playback utilizing diffuse field compensation In this mode the sound will on average be most neutral f
84. back Level knob Sets the feedback level You can vary the feedback tap output level before it is routed back into Delay Designer s input A value of 0 equals no feedback A value of 100 sends the feedback tap back into Delay Designer s input at full volume Chapter 2 Delay Effects Note If Feedback is enabled and you begin creating taps with the Tap pads Feedback is automatically turned off When you stop creating taps with the Tap pads Feedback is automatically re enabled Mix sliders Independently set the levels of the dry input signal and the post processing wet signal Working with Delay Designer in Surround Delay Designer s design is optimized for use in surround configurations With 26 taps that can be freely positioned in the surround field you can create some truly amazing rhythmic and spatial effects Delay Designer always processes each input channel independently Ina mono stereo input and surround output configuration Delay Designer processes the two stereo channels independently and the surround panner lets you place each delay around the surround field Ina surround input and surround output configuration Delay Designer processes each surround channel independently and the surround panner lets you adjust the surround balance of each tap in the surround field When you instantiate Delay Designer in any surround configuration the Pan parameter on the Tap parameter bar is replaced with a surround
85. be perfect for your project EVOC 20 TrackOscillator U V Detection Parameters Human speech consists of a series of voiced sounds tonal sounds or formants and aes sounds the nanonman nasal continuants fricatives and plosives mentioned in A rt Primer on Formants The main distinction between voiced and unvoiced sounds is that d sounds are i by an oscillation of the vocal cords whereas unvoiced sounds are produced by blocking and restricting the air flow with lips tongue palate throat and larynx Chapter 6 Filter Effects 145 146 If speech containing voiced and unvoiced sounds is used as a vocoder s analysis signal but the synthesis engine doesn t differentiate between voiced and unvoiced sounds the result will sound rather weak To avoid this problem the synthesis section of the vocoder must produce different sounds for the voiced and unvoiced parts of the signal The EVOC 20 TrackOscillator includes an Unvoiced Voiced detector for this specific purpose This unit detects the unvoiced portions of the sound in the analysis signal and then substitutes the corresponding portions in the synthesis signal with noise with a mixture of noise and synthesizer signal or with the original signal If the U V Detector detects voiced parts it passes this information to the Synthesis section which uses the normal synthesis signal for these portions e Sensitivity Mode U V Detection e Sensitivity knob Determines how re
86. button regardless of the current view mode When you release the Command and Option keys the toggle buttons return to their standard functionality in the active View mode Chapter 2 Delay Effects _ Note The first time you edit a filter or pitch transpose parameter the respective module automatically turns on This saves you the effort of manually turning on the filter or pitch transposition module before editing After you manually turn either of these modules off however you need to manually switch it back on Editing Parameters in Delay Designer s Tap Display You can graphically edit any tap parameter that is represented as a vertical line in Delay Designer s Tap display The Tap display is ideal if you want to edit the parameters of one tap relative to other taps or when you need to edit multiple taps simultaneously To edit a tap parameter in the Tap display Click the view button of the parameter you want to edit Vertically drag the bright line of the tap you wish to edit or one of the selected taps if multiple taps are selected If you have chosen multiple taps the values of all selected taps will be changed relative to each other Note The method outlined above is slightly different for the Filter Cutoff and Pan parameters See Editing Filter Cutoff in Delay Designer s Tap Display and Editing Pan in Delay Designer s Tap Display To set the values of multiple taps Command drag horizontally and vertically a
87. by dragging vertically on the Overview display to view specific taps Overview display Shows all taps in the time range See Zooming and Navigating Delay Designer s Tap Display Chapter 2 Delay Effects 53 54 Toggle buttons Click to enable or disable the parameters of a particular tap The parameter being toggled is chosen with the view buttons The label at the left of the toggle bar always indicates the parameter being toggled For more information see Using Delay Designer s Tap Toggle Buttons e Tap display Represents each tap as a shaded line Each tap contains a bright bar or dot for stereo panning that indicates the value of the parameter You can directly edit tap parameters in the Tap display area For more details see Editing Parameters in Delay Designer s Tap Display Identification bar Shows an identification letter for each tap It also serves as a time position indicator for each tap You may freely move taps backward or forward in time along this bar timeline See Moving and Deleting Taps in Delay Designer Using Delay Designer s View Buttons The view buttons determine which parameter is represented in Delay Designer s Tap display Cutoff button Shows the highpass and lowpass filter cutoff frequencies of taps Reso nance button Shows the filter resonance value of each tap Transp ose button Shows the pitch transposition of each tap Pan button Shows the pan parameter of each tap For mono
88. cabinets Amp Designer allows you to drive a small speaker with a powerful head or to pair a tiny amp with a 4x 12 cabinet There s nothing wrong with trying random combinations But if you consider the variables that determine a cabinet s sound you ll be able to make educated guesses about non traditional amplifier and cabinet combinations Some factors to consider Combos or Stacks Combo amps include both an amplifier and speakers in a single enclosure These usually have an open back so the sound resonates in multiple directions The resulting sound is open with bright airy highs and a general feeling of spaciousness Amplifier stacks consist of an amplifier head with the speakers in a separate cabinet These cabinets generally have a closed back and project the sound forward in a tight focused beam They tend to sound more powerful than open back cabinets and typically have a tighter low end response at the expense of some high end transparency Old or New Speakers Amp Designer models that are based on vintage cabinets capture the character of aged speakers These may be a bit looser and duller sounding than new speakers but many players prefer them for their smoothness and musicality Sounds based on new cabinets tend to have more snap and bite Large Speakers or Small Speakers A larger speaker doesn t guarantee a larger sound In fact the most popular bass guitar cabinet of all time uses only small 8
89. cale to adjust the overall gain of the filter curve from 30 to 30 dB Chapter5 Equalizers The left scale and the right if the Analyzer is inactive shows the cB values for the filter curve in an appropriate color Single Band EQs The sections below provide descriptions for the following single band EQ effects included in Logic Pro e Low Cut and High Cut Filter e High Pass and Low Pass Filter e High Shelving and Low Shelving EQ Parametric EQ You can find these effects by opening the plug in menu and choosing EQ gt Single Band Low Cut and High Cut Filter The Low Cut Filter attenuates the frequency range that falls below the selected frequency The High Cut Filter attenuates the frequency range above the selected frequency Use the Frequency slider and field to set the cutoff frequency x Inst 1 E Inst 1 gp View v ShowCS v Show insert v gp View ShowCS v Show insert v Bypass lt gt default Bypass lt gt default Low Cut High Cut Frequency 1000Hz Frequency 3900Hz a 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 Chapter 5 Equalizers In contrast the Low Pass Filter affects the frequency range above the selected frequency x Inst 1 Inst 1 p View v ShowCS Show insert v Ca View ShowCS v Show insert
90. ces a powerful crunch sound If you use the British Gain or Modern Gain amps the same Gain setting produces heavy distortion suitable for lead solos Bass Mids and Treble knobs Adjust the frequency range levels of the EQ models similar to the tone knobs on a hardware guitar amplifier Presence knob Adjusts the high frequency range level The Presence parameter affects only the output Master stage of Guitar Amp Pro Master knob Sets the output volume of the amplifier going to the speaker For tube amplifiers increasing the Master level typically produces a more compressed and saturated sound resulting in a more distorted and powerful that is louder signal High Master settings can produce an extremely loud output that can damage your speakers or hearing so ramp this up slowly In Guitar Amp Pro the Master parameter modifies the sonic character and the final output level is set using the Output parameter at the bottom of the interface See Setting the Guitar Amp Pro Output Level Getting to Know Guitar Amp Pro s Effects Section The effects parameters include Tremolo Vibrato and Reverb which emulate the processors found on many amplifiers You can use the pop up menu to choose either Tremolo which modulates the amplitude or volume of the sound or Vibrato which modulates the pitch Reverb can be added to either of these effects or used independently To use or adjust an effect you must first enable it by cl
91. cessors can help Denoiser eliminates or reduces noise below a threshold level Enhance Timing enhances the timing of audio recordings Exciter can add life to your recordings by generating artificial high frequency components Grooveshifter enabes you to create rhythmic variations in your recordings SubBass generates an artificial bass signal that is derived from the incoming signal This chapter covers the following Denoiser p 249 Enhance Timing p 251 Exciter p 252 Grooveshifter p 253 Speech Enhancer p 255 SubBass p 256 Denoiser The Denoiser eliminates or reduces any noise below a threshold volume level The Denoiser uses fast Fourier transform FFT analysis to recognize frequency bands of lower volume and less complex harmonic structure It then reduces these low level less complex bands to the appropriate dB level See Denoiser Main Parameters If you use the Denoiser too aggressively however the algorithm produces artifacts which are usually less desirable than the existing noise If using the Denoiser produces these artifacts you can use the three Smoothing knobs to reduce or eliminate them See Denoiser Smoothing Parameters To use the Denoiser Locate a section of the audio where only noise is audible and set the Threshold value so that only signals at or below this level are filtered out 249 2 Play the audio signal and set the Reduce value to the point where noise reduction is optimal but lit
92. ch EQ p 121 e Single Band EQs p 127 e Silver EQ p 129 111 Channel EQ The Channel EQ is a highly versatile multiband EQ It provides eight frequency bands including lowpass 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 may need adjustment You can use the Channel EQ to shape the sound of individual tracks or audio files or for tone shaping on an overall project mix The Analyzer and graphic controls make it easy to view and change the audio signal in real time Tip The parameters of the Channel EQ and Linear Phase EQ are identical enabling you to freely copy settings between them If you replace a Channel EQ with a Linear Phase EQ or vice versa in the same Insert slot the current settings are automatically transferred to the new EQ Channel EQ Parameters The left side of the Channel EQ window features the Gain and Analyzer controls The central area of the window includes the graphic display and parameters for shaping each EQ band Analyzer Channel EQ Gain and Analyzer Controls Master Gain slider and field Sets the overall output level of the signal Use it after boosting or cutting individual frequency bands Analyzer button Turns the Analyzer on or off
93. chord This allows subtle or savage pitch corrections and can be used creatively on unpitched material with high harmonic content such as cymbals and high hats Pitch Quantize Strength slider Determines how pronounced the automatic pitch correction is Chapter 6 Filter Effects Pitch Quantize Glide slider Sets the amount of time the pitch correction takes allowing sliding transitions to quantized pitches Root Scale keyboard and pop up menu Define the pitch or pitches that the tracking oscillator is quantized to Max Track value field Sets the highest frequency All frequencies above this threshold are cut making pitch detection more robust If pitch detection produces unstable results reduce this parameter to the lowest possible setting that allows all appropriate input signals to be heard or processed Quantizing the Pitch of the Tracking Oscillator You can use the Root Scale keyboard and pop up menu to define the pitch or pitches that the tracking oscillator is quantized to To choose a root or scale Click the green value field below the Root Scale label to open the pop up menu Choose the scale or chord that you want to use as the basis for pitch correction Note You can also set the root key of the respective scale or chord by vertically dragging the Root value field or by double clicking it and entering a root between C and B The Root parameter is not available when the Root Scale value is set to chromatic
94. cross several taps in the Tap display Chapter 2 Delay Effects 62 Parameter values change to match the mouse position as you drag across the taps Command dragging across several taps allows you to draw value curves much like using a pencil to create a curved line on a piece of paper Aligning Delay Designer Tap Values You can use Delay Designer s Tap display to graphically align tap parameter values that are represented as vertical lines To align the values of several taps 1 Command click in the Tap display and move the pointer while holding down the Command key This will result in a line trailing behind the pointer aimee eeeey Tar 2 Click the appropriate position to mark the end point of the line Chapter 2 Delay Effects The values of taps that fall between the start and end points are aligned along the line Editing Filter Cutoff in Delay Designer s Tap Display Whereas the steps outlined in Editing Parameters in Delay Designer s Tap Display apply to most graphically editable parameters the Cutoff and Pan parameters work in a slightly different fashion In Cutoff view each tap actually shows two parameters highpass and lowpass filter cutoff frequency The filter cutoff values can be adjusted independently by dragging the specific cutoff frequency line the upper line is lowpass and the lower line is highpass or both cutoff frequencies can be adjusted by dragging between them When the highpass filt
95. d Note The Q parameter of Band 1 and Band 8 has no effect when the slope is set to 6 dB Oct When the Q parameter is set to an extremely high value such as 100 these filters affect only a very narrow frequency band and can be used as notch filters e Link button Activates Gain Q coupling which automatically adjusts the Q bandwidth when you raise or lower the gain on any EQ band to preserve the perceived bandwidth of the bell curve Analyzer Mode buttons Extended Parameters area Choose Peak or RMS Analyzer Decay slider and field Extended Parameters area Adjust the decay rate in dB per second of the Analyzer curve peak decay in Peak mode or an averaged decay in RMS mode Chapter5 Equalizers 113 114 Gain Q Couple Strength pop up menu Extended Parameters area Choose the amount of Gain Q coupling e Choose strong to preserve most of the perceived bandwidth Choose light or medium to allow some change as you raise or lower the gain The asymmetric settings feature a stronger coupling for negative gain values than for positive values so the perceived bandwidth is more closely preserved when you cut rather than boost gain Note If you play back automation of the Q parameter with a different Gain Q Couple setting the actual Q values will be different than when the automation was recorded Using the Channel EQ The way you use the Channel EQ is obviously dependent on the audio material
96. d bright clean sounds Boutique Retro Combo A 2x 12 combo inspired by high end modern amps that combine the sounds of several great 1960s combos It excels at shimmering clean tones and crunch tones making it a good choice when you want an old fashioned flavor but with the crisp highs and defined lows of a modern amplifier Pawnshop Combo A 1x8 combo based on the inexpensive amps sold in American department stores in the 1960s Despite their limited features and budget workmanship these amps are the secret behind the sound of many rock blues and punk players The clean sounds are warm and distorted sounds are thick and satisfying despite the small speaker Transparent Preamp As the name suggests a preamp stage with no coloration You should note that the Transparent Preamp is activated in the Amp pop up menu not in the Model pop up menu Tip Try pairing the Studio Combo amp with one of the 4x 12 cabinets for a heavier sound The Boutique Retro Amp has very sensitive tone controls providing countless tonal shadings Even extreme settings can yield great results Combine the Pawnshop Combo amp with Pedalboard s Hi Drive or Candy Fuzz stompboxes to emulate hard rock tones of the late 1960s See Distortion Pedals and Pedalboard Chapter 1 Amps and Pedals 17 18 Building a Customized Amp Designer Combo You can use one of the default models or you can create your own hybrid of different amplifiers cabinets and so
97. d field Sets the amount of gain applied to the output signal Auto Gain pop up menu Choose a value to compensate for volume reductions caused by compression The choices are Off 0 dB and 12 dB Limiter Threshold slider and field Sets the threshold level for the limiter Limiter button Turns the integrated limiter on or off Output Distortion pop up menu Extended Parameters area Choose whether to apply clipping above 0 dB and the type of clipping Choices are Off Soft Hard and Clip Activity pop up menu Extended Parameters area Enables or disables the side chain Choices are Off Listen and On Mode pop up menu Extended Parameters area Choose the type of filter used for the side chain Choices are LP lowpass BP bandpass HP highpass ParEQ parametric and HS high shelving Frequency slider and field Extended Parameters area Sets the center frequency for the side chain filter Q slider and field Extended Parameters area Sets the width of the frequency band affected by the side chain filter Gain slider and field Extended Parameters area Sets the amount of gain applied to the side chain signal Mix slider and field Extended Parameters area Determines the balance between dry source and wet effect signals Using the Compressor The following section explains how to use the main Compressor parameters Setting the Compressor Threshold and Ratio The most important Compressor
98. dB and 80 dB The Analyzer display is always dB linear Note When choosing a resolution be aware that higher resolutions require significantly more processing power High resolution is necessary when trying to obtain accurate analysis of very low bass frequencies for example It is recommended that you disable the Analyzer or close the Channel EQ window after setting the appropriate EQ parameters This will free up CPU resources for other tasks DJ EQ The DJ EQ combines high and low shelving filters each with a fixed frequency and one parametric EQ You can adjust the Frequency Gain and Q Factor of the latter The DJ EQ allows the filter gain to be reduced by as much as 30 GB x Inst 1 p View ShowCS v Show insert v Bypass lt gt default DJ EQ High Shelf 0 0d8 Frequency 920Hz 0 7 0 0dB 0 0dB High Shelf slider and field Sets the amount of gain for the high shelving filter Frequency slider and field Sets the center frequency of the parametric EQ Q Factor slider and field Sets the range bandwidth of the parametric EQ Gain slider and field Sets the amount of gain for the parametric EQ Chapter5 Equalizers 115 Low Shelf slider and field Sets the amount of gain for the low shelving filter Fat EQ The Fat EQ is a versatile multiband EQ which can be used on individual sources or overall mixes The Fat EQ provides up to five individual frequency bands graphically displays EQ curves
99. ds the threshold Chapter 4 Dynamics Processors 109 110 Release knob and field Sets the amount of time it takes to return to 0 compression after the signal falls below the threshold Auto button When the Auto button is enabled the release time dynamically adjusts to the audio material Limiter button Turns limiting for the main channels on or off Threshold knob and field Sets the threshold for the limiter on the main channels Main Compressor Threshold sliders and fields Set the threshold level for each channel including the LFE channel which also has independent controls Main Output Levels sliders and fields Set the output level for each channel including the LFE channel which has independent controls Surround Compressor LFE Parameters The Surround Compressor s LFE section provides the following parameters Ratio knob and field Sets the compression ratio for the LFE channel Knee knob and field Sets the knee for the LFE channel Attack knob and field Sets the attack time for the LFE channel Release knob and field Sets the release time for the LFE channel Auto button When the Auto button is enabled the release time automatically adjusts to the audio signal Threshold knob and field Sets the threshold for the limiter on the LFE channel Limiter button Enables and disables limiting for the LFE channel Chapter 4 Dynamics Processors Equalizers 5 An equalizer commonly abbreviated a
100. e Pedal area Notes on Splitter and Mixer Utility Use Dragging a Splitter utility into the Pedal area automatically inserts a Mixer utility to the far right of all inserted pedals You cannot drag a Splitter utility to the far right of all inserted pedals to directly after an inserted Splitter utility to directly in front of an inserted Mixer utility or to an empty space in the Pedal area Dragging a Mixer utility into the Pedal area automatically creates a split point at the earliest possible the leftmost point within the signal chain You cannot drag a Mixer utility to the first slot in the Pedal area to between an inserted Splitter and Mixer utility combo or directly to the right of an inserted Mixer utility Using Pedalboard s Macro Controls Area Pedalboard provides eight Macro Targets A through H which are found in the Macro Controls area below the Pedal area These enable you to map any parameter of an inserted stompbox as a Macro A H target You can save different mappings with each Pedalboard setting Chapter 1 Amps and Pedals You use a controller assignment or create a Workspace knob for Macro A H Value MIDI hardware switches sliders or knobs can then be used to control the mapped Pedalboard Macro A H target parameters in real time See the Logic Pro User Manual for details Click the triangle at the bottom left to hide or show the Macro Controls area Macro A H Target pop up menus Determine the par
101. e and dramatic changes to the frequency spectrum Chapter 6 Filter Effects Getting to Know the EVOC 20 TrackOscillator Interface The EVOC 20 TrackOscillator window is divided into several parameter sections Synthesis In parameters Analysis In parameters Formant Filter parameters ma a Syathesis in aid Output parameters 6 Mode U V Detection Tracking Oscillator LFO parameters U V Detection parameters parameters Analysis In parameters Determine how the input signal is analyzed and used by the analysis filter bank See EVOC 20 TrackOscillator Analysis In Parameters U V Detection parameters Detect the unvoiced portions of the sound in the analysis signal improving speech intelligibility See EVOC 20 TrackOscillator U V Detection Parameters e Synthesis In parameters Determine how the input signal is used by the synthesis filter bank See EVOC 20 TrackOscillator Synthesis In Parameters Tracking Oscillator parameters Determine how the analysis input signal is used by the oscillator See Basic Tracking Oscillator Parameters Formant Filter parameters Configure the analysis and synthesis filter banks See EVOC 20 TrackOscillator Formant Filter Parameters Modulation parameters Modulate either the oscillator pitch or the Formant Shift parameter See EVOC 20 TrackOscillator Modulation Parameters Output parameters Configure the output signal of the EVOC 20 TrackOscillator See EVOC 20 TrackOscilla
102. e file the Length parameter value cannot exceed the length of the actual impulse response sample Longer impulse responses sampled or synthesized place a higher strain on the CPU Working with Space Designer s Envelope and EQ Parameters Space Designer s main interface area is used to show and edit envelope and EQ parameters It consists of three components the button bar at the top the main display and the parameter bar The button bar is used to choose the current view edit mode e The main display shows and allows you to graphically edit either the envelope or the EQ curve The parameter bar displays and allows you to numerically edit either the envelope or the EQ curve Button bar Display in Envelope view Display in EQ view Fefinition 100 is Gefinition 100 200s Vale ENGINE a init level attack time decay time enp lin end level Freq 69H 300H2 2000H2 5000H Gain 8 866dB 125dB 35dB 2508 69 s oms 16s 3 ox s Q 40 s 130 Main display Parameter bar Using Space Designer s Button Bar The button bar is used to switch the main display and parameter bar between envelope and EQ views It also includes buttons that reset the envelopes and EQ or reverse the IR e Reset button Resets the currently displayed envelope or EQ to its default values Chapter 12 Space Designer Convolution Reverb 233 234 e All button Resets all e
103. e filtered Suppressor signal Sens itivity to remove the sound from the input signal in response to the Sensitivity parameter or Off to hear the DeEsser output DeEsser Suppressor Section Suppressor Frequency knob and field Sets the frequency band that is reduced when the Detector sensitivity threshold is exceeded Strength knob and field Sets the amount of gain reduction for signals that surround the Suppressor frequency e Activity LED Indicates active suppression in real time DeEsser Center Section Detector and Suppressor frequency displays The upper display shows the Detector frequency range The lower display shows the Suppressor frequency range in Hz e Smoothing slider Sets the reaction speed of the gain reduction start and end phases Smoothing controls both the attack and release times as they are used by compressors Chapter 4 Dynamics Processors 93 94 Ducker Ducking is acommon technique used in radio and television broadcasting When the DJ or 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 Ducker provides a simple means of achieving this result with existing recordings It does not work in real time Note For technical reasons Ducker can only be inserted in output and aux channel strips Ducker Parameters The Ducker has the following parameters Ducking Lookah
104. e formants independently which means that you can turn a vocal track into a Mickey Mouse voice while maintaining the original pitch Formants are characteristic emphases of certain frequency ranges They are static and do not change with pitch Formants are responsible for the specific timbre of a given human voice The Vocal Transformer is well suited to extreme vocal effects The best results are achieved with monophonic signals including monophonic instrument tracks It is not designed for polyphonic voices such as a choir on a single track or other chordal tracks Chapter 10 Pitch Effects Vocal Transformer Parameters Vocal Transformer offers the following parameters vocal transformer Robotize ow G s Track Formant Pitch knob and field Determines the amount of transposition applied to the input signal See Setting Vocal Transformer Pitch and Formant Parameters Robotize button Enables Robotize mode which is used to augment diminish or mirror the melody See Using Vocal Transformer s Robotize Mode Pitch Base slider and field available only in Robotize mode Use to transpose the note that the Tracking parameter see below is following See Using Vocal Transformer s Robotize Mode Tracking slider field and buttons available only in Robotize mode Control how the melody is changed in Robotize mode See Using Vocal Transformer s Robotize Mode Mix slider and field Defines the level
105. e movements To change Space Designer s envelope curve shape graphically Drag the envelope curve in the main display Drag the small nodes attached to a line for fine adjustments to envelope curves These nodes are tied to the envelope curve itself so you can view them as envelope handles Synthesized IR 2 000s Move the nodes vertically or horizontally to change the shape of the N envelope curve Chapter 12 Space Designer Convolution Reverb 235 236 Working with Space Designer s Volume Envelope The volume envelope is used to set the reverb s initial level and adjust how the volume will change over time You can edit all volume envelope parameters numerically and many can also be edited graphically see Setting Space Designer s Envelope Parameters synthesized IR 2 000s Pe Init Level node Decay Time End Level node Zoom to fit A D Attack Decay Time node definition 100 2 00s D lt Uolume Envelope init level attack time decay time enp lin end level 69 s 0 23s 125s 3 19 s Init Level field Sets the initial volume level of the impulse response attack phase It is expressed as a percentage of the full scale volume of the impulse response file The attack phase is generally the loudest point of the impulse response Set Init Level to 100 to ensure maximum volume for the early reflections Attack Time field Determines the length of time before the decay phase of
106. e output parameters on the right Graphic display section Frequency band section Output section Multipressor Graphic Display Section Graphic display Each frequency band is represented graphically The amount of gain change from 0 dB is indicated by blue bars The band number appears in the center of active bands You can adjust each frequency band independently in the following ways Drag the horizontal bar up or down to adjust the gain makeup for that band Drag the vertical edges of a band to the left or right to set the crossover frequencies which adjusts the band s frequency range Crossover fields Set the crossover frequency between adjacent bands Gain Make up fields Set the amount of the gain make up for each band Multipressor Frequency Band Section Compr ession Thrsh old fields Set the compression threshold for the selected band Setting the parameter to 0 dB results in no compression of the band Compr ession Ratio fields Set the compression ratio for the selected band Setting the parameter to 1 1 results in no compression of the band Chapter 4 Dynamics Processors 101 102 Expnd Thrsh old fields Set the expansion threshold for the selected band Setting the parameter to its minimum value 60 dB means that only signals that fall below this level are expanded Expnd Ratio fields Set the expansion ratio for the selected band Expnd Reduction fields Set the amount of downward expa
107. e working on the overall surround mix Main display Analyzer parameters Goniometer parameters Goniometer shown Analyzer Balance Correlation F MEER som Reset Peak parameters Balance Correlation button Using the Surround MultiMeter Analyzer In Analyzer mode the MultiMeter s main display shows the frequency spectrum of the input signal as 31 independent frequency bands Each frequency band represents one third of an octave The Analyzer parameters are used to activate Analyzer mode and to customize the way that the incoming signal is shown in the main display PER Both Balance Correlation Analyzer parameters Scale Analyzer button Switches the main display to Analyzer mode Chapter 8 Metering Tools 175 176 Sum and Max buttons Determine whether a summed or maximum level is displayed in the Analyzer results in the main display These buttons are relevant only when multiple channels are selected with the channel buttons e Channel buttons Used to select a single channel or a combination of channels for metering The number and appearance of these buttons varies when different surround modes are chosen View fields Alter the way that values are shown in the Analyzer by setting the maximum level displayed Top and the overall dynamic range Range Mode buttons Determine how levels are displayed You can choose from Peak Slow RMS or Fast RMS characteristics The two RMS modes
108. ead a Amount Threshold Attack Release Ducking On and Off buttons Enable or disable ducking Lookahead On and Off buttons Enable to ensure that the Ducker reads the incoming signal before processing This results in no latency it is primarily intended for slower computers Amount slider and field Defines the amount of volume reduction of the music mix channel strip which is in effect the output signal Threshold slider and field Determines the lowest level that a side chain signal must attain before it begins to reduce the music mix output level by the amount set with the Intensity slider If the side chain signal level doesn t reach the threshold the music mix channel strip volume is not affected Chapter 4 Dynamics Processors Attack slider and field Controls how quickly the volume is reduced If you want the music mix signal to be gently faded out set this slider to a high value This value also controls whether or not the signal level is reduced before the threshold is reached The earlier this occurs the more latency is introduced Note This only works if the ducking signal is not live the ducking signal must be an existing recording The host application needs to analyze the signal level before it is played back in order to predefine the point where ducking begins Hold slider and field Determines the duration for which the music mix channel strip volume is reduced This control prev
109. early 1960s that helped define the sounds of blues rock and country music They have warm complex clean sounds that progress smoothly through gentle distortion to raucous overdrive as you increase the gain Even after half a century Tweeds can still sound contemporary Many modern boutique amplifiers are based on Tweed style circuitry Model Description Small Tweed Combo A 1x 12 combo that transitions smoothly from clean to crunchy making it a great choice for blues and rock For extra definition set the Treble and Presence controls to a value around 7 Large Tweed Combo This 4 x 10 combo was originally intended for bassists but was also used by blues and rock guitarists More open and transparent sounding than the Small Tweed Combo but can deliver crunchy sounds Mini Tweed Combo A small amp with a single 10 speaker used by countless blues and rock artists It is quite punchy sounding and can deliver the clean and crunch tones that the Tweed combos are known for Tip Tweed combos respond beautifully to your playing dynamics Adjust the knobs to create a distorted sound then reduce the level of your guitar s volume knob to create a cleaner tone Turn up your guitar s volume knob when the time comes for a scorching solo Classic American Combos The Blackface Brownface and Silverface models are inspired by American combos of the mid 1960s These tend to be loud and clean with tight lows and relatively restrained
110. eases the dynamic range below the Threshold the Ratio slider features a value range of 1 1 to 0 5 1 e Knee slider and field Determines the strength of expansion at levels close to the threshold Lower values result in more severe or immediate expansion hard knee Higher values result in a gentler expansion soft knee Gain slider and field Sets the amount of output gain Auto Gain button Compensates for the level increase caused by expansion When Auto Gain is active the signal sounds softer even when the peak level remains the same Note If you dramatically change the dynamics of a signal with extreme Threshold and Ratio values you may need to reduce the Gain slider level to avoid distortion In most cases turning on Auto Gain will adjust the signal appropriately Chapter 4 Dynamics Processors Limiter The Limiter works much like a compressor but with one important difference where a compressor proportionally reduces the signal when it exceeds the threshold a limiter reduces any peak above the threshold to the threshold level effectively limiting the signal to this level The Limiter is used primarily when mastering Typically you apply the Limiter as the very last process in the mastering signal chain where it raises the overall volume of the signal so that it reaches but does not exceed 0 dB The Limiter is designed in such a way that if set to 0 dB Gain and 0 dB Output Level it has no effect on a normalized
111. ecisely control the signal at any point in the signal chain Chapter 1 Amps and Pedals 35 All stompbox knobs switches and sliders can be automated Eight Macro controls enable real time changes to any pedal parameter with a MIDI controller Routing area Pedal area Pedalboard m acro 8 Target RORe Macro Controls area Pedal Browser The Pedal Browser shows all pedal effects and utilities These can be dragged into the Pedal area as part of the signal chain See Using Pedalboard s Pedal Browser This interface area is also used for the alternative import mode See Using Pedalboard s Import Mode The Pedal area is where you determine the order of effects and set effect parameters You can add replace and remove stompboxes here See Using Pedalboard s Pedal Area The Routing area is used to control signal flow in the two effects busses Bus A and Bus B available in Pedalboard See Using Pedalboard s Routing Area The Macro Controls area is used to assign eight MIDI controllers which can be used to control any stompbox parameter in real time See Using Pedalboard s Macro Controls Area The effect and utility pedals are described in the following sections e Distortion Pedals e Modulation Pedals Delay Pedals e Filter Pedals e Dynamics Pedals e Utility Pedals Chapter 1 Amps and Pedals Using Pedalboard s Pedal Browser Pedalboard offers dozens of pedal effects and utilities in the Pedal
112. ectional but any polarity can be used are equally spaced from the center and pointed directly at the sound source Spacing between microphones is extremely important for the overall stereo width and perceived positioning of instruments within the stereo field The AB technique is commonly used for recording one section of an orchestra such as the string section or perhaps a small group of vocalists It is also useful for recording piano or acoustic guitar AB is not well suited to recording a full orchestra or group as it tends to smear the stereo imaging positioning of off center instruments It is also unsuitable for mixing down to mono as you run the risk of phase cancellations between channels Chapter 7 Imaging Processors Understanding XY Miking In an XY recording two directional microphones are symmetrically angled from the center of the stereo field The right hand microphone is aimed at a point between the left side and the center of the sound source The left hand microphone is aimed at a point between the right side and the center of the sound source This results in a 45 to 60 off axis recording on each channel or 90 to 120 between channels XY recordings tend to be balanced in both channels with good positional information being encoded It is commonly used for drum recording XY recording is also suitable for larger ensembles and many individual instruments Typically XY recordings have a narrower sound field
113. ects pedals 44 Chapter 1 Amps and Pedals Stompbox Description Heavenly Chorus A rich sweet sounding chorus effect that can significantly thicken the sound Rate sets the modulation speed and can run freely or be synchronized with the host application tempo by enabling the Sync button When synchronized you can specify bar beat and note values including triplets and dotted notes Depth sets the strength of the effect Feedback sends the output of the effect back in to the input further thickening the sound or leading to intermodulations Delay sets the ratio between the original and effect signals The upper Bright switch position applies a fixed frequency internal EQ to the signal At the bottom position the EQ is bypassed Phase Tripper A simple phasing effect Rate sets the modulation speed and can run freely or be synchronized with the host application tempo by enabling the Sync button When synchronized you can specify bar beat and note values including triplets and dotted notes Depth sets the strength of the effect Feedback determines the amount of the effect signal that is routed back into the input This can change the tonal color can make the sweeping effect more pronounced or can do both Phaze 2 A very flexible dual phaser effect LFO 1 and LFO2 Rate sets the modulation speed and can run freely or be synchronized with the host application tempo by enabling the Sync button Ceiling and Floor determine the f
114. ed with the other surround channels Surround 0 setting bottom of slider The entire LFE signal is passed through the reverb unprocessed In between positions A mixture of LFE and surround channel information is processed Chapter 12 Space Designer Convolution Reverb 243 244 Using Space Designer s Latency Compensation Feature The complex calculations made by Space Designer take time This time results in a processing delay or latency between the direct input signal and the processed output signal When activated the Latency Compensation feature delays the direct signal in the Output section to match the processing delay of the effect signal Note This is not related to latency compensation in the host application This compensation feature occurs entirely within Space Designer latency compensation o Space Designer s processing latency is 128 samples at the original sample rate and it doubles at each lower sample rate division If you set Space Designer s sample rate slider to 2 the processing latency increases to 256 samples Processing latency does not increase in surround mode or at sample rates above 44 1 kHz Using Space Designer s Definition Parameter The Definition parameter emulates the diffusion of natural reverb patterns When used at values of less than 100 it also reduces CPU processing requirements Note The Definition steppers are visible below the main display only when you have loaded C
115. eeded the frequency band defined by the Center Freq and Bandwidth parameters is divided into upper and lower frequency ranges Speed slider and field Sets the modulation frequency for the defined frequency band CF Center Frequency Modulation slider and field Sets the intensity of center frequency modulation BW Band Width Modulation slider and field Sets the amount of bandwidth modulation Graphic display Shows the frequency band defined by the Center Freq and Bandwidth parameters Chapter 6 Filter Effects 157 158 Center Freq Frequency knob and field Sets the center frequency of the band that you want to process Bandwidth knob and field Sets the width of the frequency band that you want to process Super Energy knob and field Controls the level of the frequency range above the threshold High Level slider and field Blends the frequencies of the original signal above the selected frequency band with the processed signal Sub Energy and field Controls the level of the frequency range below the threshold Low Level slider and field Blends the frequencies of the original signal below the selected frequency band with the processed signal Gain slider and field Sets the output level of the Spectral Gate Using the Spectral Gate One way to familiarize yourself with the operation of the Spectral Gate would be to start with a drum loop Set the Center Freq to its minimum 20 Hz and the Bandwidt
116. enable you to adjust the balance between the direct dry and processed signals The parameters that are available are dependent on Space Designer s input configuration If you insert Space Designer as mono mono to stereo or stereo effect Space Designer offers two output sliders one for the direct signal and one for the reverb signal In surround configurations Space Designer offers four output sliders that together comprise a small surround output mixer Mono Stereo Surround C Bal Rev Dry T T max Space Designer Mono Stereo Output Configuration Parameters Dry slider Sets the level of the non effect dry signal Set this to a value of 0 mute if Space Designer is inserted in a bus channel or when using modeling impulse responses such as speaker simulations Rev erb slider Adjusts the output level of the effect wet signal Chapter 12 Space Designer Convolution Reverb 245 246 Space Designer Surround Output Configuration Parameters Center slider Adjusts the output level of the center channel independently of other surround channels e Bal ance slider Sets the level balance between the front L C R and rear Ls Rs channels In 71 ITU surround the balance pivots around the Lm Rm speakers taking the surround angles into account With 71 SDDS surround the Lc Rc speakers are considered front speakers Rev erb slider Adjusts the output level of the effect wet signal for all channel
117. encies that fall outside these boundaries will be cut The length of the horizontal blue bar at the top represents the frequency range You can move the entire frequency range by dragging the blue bar The silver handles on either end of the blue bar set the Low Frequency and High Frequency values respectively You can also use the numeric fields to adjust the frequency values separately Frequency band faders Set the level of each frequency band in Filter Bank A upper blue faders or Filter Bank B lower green faders You can quickly create complex level curves by dragging horizontally drawing across either row of faders Formant Shift knob Moves all bands in both filter banks up or down the frequency spectrum Note The use of Formant Shift can result in the generation of unusual resonant frequencies when high Resonance settings are used Bands value field Sets the number of frequency bands up to 20 in each filter bank e Lowest button Click to determine whether the lowest filter band acts as bandpass or highpass filter In the Bandpass setting the frequencies below the lowest bands and above the highest bands are ignored In the Highpass setting all frequencies below the lowest bands are filtered Chapter 6 Filter Effects Highest button Click to determine whether the highest filter band acts as bandpass or lowpass filter In the Bandpass setting the frequencies below the lowest bands and above the h
118. ents a chattering effect that can be caused by a rapidly changing side chain level If the side chain level hovers around the threshold value rather than clearly exceeding or falling short of it set the Hold parameter to a high value to compensate for any rapid volume reductions Release slider and field Controls how quickly the volume returns to the original level Set it to a high value if you want the music mix to slowly fade up after the announcement Using the Ducker The steps below show how to use the Ducker on existing recordings Note For technical reasons the Ducker plug in can be inserted only in output and aux channel strips To use the Ducker plug in Insert the plug in into an aux channel strip Assign all channel strip outputs that are supposed to duck dynamically lower the volume of the mix to a bus the aux channel strip chosen in step 1 Choose the bus that carries the ducking vocal signal in the Side Chain menu of the Ducker plug in Note Unlike all other side chain capable plug ins the Ducker side chain is mixed with the output signal after passing through the plug in This ensures that the ducking side chain signal the voice over is heard at the output Adjust the Ducker parameters Chapter 4 Dynamics Processors 95 96 Enveloper The Enveloper is an unusual processor that lets you shape the attack and release phases of a signal the signal s transients in other words This makes it a
119. er High Shelving Gain knob and field Sets the amount of gain applied to the output signal Input Gain field and slider Extended Parameters area Sets the amount of gain applied to the input signal Output Gain field and slider Extended Parameters area Sets the amount of gain applied to the output signal Distortion Effect The Distortion effect simulates the lo fi dirty distortion generated by a bipolar transistor You can use it to simulate playing a musical instrument through a highly overdriven amplifier or to create unique distorted sounds Drive slider and field Sets the amount of saturation applied to the signal Display Shows the impact of parameters on the signal Tone knob and field Sets the frequency for the high cut filter Filtering the harmonically rich distorted signal produces a softer tone Output slider and field Sets the output level This allows you to compensate for increases in loudness caused by adding distortion Chapter 3 Distortion Effects Distortion Il Distortion Il emulates the distortion circuit 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 e PreGain knob Sets the amount of gain applied to the input signal Drive knob Sets the amount of saturation applied to the signal Tone knob Sets the frequency of the highpass filter Filtering the harmonically rich distorted signal produces a
120. er adjustment graphic Drag the white dot to change the microphone position and distance relative to the cabinet Placement is limited to near field positioning Chapter 1 Amps and Pedals Mic pop up menu You can choose one of the Microphone models from the pop up menu Condenser Emulates the sound of a high end German studio condenser microphone The sound of condenser microphones is fine transparent and well balanced Dynamic Emulates the sound of popular American dynamic cardioid microphones This microphone type sounds brighter and more cutting than the Condenser model The mid range is boosted with lower mid frequencies being less pronounced making it a good choice for miking rock guitar tones It is especially useful if you want your guitar part to cut through other tracks in a mix Ribbon Emulates the sound of a ribbon microphone A ribbon microphone is a type of dynamic microphone that captures a sound often described as bright or brittle yet still warm It is useful for rock crunch and clean tones Tip Combining multiple microphone types can produce an interesting sound Duplicate the guitar track and insert Amp Designer on both tracks Select different microphones in each Amp Designer instance while retaining identical settings for all other parameters and set track signal levels to taste Setting Amp Designer s Output Level The Output slider or the Output field in the small interface is found at the lower
121. er cutoff frequency value is lower than that of the lowpass cutoff frequency only one line is shown This line represents the frequency band that passes through the filters in other words the filters act as a bandpass filter In this configuration the two filters operate serially meaning that the tap passes through one filter first then the other Chapter 2 Delay Effects 64 If the highpass filter s cutoff frequency value is above that of the lowpass filter cutoff frequency the filter switches from serial operation to parallel operation meaning that the tap passes through both filters simultaneously In this case the space between the two cutoff frequencies represents the frequency band being rejected in other words the filters act as a band rejection filter Editing Pan in Delay Designer s Tap Display The way the Pan parameter is represented in Pan view is entirely dependent on the input channel configuration mono to stereo stereo to stereo or surround Note Pan is not available in mono configurations In mono input stereo output configurations all taps are initially panned to the center To edit the pan position drag vertically from the center of the tap in the direction you wish to pan the tap or taps A white line extends outward from the center in the direction you have dragged reflecting the pan position of the tap or taps Chapter 2 Delay Effects Lines above the center position indicate pans to the left a
122. er to get a feel for how this works Folded The start and end levels of the clipped signal are unchanged but the center portion is effectively folded in half halved in the level above the threshold resulting in a softer distortion Cut The signal is abruptly distorted when the clipping threshold is exceeded Clipping that occurs in most digital systems is closest to Cut mode Chapter 3 Distortion Effects Displaced The start center and end levels of the signal above the threshold are offset resulting in a distortion which is less severe as signal levels cross the threshold The center portion of the clipped signal is also softer than in Cut mode Clip Level slider and field Sets the point below the clipping threshold of the channel strip at which the signal starts clipping Mix slider and field Extended Parameters area Sets the balance between dry original and wet effect signals Clip Distortion Clip Distortion is a nonlinear distortion effect that produces unpredictable spectra It can simulate warm overdriven tube sounds and can also generate drastic distortions Clip Distortion features an unusual combination of serially connected filters The incoming signal is amplified by the Drive value passes through a highpass filter and is then subjected to nonlinear distortion Following the distortion the signal passes through a lowpass filter The effect signal is then recombined with the original signal and this m
123. erb You can choose from several filter modes adjust resonance and also adjust the filter envelope dynamically over time See Working with Space Designer s Filter Chapter 12 Space Designer Convolution Reverb Global parameters After your IR is loaded these parameters determine how Space Designer operates on the overall signal and IR Included are input and output parameters delay and volume compensation predelay and so on See Working with Space Designer s Global Parameters Working with Space Designer s Impulse Response Parameters Space Designer can use either recorded impulse response files or its own synthesized impulse responses The circular area to the left of the main display contains the impulse response parameters These are used to determine the Impulse Response mode IR Sample mode or Synthesized IR mode load or create impulse responses and set the sample rate and length ample sample rate orig 4 IR Sample button and IR Sample menu Click the IR Sample button to switch to IR Sample mode In IRSample mode an impulse response sample is used to generate reverberation Click the down arrow next to the IR Sample button to open the IR Sample pop up menu in which you can load and manipulate impulse response samples and record and edit impulse responses with Impulse Response Utility See Working in Space Designer s IR Sample Mode sample rate slider and preserve length button The sample rate slider
124. erb parameters The interface is broken down into four parameter areas Early Reflections parameters Used to emulate the original signal s first reflections as they bounce off the walls ceiling and floor of a natural room See GoldVerb Early Reflections Parameters Reverb parameters Control the diffuse reverberations See GoldVerb Reverb Parameters Balance ER Reverb slider Controls the balance between the early reflections and reverb signal When the slider is set to either extreme position the other signal is not heard Mix slider and field Determines the balance between the effect wet and direct dry signals Chapter 11 Reverb Effects GoldVerb Early Reflections Parameters The GoldVerb offers the following Early Reflections parameters Predelay slider and field Determines the amount of time between the start of the original signal and the arrival of the early reflections Extremely short Predelay settings can color the sound and make it difficult to pinpoint the position of the signal source Overly long Predelay settings can be perceived as an unnatural echo and can divorce the original signal from its early reflections leaving an audible gap between them The optimum Predelay setting depends on the type of input signal or more precisely the envelope of the input signal Percussive signals generally require shorter predelays than signals where the attack fades in gradually A good working method is to use the longes
125. ers a maximum delay time of 10 seconds This means that if you load a setting into a project with a slower tempo than the tempo at which it was created some taps may fall outside the 10 second limit In such cases these taps will not be played but will be retained as part of the setting e Sync button Enables or disables synchronized mode Grid pop up menu Provides several grid resolutions which correspond to musical note durations The grid resolution along with the project tempo determines the length of each grid increment As you change grid resolutions the increments shown in the Identification bar change accordingly This also determines a step limitation for all taps As an example imagine a project with the current tempo set to 120 beats per minute The Grid pop up menu value is set to 1 16 notes At this tempo and grid resolution each grid increment is 125 milliseconds ms apart If Tap A is currently set to 380 ms turning on Sync mode would immediately shift Tap A to 375 ms If you subsequently moved Tap A forward in time it would snap to 500 ms 625 ms 750 ms and so on At a resolution of 1 8 notes the steps are 250 milliseconds apart so Tap A would automatically snap to the nearest division 500 ms and could be moved to 750 ms 1000 ms 1250 ms and so on Swing field Determines how close to the absolute grid position every second grid increment will be A Swing setting of 50 means that every grid increment has
126. ertical 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 G Formant As you alter the Pitch parameter you might notice that the formants don t change Formants are characteristic emphases of certain frequency ranges They are static and do not change with pitch Formants are responsible for the specific timbre of a given human voice The Pitch parameter is expressly used to change the pitch of a voice not its character If you set negative Pitch values for a female soprano voice you can turn it into an alto voice without changing the specific character of the singer s voice The Formant parameter shifts the formants while maintaining or independently altering the pitch If you set this parameter to positive values the singer sounds like Mickey Mouse By altering the parameter downward you can achieve vocals reminiscent of Darth Vader Tip If you set Pitch to 0 semitones Mix to 50 and Formant to 1 with Robotize turned off you can effectively place a singer with a smaller head next to the original singer Both will sing with the same voice in a choir of two This doubling of voices is quite effective with levels easily controlled by the Mix parameter Using Vocal Transformer s Robotize Mode When Robotize is enabled Vocal Transformer can augment or diminish the melody You can control the
127. ervals you can create special effects Natural articulations of the performance such as breath noises are preserved Any scale can be defined as a pitch reference technically speaking this is known as a pitch quantization grid with improperly intonated notes corrected in accordance with this scale Note Polyphonic recordings such as choirs and highly percussive signals with prominent noisy portions can t be corrected to a specific pitch Despite this feel free to try the plug in on drum signals 203 204 Pitch Correction Parameters The Pitch Correction effect offers the following parameters Use Global Tuning pitch correction oo Range Ref Pitch n worm ars Response Detune ict J Use Global Tuning button Enable to use the project s Tuning settings for the pitch correction process If disabled you can use the Ref Pitch field to freely set the desired reference tuning See Setting the Pitch Correction Reference Tuning Normal and Low buttons These determine the pitch range that is scanned for notes that need correction See Defining the Pitch Correction Effect s Quantization Grid Ref Pitch field Sets the desired reference tuning in cents relative to the root See Setting the Pitch Correction Reference Tuning Root pop up menu and field Click to choose the root note of the scale from the Root pop up menu See Defining the Pitch Correction Effect s Quantization Grid Scale pop up
128. es to the left of the center positions are synchronized with the host application tempo and include bar values triplet values and more Values to the right of the center positions are non synchronized and are displayed in Hertz cycles per second Note The ability to use synchronous bar values could be used to perform a formant shift every four bars on a cycled one bar percussion part for example Alternatively you could perform the same formant shift on every eighth note triplet within the same part Either method can generate interesting results and can lead to new ideas or add new life to old audio material EVOC 20 TrackOscillator Output Parameters The Output section provides control over the type stereo width and level of signal that is sent from the EVOC 20 TrackOscillator Level Stereo Mode a Stereo Width Signal menu Determines the signal that is sent to the EVOC 20 TrackOscillator main outputs You can choose among the following settings Voc oder Choose to hear the vocoder effect Syn thesis Choose to hear only the synthesizer signal Anaj lysis Choose to hear only the analysis signal Note The last two settings are mainly useful for monitoring purposes Level slider Controls the volume of the EVOC 20 TrackOscillator output signal Chapter 6 Filter Effects Stereo Mode pop up menu Sets the input output mode of the EVOC 20 Filterbank The choices are m s mono input to stereo outpu
129. est and highest amplitudes Dynamics processors enable you to adjust the dynamic range of individual audio files tracks or an overall project This can be to increase the perceived loudness and or to highlight the most important sounds while ensuring that softer sounds are not lost in the mix This chapter covers the following Types of Dynamics Processors p 86 Adaptive Limiter p 87 Compressor p 88 DeEsser p 92 Ducker p 94 Enveloper p 96 Expander p 98 Limiter p 99 Multipressor p 100 Noise Gate p 103 Silver Compressor p 106 Silver Gate p 107 Surround Compressor p 107 85 86 Types of Dynamics Processors There are four types of dynamics processors included in Logic Pro These are each used for different audio processing tasks Compressors Logic Pro features a number of downward compressors These behave like an automatic volume control lowering the volume whenever it rises above a certain level called the threshold So why would you want to reduce the dynamic level By reducing the highest parts of the signal called peaks a compressor raises the overall level of the signal increasing the perceived volume This gives the signal more focus by making the louder foreground parts stand out while keeping the softer background parts from becoming inaudible Compression also tends to make sounds tighter or punchier because transients are emphasized depending on attack and
130. eter Direction knob and field Determines the pan position for the middle the center of the stereo base of the recorded stereo signal See Using the Direction Mixer s Direction Parameter Using the Direction Mixer s Spread Parameter The Direction Mixer s Spread parameter behavior changes when fed LR or MS signals These differences are outlined below When working with LR signals the following applies to the Direction Mixer s Spread parameter At a neutral value of 1 the left side of the signal is positioned precisely to the left and the right side precisely to the right As you decrease the Spread value the two sides move toward the center of the stereo image A value of 0 produces a summed mono signal both sides of the input signal are routed to the two outputs at the same level At values greater than 1 the stereo base is extended out to an imaginary point beyond the spatial limits of the speakers The following applies when working with MS signals e Values of 1 or higher increase the level of the side signal making it louder than the middle signal At a value of 2 you hear only the side signal Chapter 7 Imaging Processors 163 164 Using the Direction Mixer s Direction Parameter When Direction is set to a value of 0 the midpoint of the stereo base in a stereo recording is perfectly centered within the mix The following applies when working with LR signals At 90 the center of the stereo base
131. etric EQ Gain 0 0dB 4 Frequency 980Hz a 0 74 amp Gain slider and field Sets the amount of cut or boost Frequency slider and field Sets the cutoff frequency Q Factor slider and field Adjusts the Q bandwidth Silver EQ The Silver EQ includes three bands a high shelving EQ a parametric EQ and a low shelving EQ You can adjust the cutoff frequencies for the high shelving and low shelving EQs You can adjust the center frequency gain and Q for the parametric EQ x Inst 1 p View v ShowCS v Showinsert Bypass lt gt default Silver EQ High Shelf 00de High Frequency 9500Hz Frequency 920Hz 0 7 0 0dB 0 0dB Low Frequency 53Hz High Shelf slider and field Sets the level of the high shelving EQ High Frequency slider and field Sets the cutoff frequency for the high shelving EQ Frequency slider and field Sets the center frequency of the parametric EQ Q Factor slider and field Adjusts the range bandwidth of the parametric EQ Chapter5 Equalizers 129 130 Gain slider and field Sets the amount of cut or boost for the parametric EQ e Low Shelf slider and field Sets the level of the low shelving EQ Low Frequency slider and field Sets the cutoff frequency for the low shelving EQ Chapter5 Equalizers Filter Effects 6 Filters are used to emphasize or suppress frequencies in an audio signal resulting in a change to the tonal color of the audio Logic Pro
132. ets the ratio between the source and distorted signals The Bright Fat switch changes between two fixed high shelving filter frequencies Blue and red LEDs indicate each switch position respectively Chapter 1 Amps and Pedals 43 Stompbox Description Fuzz Machine An American fuzz distortion effect Fuzz controls the input gain Overall output gain is set with Level The Tone knob increases treble while simultaneously rolling off low frequencies as you move it to higher values Grinder Grinder is a lo fi dirty metal distortion Grind sets the amount of drive applied to the input signal Tone is controlled with the Filter knob making the sound harsher and more crunchy at higher values The Full Scoop switch alternates between two fixed Gain Q filter settings At the Full position filtering is less pronounced than at the Scoop position Overall output level is controlled with the Level knob Happy Face Fuzz A softer full sounding distortion effect Fuzz sets the amount of saturation applied to the input signal Volume sets the output level Hi Drive An overdrive effect that can emphasize high frequency content in the signal Level controls the effect output The Treble Full switch sets a fixed shelving frequency allowing either the treble portion or the full range input signal to be processed Monster Fuzz A saturated somewhat harsh distortion Roar sets the amount of gain applied to the input signal Growl se
133. ettings menu which you use to load a setting for the entire Pedalboard plug in this feature can be used to load a setting for a specific stompbox type Import Mode To activate or deactivate import mode Click the Import Mode button to show all pedals used in the most recent Pedalboard setting When the Import Mode button is active the Pedal Browser switches to an alternate view mode that displays imported settings When import mode is inactive the normal Pedal Browser view is shown To import pedal settings into the Pedal Browser Click the Import Mode button to activate import mode Note that the View menu changes to the Select Setting button Note If this is your first attempt to import settings a dialog opens where you can select a setting to import Chapter 1 Amps and Pedals Click the Select Setting button and select a setting then click Open Dependent on the chosen setting one or more stompboxes appear in the Pedal Browser The name of the imported setting is shown at the bottom of the Pedal Browser To add an imported pedal to the Pedal area Do one of the following Drag the stompbox that you want to add from the Pedal Browser to the appropriate Pedal area position This can be to the left to the right or in between existing pedals Ensure that no pedal is selected in the Pedal area then double click a stompbox in the Pedal Browser to add it to the right of all existing effects in the Pedal area Note The parame
134. etween the generated and original signals with the Wet and Dry sliders Use the High and Low parameters to define the two frequency bands which SubBass uses to generate tones High Center and Low Center define the center frequency of each band and High Bandwidth and Low Bandwidth define the width of each frequency band The High Ratio and Low Ratio knobs define the transposition amount for the generated signal in each band This is expressed as a ratio of the original signal For example Ratio 2 transposes the signal down one octave Important Within each frequency band the filtered signal should have a reasonably stable pitch in order to be analyzed correctly In general narrow bandwidths produce the best results because they avoid unwanted intermodulations Set High Center a fifth higher than Low Center which means a factor of 1 5 for the center frequency Derive the sub bass to be synthesized from the existing bass portion of the signal and transpose by one octave in both bands Ratio 2 Do not overdrive the process or you will introduce distortion If you hear frequency gaps move one or both Center frequency knobs or widen the Bandwidth of one or both frequency ranges a little Tip Be prudent when using SubBass and compare the extreme low frequency content of your mixes with other productions It is very easy to go overboard with it Chapter 13 Specialized Effects and Utilities 257 Utilities and Tools 14 The t
135. evel meter Shows the overall output level Using the Multipressor In the graphic display the blue bars show the gain change not merely the gain reduction as with a standard compressor The gain change display is a composite value consisting of the compression reduction plus the expander reduction plus the auto gain compensation plus the gain make up Chapter 4 Dynamics Processors Setting Multipressor Compression Parameters The Compression Threshold and Compression Ratio parameters are the key parameters for controlling compression Usually the most useful combinations of these two settings are a low Compression Threshold with a low Compression Ratio or a high Compression Threshold with a high Compression Ratio Setting Multipressor Downward Expansion Parameters The Expansion Threshold Expansion Ratio and Expansion Reduction parameters are the key parameters for controlling downward expansion They determine the strength of the expansion applied to the chosen range Setting Multipressor Peak RMS and Envelope Parameters Adjusting the parameter between Peak 0 ms minimum value and RMS root mean square 200 ms maximum value is dependent on the type of signal you want to compress An extremely short Peak detection setting is suitable for compression of short and high peaks of low power which do not typically occur in music The RMS detection method measures the power of the audio material over time and thus works much more m
136. evels will result in audible reflections patterns and discreet echoes Ramp Time field Adjusts the length of time elapsed between the Initial and End Density levels End Level field Sets the density of the reverb tail If you select an End Level value that is too low the reverb tail will sound grainy You may also find that the stereo spectrum is affected by lower values Reflection Shape slider Determines the steepness shape of the early reflection clusters as they bounce off the walls ceiling and furnishings of the virtual space Small values result in clusters with a sharp contour and large values result in an exponential slope and a smoother sound This is handy when recreating rooms constructed of different materials Reflection Shape in conjunction with suitable settings for the envelopes density and early reflection will assist you in creating rooms of almost any shape and material Chapter 12 Space Designer Convolution Reverb 237 Working with Space Designer s EQ Space Designer features a four band EQ comprised of two parametric mid bands plus two shelving filters one low shelving filter and one high shelving filter You can edit the EQ parameters numerically in the parameter bar or graphically in the main display Input Output IR Sample sample rate orig EQ On Off button Individual EQ band buttons EQ On Off button Enables or disables the entire EQ section Individual EQ band buttons
137. everb effect Additional Resources Along with the documentation that comes with Logic Pro there are a variety of other resources you can use to find out more Release Notes and New Features Each application offers detailed documentation that covers new or changed features and functions This documentation can be accessed in the following location Click the Release Notes and New Features links in the application Help menu Logic Pro Website For general information and updates as well as the latest news on Logic Pro go to httg pple com logicpro Apple Service and Support Websites For software updates and answers to the most frequently asked questions for all Apple products go to the general Apple Support webpage You ll also have access to product specifications reference documentation and Apple and third party product technical articles 1pple com support Preface An Introduction to the Logic Pro Effects For software updates documentation discussion forums and answers to the most frequently asked questions for Logic Pro go to e http www apple com support logicpro For discussion forums for all Apple products from around the world where you can search for an answer post your question or answer other users questions go to http discussions apple com Preface An Introduction to the Logic Pro Effects Amps and Pedals 1 Logic Pro features an extensive collection of guitar and bass amplifiers
138. fect Order Buttons Chapter 6 Filter Effects 153 154 Wah parameters Provide control over the type and tone of the wah wah effect See Wah Parameters Auto Wah parameters Set the depth and envelope times for the automatic wah wah effect See Auto Wah Parameters Fuzz parameters Set the compression ratio and control the tone and level of the integrated distortion circuit See Fuzz Parameters Effect Order Buttons These buttons determine the signal flow of the Fuzz Wah effect Click Wah Fuzz or Fuzz Wah to choose the desired flow Note that the Fuzz Wah plug in features an integrated compression circuit The compressor always precedes the fuzz effect When Wah Fuzz is selected the compressor is positioned between the wah wah and the fuzz effect When Fuzz Wah is selected however the compressor is placed first in the signal chain Wah Parameters This group of parameters controls the tone and behavior of the wah wah effect a Wah Mode pop up menu Includes the following Wah Wah effect settings Off Wah Wah effect is disabled ResoLP Resonating Lowpass Filter In this mode the Wah Wah works as a resonance capable lowpass filter At the minimum pedal position only low frequencies can pass ResoHP Resonating Highpass Filter In this mode the Wah Wah works as a resonance capable highpass filter At the maximum pedal position only high frequencies can pass Peak In this mode the Wah Wah works as a peak
139. for interesting tempo pitch and retro digital sounding effects If you are running Space Designer in a project that uses a higher sample rate than the impulse response you may also want to reduce the impulse response sample rate Make sure the preserve length function is enabled This cuts CPU power consumption without compromising reverb quality There is no loss in reverb quality because the impulse response does not benefit from the higher project sample rate You can make similar adjustments while running in Synthesized IR mode Most typical reverb sounds don t feature an excessive amount of high frequency content If you were running at 96 kHz for example you would need to make use of some deep lowpass filtering to obtain the mellow frequency response characteristics of many reverb sounds A better approach would be to first reduce the high frequencies by 1 2 or even 1 4 using the sample rate slider and then apply the lowpass filter This conserves a considerable amount of CPU power Setting Impulse Response Lengths in Space Designer You can use the Length parameter to set the length of the impulse response sampled or synthesized Chapter 12 Space Designer Convolution Reverb All envelopes are automatically calculated as a percentage of the overall length which means that if this parameter is altered your envelope curves will stretch or shrink to fit saving you time and effort When you are using an impulse respons
140. for peaks Remove DC checkbox Extended Parameters area Enable to activate a highpass filter that removes direct current DC from the signal DC can be introduced by lower quality audio hardware 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 mix into focus while preventing softer parts from becoming inaudible It is probably the most versatile and widely used sound shaping tool in mixing next to EQ You can use the Compressor with individual tracks including vocal instrumental and effects tracks as well as on the overall mix Usually you insert the Compressor directly into a channel strip Chapter 4 Dynamics Processors Compressor Parameters The Compressor offers the following parameters Circuit Type pop up menu Choose the type of circuit emulated by the Compressor The choices are Platinum Class ic A_R Class ic A_U VCA FET and Opto optical Side Chain Detection pop up menu Determines if the Compressor uses the maximum level of each side chained signal Max or the summed level of all side chained signals Sum to exceed or fall below the threshold e If either of the stereo channels exceeds or falls below t
141. from a higher to a lower value Chapter9 Modulation Effects 195 196 Modulating the Ringshifter with the LFO The oscillator Frequency and Dry Wet parameters can be modulated with the LFO and the envelope follower see Modulating the Ringshifter with the Envelope Follower The oscillator frequency even allows modulation through the 0 Hz point thus changing the oscillation direction The LFO produces continuous cycled control signals Power button Turns the LFO on or off and enables the following parameters Symmetry and Smooth sliders and fields These controls on either side of the Waveform display change the shape of the LFO waveform Waveform display The LFO waveform display provides visual feedback about the waveform shape Rate knob and field Sets the waveform cycle speed of the LFO Sync button Synchronizes the LFO cycles LFO rate with the project tempo using musical note values Controlling the Ringshifter Output Parameters The output parameters are used to set the balance between the effect and input signals and also to set the width and feedback of the Ringshifter Dry Wet knob and field Sets the mix ratio of the dry input signal and the wet effect signal Chapter9 Modulation Effects Feedback knob and field Sets the amount of the signal that is routed back to the effect input Feedback adds an edge to the Ringshifter sound and is useful for a variety of special effects It produces a rich phasi
142. from the Pedal area Chapter 1 Amps and Pedals 41 42 Remove all stompboxes from the Pedal area This automatically removes an existing Mixer utility To remove an effect from the second bus Click the name of the pedal or on either of the gray lines in the Routing area Note The removal of all effects from Bus B does not remove the second bus The Mixer utility pedal remains in the Pedal area even when a single stompbox effect is in the Pedal area This allows parallel routing of wet and dry signals Only when all pedal effects are removed from the Pedal area is the Mixer utility and second bus removed To determine the split point between busses When more than one bus is active a number of dots appear along the cables gray lines in the Routing area These represent the output the socket of the pedal to the lower left of the dot Click the appropriate dot to determine where the split point where the signal is routed between busses A cable appears between the busses when you click a dot Note You can not create a split point directly before or after the Mixer utility To switch between a Splitter utility and bus split point Double click a bus split point dot in the Routing area to replace it with a Splitter utility The Splitter utility is shown in the Pedal area Double click the Splitter label in the Routing area to replace the Splitter utility with a bus split point dot The Splitter utility is removed from th
143. gnal Intensity slider and field Determines the modulation amount Speed knob and field Defines the frequency of the built in LFO and therefore the speed of the modulation Channel Delay slider and field Determines the delay time in samples e Mix slider and field Sets the balance between the effect and input signals Chapter9 Modulation Effects 201 202 Tremolo Effect The Tremolo effect modulates the amplitude of the incoming 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 Depth slider and field Determines the modulation amount Waveform display Shows the resulting waveform Rate knob and field Sets the frequency of the LFO e Symmetry and Smoothing knobs and fields Use these to alter the shape of the LFO waveform If Symmetry is set to 50 and Smoothing to 0 the LFO waveform has a rectangular shape This means that the timing of the highest and lowest volume signals is equal with the switch between both states occurring abruptly e Phase knob and field Available only in stereo and surround instances Controls the phase relationship between the individual channel modulations At 0 modulation values are reached simultaneously for all channels Values of 180 or 180 indicate the greatest possible distance between the modulati
144. gs American Bright Based on the American Basic amp this model emphasizes the upper mid frequencies from 4 5 kHz upward New American Basic 1980s era American bass amp well suited for blues and rock recordings New American Bright Based on the New American Basic amp this model strongly emphasizes the frequency range above 2 kHz Well suited for rock and heavy metal Chapter 1 Amps and Pedals Top Class DI Warm Famous DI box simulation well suited for reggae and pop recordings Mid frequencies in the range between 500 and 5000 Hz are de emphasized Top Class DI Deep Based on the Top Class DI Warm this model is well suited for funk and fusion The mid frequency range is strongest around 700 Hz Top Class DI Mid Based on the Top Class DI Warm this model features an almost linear frequency range with no frequencies emphasized It is suitable for blues rock and jazz recordings Pre Gain slider Sets the pre amplification level of the input signal Bass Mid and Treble sliders Adjusts the bass mid and treble levels Mid Freq slider Sets the center frequency of the mid band between 200 Hz and 3000 Hz Output Level slider Sets the final output level for Bass Amp Guitar Amp Pro Guitar Amp Pro can simulate the sound of popular guitar amplifiers and the speakers used with them You can process guitar signals directly which enables you to reproduce the sound of your guitar through a number of high qual
145. h is activated or by activating Match after a new spectrum has been loaded any existing changes to the filter curve are discarded and Apply is set to 100 By default the Apply slider is set to 100 when you learn the frequency curve of an audio signal In many cases you may want to lower it slightly to avoid extreme spectral changes to your mix It is also recommended that you use the Smoothing slider to adjust the spectral detail of the generated EQ curve Using the Match EQ Shortcut Menu Control click or right click either Learn button to open a shortcut menu This offers commands that can be applied to the spectrum of the template or the current material e Clear Current Material Spectrum Clears the current spectrum Chapter5 Equalizers 125 126 Copy Current Spectrum Copies the current spectrum to the Clipboard this can be used by any Match EQ instance in the current project Paste Current Spectrum Pastes the Clipboard contents to the current Match EQ instance Load Current Material Spectrum from setting file Loads the spectrum from a stored setting file Generate Current Material Spectrum from audio file Generates a frequency spectrum for an audio file that you have chosen Editing the Match EQ Filter Curve You can graphically edit the filter curve in the graphic display by adjusting the various points shown in each band As you drag the current values appear in a small box inside the graphic display allowing yo
146. h to its maximum 20 000 Hz value so that the entire frequency range is processed Turn up the Super Energy and Sub Energy knobs one at a time then try different Threshold settings This should give you a good sense of how different Threshold levels affect the sound of Super Energy and Sub Energy When you come across a sound that you like or consider useful narrow the Bandwidth drastically gradually increase the Center Freq and then use the Low Level and High Level sliders to mix in some treble and bass from the original signal At lower Speed settings turn up the CF Mod or BW Mod knobs Follow these steps to acquaint yourself with the Spectral Gate Set the frequency band you want to process by using the Center Freq and Bandwidth parameters The graphic display visually indicates the band defined by these two parameters After the frequency band is defined use the Threshold parameter to set the appropriate level All incoming signals above and below the threshold level are divided into upper and lower frequency ranges Use the Super Energy knob to control the level of the frequencies above the Threshold and use the Sub Energy knob to control the level of the frequencies below the Threshold You can mix the frequencies that fall outside the frequency band defined by the Center Freq and Bandwidth parameters with the processed signal a Use the Low Level slider to blend the frequencies below the defined frequency band with the
147. he Threshold both channels are compressed If Sum is chosen the combined level of both channels must exceed the Threshold before compression occurs Gain Reduction meter Shows the amount of compression in real time e Attack knob and field Determines the amount of time it takes for the compressor to react when the signal exceeds the threshold Compression curve display Shows the compression curve created by the combination of Ratio and Knee parameter values Input level is shown on the x axis and output level on the y axis e Release knob and field Determines the amount of time it takes for the compressor to stop reducing the signal after the signal level falls below the threshold Auto button When the Auto button is active the release time dynamically adjusts to the audio material Ratio slider and field Sets the compression ratio the ratio of signal reduction when the threshold is exceeded Knee slider and field Determines the strength of compression at levels close to the threshold Lower values result in more severe immediate compression hard knee Higher values result in gentler compression soft knee Chapter 4 Dynamics Processors 89 90 Compressor Threshold slider and field Sets the threshold level signals above this threshold value are reduced in level Peak RMS buttons Determines whether signal analysis is with the Peak or RMS method when using the Platinum circuit type Gain slider an
148. he graphic display Each band has a pivot point a small circle on the curve at the location of the band s frequency you can adjust the Q or width of the band by dragging the pivot point vertically You can also adjust the decibel scale of the graphic display by vertically dragging either the left or right edge of the display where the dB scale is shown when the Analyzer is not active When the Analyzer is active dragging the left edge adjusts the linear dB scale and dragging the right edge adjusts the Analyzer dB scale To increase the resolution of the EQ curve display in the most interesting area around the zero line drag the dB scale on the left side of the graphic display upward Drag downward to decrease the resolution Using the Linear Phase EQ Analyzer The Analyzer when active makes uses of a mathematical process called a Fast Fourier Transform FFT to provide a real time curve of all frequency components in the incoming signal This is superimposed over any EQ curves you have set The Analyzer curve uses the same scale as the EQ curves making it easy to recognize important frequencies in the incoming audio This also simplifies the task of setting EQ curves to raise or lower the levels of frequencies or frequency ranges The bands derived from FFT analysis are divided in accordance with the frequency linear principle there are more bands in higher octaves than in lower ones Chapter5 Equalizers As soon as the Analyzer
149. her fidelity than the Vintage EQ with tighter lows and crisper highs This EQ is useful if you want to brighten your tone and reduce distortion Chapter 1 Amps and Pedals EQ type Description Modern Based on a digital EQ unit popular in the 1980s and 1990s This EQ is useful for sculpting the hyped highs booming lows and scooped mids associated with the era s rock and metal music styles Boutique Replicates the tone section of a retro modern boutique amp It excels at precise EQ adjustments though its tone may be cleaner than desired when used with vintage amplifiers This EQ is a good choice if you want a cleaner brighter sound Using Amp Designer s Gain Presence and Master Controls The amp parameters include controls for the input gain presence and master output The Gain knob is found to the left in the knobs section and the Presence and Master knobs are to the right Gain Presence Master e Gain knob Sets the amount of pre amplification applied to the input signal This control affects various amp models differently For example when you are using the British Amp the maximum gain setting produces a powerful crunch sound When you are using the Vintage British Head or Modern British Head the same gain setting produces heavy distortion suitable for lead solos Presence knob Adjusts the high frequency range above the range of the Treble control The Presence parameter affects only the output Master
150. hold for all metering tools in the MultiMeter as follows Analyzer A small yellow segment above each 1 3 octave level bar indicates the most recent peak level Goniometer All illuminated pixels are held during a peak hold Correlation Meter The horizontal area around the white correlation indicator denotes phase correlation deviations in real time in both directions A vertical red line to the left of the correlation indicator shows the maximum negative phase deviation value You can reset this line by clicking on it during playback Level Meter A small yellow segment above each stereo level bar indicates the most recent peak level Hold Time pop up menu When peak hold is active sets the hold time for all metering tools to 2 4 or 6 seconds or infinite e Reset button Click to reset the peak hold segments of all metering tools Surround MultiMeter The surround version of the MultiMeter is specifically designed for analysis and metering of multichannel surround files You can view either the Analyzer Goniometer or Correlation Meter results in the main display area Use the controls on the left side of the interface to switch the view and set other MultiMeter parameters The Peak RMS Level Meter is visible on the right Chapter 8 Metering Tools Although you can insert the Surround MultiMeter directly into any channel strip it is more commonly used in the master channel strip of the host application when you ar
151. hold time for all metering tools to 2 4 or 6 seconds or infinite Reset button Click to reset the peak hold segments of all metering tools Tuner You can tune instruments connected to your system with the Tuner utility This ensures that your external instrument recordings will be in tune with any software instruments samples or existing recordings in your projects Precision button Graphic tuning display Keynote Octave displays Tuning adjustment slider and field e Precision button The Graphic tuning display defaults to a linear scale Enable the Precision button to change the scaling so that it stretches outwards from the center Graphic tuning display Indicates the pitch of the note in the semicircular area around the Keynote Octave displays At the centered 12 o clock position the note is correctly tuned If the indicator moves to the left of center the note is flat If the indicator moves to the right of center the note is sharp The numbers around the edge of the display show the variance in cents from the target pitch The range is marked in single semitone steps for the first 6 semitones sharp or flat Thereafter larger increments are shown Chapter 8 Metering Tools Keynote Octave displays The upper Keynote display shows the target pitch of the note being played the closest tuned pitch The lower Octave display indicates the octave that the incoming note falls into This matches the MIDI octave
152. ht Link button is not enabled LFO Left Right Link button Available only in stereo and surround instances it links the modulation rates of the left and right stereo channels Adjustment of either Rate knob will affect the other channels Chapter9 Modulation Effects LFO Phase knob and field Available only in stereo and surround instances it controls the phase relationship between 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 Note The LFO Phase parameter is available only if the LFO Left Right Link button is active Distribution pop up menu Available only in surround instances it defines how the phase offsets between the individual channels are distributed in the surround field You can choose from circular leftesright fronteorear random and new random distributions Note When you load a setting that uses the random option the saved phase offset value is recalled If you want to randomize the phase setting again choose new random from the Distribution pop up menu Volume Mod ulation slider and field Determines the impact that LFO modulation has on the amplitude of the effect signal Output Mix slider and field Determines the balance between dry and wet signals All Pass button Ex
153. icking the corresponding On button to the left The On button is red when active Note The Effects section is placed before the Presence and Master controls in the signal flow and receives the preamplified pre Master signal Tremolo Vibrato and Reverb are described in the following sections e Using Guitar Amp Pro s Tremolo and Vibrato Effects e Using Guitar Amp Pro s Reverb Effect Chapter 1 Amps and Pedals 33 Using Guitar Amp Pro s Tremolo and Vibrato Effects Tremolo and vibrato are controlled by an On button the FX pop up menu the Depth and Speed knobs and the Sync button in the Effects section Tremolo modulates the amplitude or volume of the sound and vibrato modulates the pitch FX pop up menu You can choose either Tremolo or Vibrato Depth knob Sets the intensity of the modulation e Speed knob Sets the speed of the modulation in Hertz Lower settings produce a smooth and floating sound while higher settings produce a rotor like effect Sync button When the Sync button is turned on the modulation speed is synchronized to the project tempo You can adjust the Speed knob to select bar beat and musical note values including triplet and dotted notes When the Sync button is turned off the modulation speed can be set to any available value with the Speed knob Using Guitar Amp Pro s Reverb Effect Reverb is controlled by an On button the Reverb pop up menu and a Level knob in the Reverb section near
154. ield Determines the reverb time of the high band Most natural rooms have a reverb time somewhere in the range of 1 to 3 seconds This time is reduced by absorbent surfaces such as carpet and curtains and soft or dense furnishings such as sofas armchairs cupboards and tables Large empty halls or churches have reverb times of up to 8 seconds with some cavernous or cathedral like venues extending beyond that PlatinumVerb Output Parameters The PlatinumVerb offers the following Output parameters Dry slider and field Controls the amount of the original signal Wet slider and field Controls the amount of the effect signal 224 Chapter 11 Reverb Effects SilverVerb The SilverVerb is similar to the AVerb but it provides an additional LFO that can modulate the reverberated signal It also includes a high cut and a low cut filter allowing you to filter frequencies from the reverb signal High frequencies usually sound somewhat unpleasant hamper speech intelligibility or mask the overtones of the original signals Long reverb tails with a lot of bottom end generally result in an indistinct mix Predelay slider and field Determines the time between the original signal and the reverb signal Reflectivity slider and field Defines how reflective the imaginary walls ceiling and floor are Room Size slider and field Defines the dimensions of a simulated room Density Time slider and field Determines both the density and
155. ighest bands are ignored In the Lowpass setting all frequencies above the highest bands are filtered Resonance knob Determines the basic sonic character of both filter banks Increasing Resonance emphasizes the center frequency of each band Low settings result in a softer character high settings result in a sharper brighter character Boost A and B knobs Set the amount of boost or cut applied to the frequency bands in Filter Bank A or B This allows you to compensate for the reduction in volume caused by lowering the level of one or more bands If you use Boost to set the level mix relationship between the filter banks you can use Fade A B see Fade AB slider below to alter the tonal color but not the levels Slope pop up menu Sets the amount of filter attenuation applied to all filters in both filter banks Choices are 1 6 dB Oct and 2 12 dB Oct 1 sounds softer 2 sounds tighter Fade AB slider Crossfades between Filter Bank A and Filter Bank B At the top position only Bank A is audible At the bottom position only Bank B is audible In the middle position the signals passing through both banks are evenly mixed EVOC 20 Filterbank Modulation Parameters The Modulation section offers two LFOs The LFO Shift parameters on the left side control the Formant Shift parameter The LFO Fade parameters on the right side control the Fade AB parameter E OE sae Intensity 5 Intensity LFO Shift Intensi
156. ight channels EVOC 20 TrackOscillator The EVOC 20 TrackOscillator is a vocoder with a monophonic pitch tracking oscillator The tracking oscillator tracks or follows the pitch of a monophonic input signal If the input signal is a sung vocal melody the individual note pitches are tracked and mirrored or played by the synthesis engine The EVOC 20 TrackOscillator features two formant filter banks an analysis bank and a synthesis filter bank Each offers multiple input options You can capture an analysis signal source by using the audio arriving at the input of the channel strip that the EVOC 20 TrackOscillator is inserted in or by using a side chained signal from another channel strip The synthesis source can be derived from the audio input of the channel strip that the EVOC 20 TrackOscillator is inserted in a side chain signal or the tracking oscillator As you can freely select both the analysis and synthesis input signals the EVOC 20 TrackOscillator is not limited to pitch tracking effects It is extremely useful for unusual filter effects For example you could filter an orchestral recording on one channel strip with train noises side chained from another channel strip Another great use is for processing drum loops with side chained signals such as other drum loops or rhythmic guitar clavinet and piano parts What Is a Vocoder The word vocoder is an abbreviation for VOice enCODER A vocoder analyzes the sonic characte
157. ignal with the original signal e Semi Tones slider and field Sets the pitch shift value in semitones Cents slider and field Controls detuning of the pitch shift value in cents 1 100th of a semitone Drums Speech and Vocals buttons Select one of three optimized algorithms for common types of audio material Drums Maintains the groove rhythmic feel of the source signal Speech Provides a balance between 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 Vocals Retains the intonation of the source making it well suited for signals that are inherently harmonic or melodious such as string pads e Mix slider and field Sets the balance between the effect and original signals Chapter 10 Pitch Effects 207 208 Timing pop up menu Extended Parameters area Determines how timing is derived by following the selected algorithm Preset by analyzing the incoming signal Auto or by using the settings of the Delay Crossfade and Stereo Link parameters described below Manual Note The following three parameters are active only when Manual is chosen in the Timing pop up menu Delay slider and field Extended Parameters area Sets the amount of delay applied to the input signal The lower the frequencies of the input signal the higher longer a delay time you sho
158. ilance from the signal The advantage of using the DeEsser rather than an EQ 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 The DeEsser has extremely fast attack and release times When using the DeEsser you can set the frequency range being compressed the Suppressor frequency independently of the frequency range being analyzed the Detector frequency The two ranges can be easily compared in the DeEsser s Detector and Suppressor frequency range displays The Suppressor frequency range is reduced in level for as long as the Detector frequency threshold is exceeded The DeEsser does not use a frequency dividing network a crossover utilizing lowpass and highpass filters Rather it isolates and subtracts the frequency band resulting in no alteration of the phase curve Chapter 4 Dynamics Processors The Detector parameters are on the left side of the DeEsser window and the Suppressor parameters are on the right The center section includes the Detector and Suppressor displays and the Smoothing slider DeEsser Detector Section Detector Frequency knob and field Sets the frequency range for analysis Detector Sensitivity knob and field Sets the degree of responsiveness to the input signal Monitor pop up menu Choose Det ector to monitor the isolated Detector signal Sup pressor to monitor th
159. image Clicking these buttons reverses the tap position from left to right or vice versa For example if a tap is set to 55 left clicking the flip button will swap it to 55 right Chapter 2 Delay Effects Pan field Controls the pan position for mono input signals stereo balance for stereo input signals and surround angle when used in surround configurations e Pan displays a percentage between 100 full left and 100 full right which represents the pan position or balance of the tap A value of 0 represents the center panorama position When used in surround a surround panner replaces the percentage representation For more information see igner in Surround rking with Delay De Spread field When a stereo to stereo or stereo to surround instance of Delay Designer is used Spread sets the width of the stereo spread for the selected tap Mute button Mutes or unmutes the selected tap Level field Determines the output level for the selected tap Editing Delay Designer Taps with the Shortcut Menu Control click or right click a tap in Delay Designer s Tap display to open a shortcut menu containing the following commands Copy sound parameters Copies all parameters except the delay time of the selected tap or taps into the Clipboard Paste sound parameters Pastes the tap parameters from the Clipboard into the selected tap or taps If there are more taps in the Clipboard than are selected i
160. imple compressor Sustain sets the threshold level Signals above this are reduced in level Level determines the output gain The Attack switch can be set to Fast for signals with fast attack transients such as drums or to Slow for signals with slow attack phases such as strings Utility Pedals This section describes the parameters of the Mixer and Splitter pedals Stompbox Description Mixer A utility that is used to control the level relationship between Bus A and Bus B signals It can be inserted anywhere in the signal chain but is typically used at the end of the chain at the extreme right of the Pedal area See Using Pedalboard s Routing Area for details on use The A Mix B switch solos the A signal mixes the A and B signals or solos the B signal The level setting of the Mix fader is relevant for all A Mix B switch positions In stereo instances the Mixer utility also provides discrete Pan controls for each bus Chapter 1 Amps and Pedals Stompbox Description Splitter A utility that can be inserted anywhere in the signal chain Splitter can be used in two ways When set to Freq it works as a frequency dependent signal splitter that divides the incoming signal Signals above the frequency set with the Frequency knob are sent to Bus B Signals below this frequency are sent to Bus A When set to Split the incoming signal is routed equally to both buses The Frequency knob has no impact in this
161. in a waveform with rich harmonic content Chapter 6 Filter Effects 147 148 Important The parameters discussed in this section are available only if the Synthesis In menu is set to Osc FM Int FM Ratio FM Ratio field 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 whereas 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 3 000 produces results resembling a square wave with a pulse width of 33 FM Int knob Determines the intensity of modulation Higher values result in a more complex waveform with more overtones Ata value of 0 the FM tone generator is disabled and a sawtooth wave is generated At values above 0 the FM tone generator is activated Higher values result in a more complex and brighter sound Coarse Tune value field Sets the pitch offset of the oscillator in semitones e Fine Tune value field Sets the pitch offset in cents Tracking Oscillator Pitch Correction Parameters The tracking oscillator pitch parameters control the automatic pitch correction feature of the tracking oscillator They can be used to constrain the pitch of the tracking oscillator to a scale or
162. ing Amp Designer Microphone Parameters Amp Designer offers a choice between three different virtual microphones As with every other component in the tone chain different selections yield very different results After choosing a cabinet you can set the type of microphone you want to be emulated and where the microphone is placed in relation to the cabinet The Mic pop up menu is available near the right end of the black bar at the bottom and the speaker adjustment graphic appears when you move your mouse to the area above the Mic pop up menu Note The parameters described in this section are accessible only in the full Amp Designer interface If you are in the small interface click the disclosure triangle to the right of the Output field at the bottom right edge of the interface to switch back to the full interface Move your mouse above the Mic pop up menu to display the speaker adjustment graphic Cabinet and speaker adjustment graphic By default the microphone is placed in the center of the speaker cone on axis This placement produces a fuller more powerful sound suitable for blues or jazz guitar tones If you place the microphone on the rim of the speaker off axis you obtain a brighter thinner tone making it suitable for cutting rock or R amp B guitar parts Moving the microphone closer to the speaker emphasizes bass response The microphone position is shown on the cabinet and indicated by the white dot in the speak
163. intensity of this distortion with the Tracking parameter Robotize Tracking Chapter 10 Pitch Effects The Tracking slider and field feature is enhanced by four buttons which immediately set the slider to the most useful values as follows e sets the slider to 100 All intervals are mirrored 0 sets the slider to 0 Delivers interesting results with every syllable of the vocal track being sung at the same pitch Low values turn sung lines into spoken language 1 sets the slider to 100 The range of the melody is maintained Higher values augment and lower values diminish the melody 2 sets the slider to 200 The intervals are doubled The Pitch Base parameter is used to transpose the note that the Tracking parameter is following As an example With Tracking set to 0 the pitch of the spoken note will be transposed to the chosen base pitch value Chapter 10 Pitch Effects 211 Reverb Effects You can use Reverb effects to simulate the sound of acoustic environments such as rooms concert halls caverns or an open space Sound waves repeatedly bounce off the surfaces walls ceilings windows and so on of any space or off objects within a space gradually dying out until they are inaudible These bouncing sound waves result in a reflection pattern more commonly known as a reverberation or reverb The starting portion of a reverberation signal consists of a number of discrete reflec
164. interface is divided into three main sections the Formant Filter parameters section in the center of the window the Modulation parameters section at the bottom center and the Output parameters section along the right side Formant Filter parameters evoc 20 FB VQ Formant Shift bands lowest slope Output parameters 6 Resonance LO sate A Intensity intensity Medulation parameters Formant Filter parameters Control the frequency bands in the two filter banks Filter Bank A top blue and Filter Bank B bottom green See EVOC 20 Filterbank Formant Filter Parameters Modulation parameters Control how Formant Filter parameters are modulated See EVOC 20 Filterbank Modulation Parameters Output parameters Control the overall output level and panning of the EVOC 20 Filterbank See EVOC 20 Filterbank Output Parameters Chapter 6 Filter Effects 137 138 EVOC 20 Filterbank Formant Filter Parameters The parameters in this section provide precise level and frequency control of the filters Formant Shift knob i High and Low Frequency Lowest button parameters evVoc 20 FB Boost A knob Fade AB slider IG Formant Shilt Slope pop up menu 1 Bands lowest Highest button Boost B knob Resonance knob Frequency band faders Bands value field e High and Low Frequency parameters Determine the lowest and highest frequencies allowed to pass by the filter banks Frequ
165. is panned hard left At 90 the center of the stereo base is panned hard right Higher values move the center of the stereo base back toward the center of the stereo mix but this also has the effect of swapping the stereo sides of the recording For example at values of 180 or 180 the center of the stereo base is dead center in the mix but the left and right sides of the recording are swapped The following applies when working with MS signals At 90 the middle signal is panned hard left At 90 the middle signal is panned hard right Higher values move the middle signal back toward the center of the stereo mix but this also has the effect of swapping the side signals of the recording For example at values of 180 or 180 the middle signal is dead center in the mix but the left and right sides of the side signal are swapped Getting to Know Stereo Miking Techniques There are three commonly used stereo miking variants used in recording AB XY and MS A stereo recording put simply is one that contains two channel signals AB and XY recordings both record left and right channel signals but the middle signal is the result of combining both channels MS recordings record a real middle signal but the left and right channels need to be decoded from the side signal which is the sum of both left and right channel signals Understanding AB Miking In an AB recording two microphones commonly omnidir
166. is section describes the filter effects pedals Chapter 1 Amps and Pedals 47 Stompbox Description Auto Funk An auto wah filter effect Sensitivity sets a threshold which determines how the filter responds to incoming signal levels Cutoff sets the center frequency for the filter The BP LP switch enables either a bandpass or lowpass filter circuit Signal frequencies just above and below the cutoff point are filtered when the BP switch position is chosen When the LP switch position is active only signals below the cutoff point are allowed through the filter The Hi Lo switch chooses one of two preset filter resonance settings The Up Down switch activates a positive or negative modulation direction the wah filtering occurs above or below the source signal frequency Classic Wah A funky wah effect straight from 1970 s TV police show soundtracks You control it by dragging the pedal Modern Wah A more aggressive wah effect You control it by dragging the pedal Mode enables you to choose from the following Retro Wah Modern Wah Opto Wah 1 Opto Wah 2 Volume Each has a different tonal quality The Q knob determines the resonant characteristics Low Q values affect a wider frequency range resulting in softer resonances High Q values affect a narrower frequency range resulting in more pronounced emphasis Dynamics Pedals This section describes the dynamics pedals Stompbox Description Squash Compressor A s
167. is typically used as a mastering tool and is therefore generally inserted into master or output channel strips The way you use the Linear Phase EQ is obviously dependent on the audio material and what you intend to do with it but a useful workflow for many situations is as follows Set the Linear Phase EQ to a flat response no frequencies boosted or cut turn on the Analyzer and play the audio signal Keep an eye on the graphic display to see which parts of the frequency spectrum have frequent peaks and which parts of the spectrum stay at a low level Pay particular attention to sections where the signal distorts or clips Use the graphic display or parameter controls to adjust the frequency bands as desired You can reduce or eliminate unwanted frequencies and you can raise quieter frequencies to make them more pronounced You can adjust the center frequencies of bands 2 through 7 to affect a specific frequency either one you want to emphasize such as the root note of the music or one you want to eliminate such as hum or other noise While doing so change the Q parameter s so that only a narrow range of frequencies are affected or widen it to alter a broad area Each EQ band has a different color in the graphic display You can graphically adjust the frequency of a band by dragging horizontally Drag vertically to adjust the amount of gain for the band For bands 1 and 8 the slope values can be changed only in the parameter area below t
168. itself using an internal sidechain This means that the incoming signal modulates its own phase position The input signal only passes the delay line and is not affected by any other process The Mix parameter blends the effect signal with the original signal Monitor button Enable to hear the input signal in isolation Disable to hear the mixed signal Cutoff knob and field Sets the center cutoff frequency of the lowpass filter Resonance knob and field Emphasizes frequencies surrounding the cutoff frequency Display Shows the impact of parameters on the signal Mix slider and field Adjusts the percentage of the effect signal mixed with the original signal Max Modulation slider and field Sets the maximum delay time Intensity slider and field Sets the amount of modulation applied to the signal Chapter 3 Distortion Effects Phase Reverse checkbox Extended Parameters area Enable to reduce the delay time on the right channel when input signals that exceed the cutoff frequency are received Available only for stereo instances of the Phase Distortion effect Chapter 3 Distortion Effects 83 Dynamics Processors 4 The Dynamics processors 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 low
169. ity guitar amplification systems Guitar Amp Pro can also be used for experimental sound design and processing You can freely use it with other instruments applying the sonic character of a guitar amp to a trumpet or vocal part for example The amplifier speaker and EQ models emulated by Guitar Amp Pro can be combined in a number of ways to radically or subtly alter the tone Virtual microphones are used to pick up the signal of the emulated amplifier and cabinet You can choose from two different microphone types and you can reposition them Guitar Amp Pro also emulates classic guitar amplifier effects including reverb vibrato and tremolo Chapter 1 Amps and Pedals 29 30 The Guitar Amp Pro window is organized into sections according to different kinds of parameters Amp section CERTI SPEAKER Microphone Position section Effects section Microphone Type section Amp section The model parameters at the top are used to choose the type of amp EQ model and speaker See Building Your Guitar Amp Pro Model Farther down in the Amp section the knobs in the V shaped formation are used to set tone gain and level See Using Guitar Amp Pro s Gain Tone Presence and Master Controls Effects section Provides parameters to control the built in tremolo vibrato and reverb effects See Using Guitar Amp Pro s Reverb Effect and Using Guitar Amp Pro s Tremolo and Vibrato Effects Microphone Position and Type secti
170. ixed signal is sent through a further lowpass filter All three filters have a slope of 6 dB octave This unique combination of filters allows for gaps in the frequency spectra that can sound quite good with this sort of nonlinear distortion Drive slider and field Sets the amount of gain applied to the input signal After being amplified by the Drive value the signal passes through a highpass filter Tone slider and field Sets the cutoff frequency in Hertz of the highpass filter Clip Circuit display Shows the impact of all parameters with the exception of the High Shelving filter parameters Symmetry slider and field Sets the amount of nonlinear asymmetrical distortion applied to the signal Clip Filter slider and field Sets the cutoff frequency in Hertz of the first lowpass filter Chapter 3 Distortion Effects 79 80 Mix slider and field Sets the ratio between the effect wet signal and original dry signals following the Clip Filter Sum LPF knob and field Sets the cutoff frequency in Hertz of the lowpass filter This processes the mixed signal High Shelving Frequency knob and field Sets the frequency in Hertz of the high shelving filter 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 these types of treble controls however you can boost or cut the signal by up to 30 dB with the Gain paramet
171. ker Link button Located between the Amp and Speaker pop up menus links these pop up menus so that when you change the amp model the speaker associated with that amp is loaded automatically Choosing a Guitar Amp Pro Equalizer The EQ pop up menu and the Amp EQ Link button are near the top of the interface EQ pop up menu Contains the following EQ models British1 British2 American and Modern Each EQ model has unique tonal qualities that affect the way the Bass Mids and Treble knobs in the Amp section respond Amp EQ Link button Located between the Amp and EQ pop up menus links these pop up menus so that when you change the amp model the EQ model associated with that amp is loaded automatically Each amp model has a speaker and EQ model associated with it The default combinations of amp speaker and EQ settings recreate a well known guitar sound You are of course free to combine any speaker or EQ model with any amp by turning off the two Link buttons Chapter 1 Amps and Pedals Using Guitar Amp Pro s Gain Tone Presence and Master Controls The Gain Bass Mids Treble Presence and Master knobs run from left to right in the V shaped formation in the upper half of the interface Gain knob Sets the amount of pre amplification applied to the input signal This control has different effects depending on which Amp model is chosen For example when you are using the British Clean amp model the maximum Gain setting produ
172. l Tuning button in the Pitch Correction window the host application Tuning settings will be used for the pitch correction process If this parameter is turned off you can use the Ref Pitch field to freely set the desired reference tuning to the root key note For example the intonation of a vocal line is often slightly sharp or flat throughout an entire song Use the Reference Pitch parameter to address this issue at the input of the pitch detection process Set the Reference Pitch to reflect the constant pitch deviation in cent values This allows the pitch correction to perform more accurately Chapter 10 Pitch Effects Note Tunings that differ from software instrument tuning can be interesting when you want to individually correct the notes of singers in a choir If all voices were individually and perfectly corrected to the same pitch the choir effect would be partially lost You can prevent this by de tuning the pitch corrections individually Automating the Pitch Correction Effect The Pitch Correction effect can be fully automated This means that you can automate the Scale and Root parameters to follow harmonies in the project Depending on the accuracy of the original intonation setting the appropriate key Scale parameter may suffice Less precise intonations may need more significant changes to the Scale and Root parameters Pitch Shifter II Pitch Shifter Il provides a simple way to combine a pitch shifted version of the s
173. lay the entire impulse response waveform in the main display Any envelope length changes are automatically reflected AandD buttons Click to limit the Zoom to Fit function to the attack and decay portions of the currently selected envelope shown in the main display The A and D buttons are available only when you are viewing the volume and filter envelopes Chapter 12 Space Designer Convolution Reverb Setting Space Designer s Envelope Parameters You can edit the volume and filter envelopes of all IRs and the density envelope of synthesized IRs All envelopes can be adjusted both graphically in the main display and numerically in the parameter bar Whereas some parameters are envelope specific all envelopes consist of the Attack Time and Decay Time parameters The combined total of the Attack Time and Decay Time parameters is equal to the total length of the synthesized or sampled impulse response unless the Decay time is reduced See Setting Impulse Response Lengths in Space Designer The large nodes are value indicators of the parameters shown in the parameter bar below Init Level Attack Time Decay Time and so on If you edit any numerical value in the parameter bar the corresponding node moves in the main display To move an envelope node graphically in Space Designer Drag the node in one of the available directions Two arrows are shown when you move the cursor over any node in the main display indicating possibl
174. level Sustain knob and field Sets the level of the reverb that remains constant throughout the sustain phase It is expressed as a percentage of the full scale volume of the reverb signal Hold knob and field Sets the duration the time of the sustain phase Release knob and field Sets the time that the reverb takes to fade out completely after it has completed the sustain phase Chapter 11 Reverb Effects EnVerb Sound Parameters EnVerb offers the following tone control parameters Density slider and field Sets the reverb density Spread slider and field 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 slider and field Frequencies above the set value are filtered out of the reverb tail Crossover slider and field Defines the frequency that is used to split the input signal into two frequency bands for independent processing Low Freq Level slider and field Determines the relative level of reverb signal frequencies below the crossover frequency In most cases you get better sounding results when you set negative values for this parameter Chapter 11 Reverb Effects 217 218 GoldVerb GoldVerb allows you to edit both the early reflections and diffuse reverb tail separately making it easy to precisely emulate real rooms Early Reflections parameters Balance ER Reverb slider Mix slider and field Rev
175. ll outside these boundaries will be cut The length of the blue bar represents the frequency range for both analysis and synthesis unless Formant Stretch or Formant Shift are used as discussed in Formant Stretch knob and Formant Shift knob below You can move the entire frequency range by dragging the horizontal blue bar at the top The silver handles on either end of the blue bar set the Low Frequency and High Frequency values respectively You can also use the numeric fields to adjust the frequency values separately e Lowest button Click to determine whether the lowest filter band acts as bandpass or highpass filter In the Bandpass setting the frequencies below the lowest bands and above the highest bands are ignored In the Highpass setting all frequencies below the lowest bands are filtered e Highest button Click to determine whether the highest filter band acts as bandpass or lowpass filter In the Bandpass setting the frequencies below the lowest bands and above the highest bands are ignored In the Lowpass setting all frequencies above the highest bands are filtered Formant Stretch knob Alters the width and distribution of all bands in the synthesis filter bank This can be a broader or narrower frequency range than that defined by the blue bar see High and Low Frequency parameters above Formant Shift knob Moves all bands in the synthesis filter bank up and down the frequency spectrum Chap
176. lse response recording of an acoustic environment This is convolved with the incoming audio signal to place it in the acoustic space provided by the IR sample rate 2 lt To activate IR Sample mode Click the IR Sample button in the circular area to the left of the main display and then select the desired impulse response file from any folder Note If you have already loaded an impulse response file clicking the IR Sample button switches the mode from Synthesized IR to IR Sample mode To manage the loaded IR file Click the down arrow next to the IR Sample button to open a pop up menu with the following commands Load IR Loads an impulse response sample without changing the envelopes Load IR amp Init Loads an impulse response sample and initializes the envelopes Show in Finder Opens a Finder window that shows the location of the currently loaded IR file Open IR Utility Opens Impulse Response Utility where you can record and edit impulse responses See the mpulse Response Utility Manual for details on use All impulse responses that ship with Logic Pro are installed in the Library Audio Impulse Responses Apple folder Deconvolution files have an sdir file extension Chapter 12 Space Designer Convolution Reverb Any mono stereo AIFF SDII or WAV file can be used as an IR In addition surround formats up to 71 discreet audio files and B format audio files that comprise a single surround IR can also be
177. lue bars When the level exceeds 0 dB the portion of the bar above the 0 dB mark turns red Current peak values are displayed numerically in dB increments above the Level Meter Click in the display to reset peak values Using the Surround MultiMeter Balance Correlation Parameters The Surround MultiMeter s Correlation Meter gauges the balance or sound placement between all incoming signals Strongly correlated signals are shown as sharp markers and less strongly correlated signals as a blurred area Activate the Surround MultiMeter s Balance Correlation button to view the Correlation Meter in the main display As WFast Goniometer Chapter 8 Metering Tools Depending on the chosen surround format a number of points that indicate speaker positions are shown L R C Ls Rs in a 5 1 configuration is displayed in the figure Lines connect these points The center position of each connecting line is indicated by a blue marker A gray ball indicates the surround field sound placement As you move the surround panner of the channel strip the ball in the Correlation Meter mirrors your movements The blue markers also move in real time with shaded gray lines indicating the divergence from the centered positions on each of the connecting lines The LFE channel Correlation Meter is shown at the bottom of the main display The horizontal area around the white correlation indicator denotes phase correlation deviations in real time This
178. ly left and right channel coherence but also the front to rear coherence Analyzer Balance Correlation Peer Sa Reset Goniometer button Displays the Goniometer results in the main display Auto Gain field Sets the amount of display compensation for low input levels You can set Auto Gain levels in 10 increments or set it to off Note To avoid confusion with the Auto Gain parameter found in other effects and processors such as the compressors Auto Gain is only used as a display parameter in the meters It increases display levels to enhance readability It does not change the actual audio levels Decay field Determines the time it takes for the Goniometer trace to fade to black L R Ls Rs Both buttons Determine which channel pairs are shown in the main display When you are using the Surround MultiMeter in configurations with exactly two channel pairs quad 5 1 and 6 1 configurations the Goniometer can display both pairs if you select Both One pair for L R appears in the upper half of the main display and one for Ls Rs appears in the lower half Chapter 8 Metering Tools 177 178 Using the Surround MultiMeter Level Meter The Level Meter displays the current signal level on a logarithmic decibel scale The signal level for each channel is represented by a blue bar RMS and Peak levels are shown simultaneously with RMS levels appearing as dark blue bars and Peak levels appearing as light b
179. menu and field Click to choose different pitch quantization grids from the Scale pop up menu See Defining the Pitch Correction Effect s Quantization Grid Keyboard Click a key to exclude the corresponding note from pitch quantization grids This effectively removes this key from the scale resulting in note corrections that are forced to the nearest available pitch key See Excluding Notes from Pitch Correction Byp ass buttons Click to exclude the corresponding note from pitch correction In other words all notes that match this pitch will not be corrected This applies to both user and built in scale quantization grids See Excluding Notes from Pitch Correction Bypass All button Provides a quick way to compare the corrected and original signals or for automation changes Show Input and Show Output buttons Click to display the pitch of the input or output signal respectively on the notes of the keyboard Chapter 10 Pitch Effects Correction Amount display Indicates the amount of pitch change The red marker indicates the average correction amount over a longer time period You can use the display when discussing and optimizing the vocal intonation with a singer during a recording session Response slider and field Determines how quickly the voice reaches the corrected destination pitch Singers use portamenti and other gliding techniques If you choose a Response value that s too high seamless portamenti turn into semi
180. mode See Using Pedalboard s Routing Area for details on use Chapter 1 Amps and Pedals 49 Delay Effects 2 Delay effects store the input signal and hold it for a short time before sending it to the effect input or output The held and delayed signal is repeated after a given time period creating a repeating echo effect Each subsequent repeat is a little quieter than the previous one Most delays also allow you to feed a percentage of the delayed signal back to the input This can result in a subtle chorus like effect or cascading chaotic audio output 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 You can use delays to double individual sounds to resemble a group of instruments playing the same melody to create echo effects to place the sound in a large space to generate rhythmic effects or to enhance the stereo position of tracks in a mix Delay effects are generally used as channel insert or bussed effects They are rarely used onan overall mix in an output channel unless you re trying to achieve an unusual effect This chapter covers the following Delay Designer p 51 e Echo p 72 e Sample Delay p 72 Stereo Delay p 73 Tape Delay p 75 Delay Designer Delay Designer is a multitap delay Unlike traditional delay units that offer only one or two delays or taps that may or m
181. mpact when using the Space Designer as a mono plug in Spread Spread 0 56 s Xover 710Hz Spread knob and field Extends the stereo or surround base to frequencies that fall below the frequency determined by the Xover crossover parameter Ata Spread value of 0 00 no stereo or surround information is added although the inherent stereo or surround information of the source signal and reverb are retained At a value of 1 00 the left and right channel divergence is at its maximum Xover knob and field Sets the crossover frequency in Hertz Any synthesized impulse response frequency that falls below this value will be affected by the Spread parameter at values over 0 Automating Space Designer Space Designer cannot be fully automated unlike most other plug ins which can be This is because Space Designer needs to reload the impulse response and recalculate the convolution before audio can be routed through it Chapter 12 Space Designer Convolution Reverb 247 You can however record edit and play back any movement of the following Space Designer parameters in a suitable host application Stereo Crossfeed e Direct Output e Reverb Output 248 Chapter 12 Space Designer Convolution Reverb Specialized Effects and Utilities 13 Logic Pro includes a bundle of specialized effects and utilities designed to address tasks often encountered during audio production As examples of where these pro
182. n the vowel a for example If you want to compensate for people s inability to sustain sung notes for a long period without taking a breath you can use the Freeze button If the synthesis signal needs to be sustained but the analysis source signal a vocal part is not sustained use the Freeze button to lock the current formant levels of a sung note even during gaps in the vocal part between words in a vocal phrase The Freeze parameter can be automated which may be useful in this situation Setting the Number of Bands The greater the number of bands the more precisely the sound can be reshaped As the number of bands is reduced the source signal s frequency range is divided up into fewer bands and the resulting sound will be formed with less precision by the synthesis engine You may find that a good compromise between sonic precision allowing incoming signals speech and vocals in particular to remain intelligible and resource usage is around 10 to 15 bands Tip To ensure the best possible pitch tracking it is essential to use a mono signal with no overlapping pitches Ideally the signal should be unprocessed and free of background noises Using a signal processed with even a slight amount of reverb for example will produce strange and probably undesirable results Even stranger results will result when a signal with no audible pitch such as drum loop is used In some situations however the resulting artifacts might
183. n the Tap display the extra taps in the Clipboard are ignored Reset sound parameters to default values Resets all parameters of all selected taps except the delay time to the default values 2x delay time Doubles the delay time of all selected taps For example the delay times of three taps are set as follows Tap A 250 ms Tap B 500 ms Tap C 750 ms If you select these three taps and choose the 2 x delay time shortcut menu command the taps will be changed as follows Tap A 500 ms Tap B 1000 ms Tap C 1500 ms In other words a rhythmic delay pattern will unfold half as fast In musical terms it will be played in half time 1 2 x delay time Halves the delay time of all selected taps Using the example above use of the 1 2 x delay time shortcut menu command changes the taps as follows Tap A 125 ms Tap B 250 ms Tap C 375 ms In other words a rhythmic delay pattern will unfold twice as fast In musical terms it will be played in double time Delete tap s Deletes all selected taps Resetting Delay Designer Tap Values You can use Delay Designer s Tap display and Tap parameter bar to reset tap parameters to their default values Chapter 2 Delay Effects 67 68 To reset the value of a tap Do one of the following In the Tap display Option click a tap to reset the chosen parameter to its default setting If multiple taps are selected Option clicking any tap will reset the chosen
184. nal level after it falls below the threshold level Set a higher Release value to smooth out dynamic differences in the signal Set lower Release values if you want to emphasize dynamic differences Important The discussion above is highly reliant on not only the type of source material but also the compression ratio and threshold settings Setting the Compressor Knee The Knee parameter determines whether the signal is slightly or severely compressed as it approaches the threshold level Setting a Knee value close to 0 zero results in no compression of signal levels that fall just below the threshold while levels at the threshold are compressed by the full Ratio amount This is known as hard knee compression which can cause abrupt and often unwanted transitions as the signal reaches the threshold Increasing the Knee parameter value increases the amount of compression as the signal approaches the threshold creating a smoother transition This is called soft knee compression Setting Other Compressor Parameters As the compressor reduces levels the overall volume at its output is typically lower than the input signal You can adjust the output level with the Gain slider You can also use the Auto Gain parameter to compensate for the level reduction caused by compression choose either 12 dB or 0 dB When you use the Platinum circuit type the Compressor can analyze the signal using one of two methods Peak or root mean square R
185. nd effect signals This can result in flanger chorus effects or in metallic sounding modulations particularly when used with high Feedback values Chapter 1 Amps and Pedals 45 Stompbox Description Roswell Ringer A ring modulation effect that can make incoming audio sound metallic or unrecognizable can deliver tremolos brighten up signals and more The Freq knob sets the core filter cutoff frequency Fine is a fine tuning knob for the filter frequency The Lin Exp switch determines if the frequency curve is linear 12 notes per octave or exponential FB feedback determines the amount of the effect signal that is routed back into the input This can change the tonal color can make the effect more pronounced or can do both Balance between the original and effect signals is set with the Mix knob See Ringshifter for background information on ring modulation Roto Phase A phaser effect that adds movement to and alters the phase of the signal Rate sets the modulation speed and can run freely or be synchronized with the host application tempo by enabling the Sync button When synchronized you can specify bar beat and note values including triplets and dotted notes with the Rate knob Intensity sets the strength of the effect The Vintage Modern switch activates a fixed frequency internal EQ when switched to Vintage and deactivates it when switched to Modern Spin Box Emulation of a Leslie rotor speaker cabinet commo
186. nd highpass HP filters Feedback slider and field Determines the amount of the effect signal that is routed back into the input of the effect Phaser Sweep Section Ceiling and Floor sliders and fields Use the individual slider handles to determine the frequency range 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 The 4 6 8 10 and 12 settings put five different phaser algorithms at your fingertips All are modeled on analog circuits with each designed for a specific application You are free to select odd numbered settings 5 7 9 11 which strictly speaking don t generate actual phasing The more subtle comb filtering effects produced by odd numbered settings can however come in handy on occasion Env Follow slider and field Determines the impact of incoming signal levels on the frequency range as set with the Ceiling and Floor controls Chapter9 Modulation Effects Phaser LFO Section LFO 1 and LFO 2 Rate knobs and fields Set the speed for each LFO LFO Mix slider and fields Determines the ratio between the two LFOs Env Follow slider and field Determines the impact of incoming signal levels on the speed of LFO 1 Phase knob and field Available only in stereo and surround instances Controls the phase relationship between the individual channel modulations At 0 the extreme
187. nd lines below the center position denote pans to the right Left blue and right green channels are easily identified In stereo input stereo output configurations the Pan parameter adjusts the stereo balance not the position of the tap in the stereo field The Pan parameter appears as a dot on the tap which represents stereo balance Drag the dot up or down the tap to adjust the stereo balance By default stereo spread is set to 100 To adjust this drag either side of the dot As you do so the width of the line extending outwards from the dot changes Keep an eye on the Spread parameter in the Tap parameter bar while you are adjusting In surround configurations the bright line represents the surround angle For more information see Working with Delay Designer in Surround Chapter 2 Delay Effects 65 66 Editing Taps in Delay Designer s Tap Parameter Bar The Tap parameter bar provides instant access to all parameters of the chosen tap The Tap parameter bar also provides access to several parameters that are not available in the Tap display such as Transpose and Flip Editing in the Tap parameter bar is fast and precise when you want to edit the parameters of a single tap All parameters of the selected tap are available with no need to switch display views or estimate values with vertical lines If you choose multiple taps in the Tap display the values of all selected taps are changed relative to each other Op
188. ng sound when used in combination with a slow oscillator sweep Comb filtering effects are created by using high Feedback settings with a short delay time less than 10 ms Use of longer delay times in conjunction with high Feedback settings creates continuously rising and falling frequency shift effects Stereo Width knob and field Determines the breadth of the effect signal in the stereo field Stereo Width affects only the effect signal of the Ringshifter not the dry input signal Env Follower slider and field Determines the amount of Dry Wet parameter modulation by the input signal level LFO slider and field Sets the LFO modulation depth of the Dry Wet parameter Rotor Cabinet Effect 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 that pick up the sound Basic Rotor Speaker Parameters The Rotor Cabinet offers the following basic rotor speaker parameters Rotor Speed buttons These switch the rotor speed in the following ways Chorale Slow movement Tremolo Fast movement e Brake Stops the rotor Chapter9 Modulation Effects 197 198 Cabinet Type pop up menu You can choose from the following cabinet models Wood Mimics a Leslie with a wooden enclosure and sounds like the Leslie 122 or 147 models Proline Mimics a Leslie with a more open enclosure
189. nly used with the Hammond B3 organ Cabinet sets the type of speaker box Fast Rate sets the maximum modulation speed only applies when Fast button is active Response determines the amount of time required for the rotor to reach its maximum and minimum speed Drive increases the input gain introducing distortion to the signal The Bright switch activates a high shelving filter when turned on The Slow Brake and Fast buttons determine how the speaker behaves Slow rotates the speaker slowly Fast rotates the speaker quickly up to the maximum speed determined by the Fast Rate knob Brake stops the speaker rotation See Rotor Cabinet Effect for background information on the Leslie effect Total Tremolo A flexible tremolo effect modulation of the signal level Rate sets the modulation speed and can run freely or be synchronized with the host application tempo by enabling the Sync button When synchronized you can specify bar beat and note values including triplets and dotted notes Depth sets the strength of the effect Wave and Smooth work in combination to alter the waveform shape of the LFO This enables you to create floating changes in level or abrupt steps Volume determines the output level of the effect The 1 2 and 2x Speed buttons immediately halve or double the current Rate value Hold down the Speed Up and Slow Down buttons to gradually accelerate or reduce the current Rate value to the maximum or minimum possible value
190. nsion for the selected band e Peak RMS fields Enter a smaller value for shorter peak detection or a larger value for RMS detection in milliseconds Attack fields Set the amount of time before compression starts for the selected band after the signal exceeds the threshold Release fields Set the time required before compression stops on the selected band after the signal falls below the threshold Band on off buttons 1 2 3 and 4 Enable disable each band 1 to 4 When enabled the button is highlighted and the corresponding band appears in the graphic display area above Byp ass buttons Enable to bypass the selected frequency band Solo buttons Enable to hear compression on only the selected frequency band Level meters The bar on the left shows the input level and the bar on the right shows the output level Threshold arrows Two arrows appear to the left of each Level meter The upper arrow adjusts the Compression Threshold Compr Thrsh The lower arrow adjusts the Expansion Threshold Expnd Thrsh Multipressor Output Section Auto Gain pop up menu When you choose On it references the overall processing of the signal to 0 dB making the output louder Lookahead value field Adjusts how far the effect looks forward in the incoming audio signal in order to react earlier to peak volumes and therefore achieve smoother transitions Out slider Sets the overall gain at the Multipressor output L
191. nvelopes and the EQ to default values e Volume Env button Displays the volume envelope in the foreground of the main display The other envelope curves are shown as transparencies in the background See Working with Space Designer s Volume Envelope Filter Env button Displays the filter envelope in the foreground of the main display The other envelope curves are shown as transparencies in the background See Working with Space Designer s Filter Density Env button Displays the density envelope in the foreground of the main display The other envelope curves are shown as transparencies in the background See Working in Space Designer s Synthesized IR Mode EQbutton Displays the four band parametric EQ in the main display See Working with Space Designer s EQ Reverse button Reverses the impulse response and envelopes When the impulse response is reversed you are effectively using the tail rather than the front end of the sample You may need to change the Pre Dly and other parameter values when reversing Zooming and Navigating Space Designer s Envelope View When displaying envelopes the main display offers the following zoom and navigation parameters not shown in EQ view Overview display Zoom to fit A Overview display Indicates which portion of the impulse response file is currently visible in the main display helping you to orientate yourself when zooming Zoom to Fit button Click to disp
192. o add a pedal to the Pedal area Do one of the following Drag the stompbox that you want to insert from the Pedal Browser to the appropriate Pedal area position This can be to the left to the right or in between existing pedals Ensure that no pedal is selected in the Pedal area then double click a stompbox in the Pedal Browser to add it to the right of all existing effects in the Pedal area Note You insert Mixer and Splitter utility pedals in a different way See Using Pedalboard s To change an effect pedal position in the Pedal area Drag the stompbox to a new position either to the right or the left Automation and bus routings if active are moved with the effect pedal For information about automation and bus routings see Using Pedalboard s Routing Area Note There are two exceptions to the bus routing rule If the dragged pedal is the only pedal between a Splitter and Mixer utility both utility pedals are automatically removed If the second Bus B is not active at the destination the pedal is inserted into Bus A To change a Mixer utility position in the Pedal area Drag the Mixer utility to a new position either to the left or the right When moved to the left The downmix of Bus A and B will occur at the earlier insertion point Relevant effect pedals are moved to the right and are inserted into Bus A When moved to the right The downmix of Bus A and B will occur at the later insertion point Relevant
193. odels are inspired by the powerful ultra high gain amplifier heads that put the chunk into modern hard rock and metal music All are paired with 4 x 12 cabinets Their signature tones range from heavy distortion to extremely heavy distortion If you want powerful lows razor edged highs and serious sustain these are the models you should look to first Chapter 1 Amps and Pedals Model Description Modern American Stack A powerful ultra high gain amp that is ideal for heavy rock and metal Use the Mids knob to set an ideal amount of scoop or boost High Octane Stack Although a powerful high gain amp this model offers a smooth transition between gain settings and excellent natural compression It is a great choice for fast soloing and for two and three note chords Turbo Stack An aggressive sounding amp with spiky highs and noisy harmonics especially at high gain settings Try the Turbo Stack when you need to slice through a mix Tip Combining the Turbo Stack with distortion and fuzz pedals may actually diminish the amp s edge A dry sound is often the best choice for high impact riffs Additional Combos The combos and utility models in this category are versatile amps that can be used for a wide variety of musical styles Model Description Studio Combo A 1x 12 combo based on boutique combos of the 1980s and 1990s that use multiple gain stages to generate smooth sustain heavy distortion without sacrificing bol
194. of the delay line with the original signal Mixing a vibrato signal with an original statically pitched signal results in a chorus effect This organ style chorus sounds different from the Chorus plug in If the CO setting is chosen neither the chorus nor vibrato is enabled Chorus Int knob Sets the intensity of a chosen chorus effect type If a vibrato effect type is chosen this parameter has no effect Stereo Phase knob When set to a value between 0 and 360 Stereo Phase determines the phase relationship between left and right channel modulations thus enabling synchronized stereo effects If you set the knob to free you can set the modulation speed of the left and right channel independently Rate Left knob Sets the modulation speed of the left channel when Stereo Phase is set to free If Stereo Phase is set to a value between 0 and 360 Rate Left sets the modulation speed for both the left and right channels Rate Right has no function when in this mode Rate Right knob Sets the modulation speed of the right channel when Stereo Phase is set to free Chapter9 Modulation Effects Spreader Spreader widens the stereo spectrum of a signal The Spreader effect periodically shifts the frequency range of the original signal thus changing the perceived width of the signal The delay between channels can also be specified in samples adding to the perceived width and channel separation of a stereo input si
195. ol the integrated guitar effects See Getting to Know Amp Designer s Effects Parameters Microphone parameters Located slightly above the right end of the black bar at the bottom these parameters are used to set the type and position of the microphone that captures the amplifier and cabinet sound See Setting Amp Designer Microphone Parameters Chapter 1 Amps and Pedals To switch between full and smaller versions of the interface Click the disclosure triangle between the Cabinet and Mic pop up menus in the full interface to switch to the smaller version To switch back to the full interface click the disclosure triangle beside the Output field in the small interface You can access all the parameters with the exception of microphone selection and positioning in the small interface Click here in Click here in full interface small interface Choosing an Amp Designer Model You can choose a preconfigured model consisting of an amplifier a cabinet an EQ type and a microphone type from the Model pop up menu at the left end of the black bar at the bottom of the Amp Designer interface Your choices include several combinations in each of the following categories Tweed Combos e Classic American Combos e British Stacks British Combos e British Alternatives e Metal Stacks Additional Combos Chapter 1 Amps and Pedals 13 Tweed Combos The Tweed models are based on American combos from the 1950s and
196. on phases of the channels e Distribution pop up menu Available only in surround instances Defines how phase offsets between individual channels are distributed in the surround field You can choose from circular lefteoright frontcrear random and new random distributions to randomize the phase choose new random Offset slider and field Extended Parameters area Sets the amount that the modulation cycle is shifted to the left or right resulting in subtle or significant tremolo variations Chapter9 Modulation Effects Pitch Effects 10 You can use the Pitch effects included in Logic Pro to transpose or correct the pitch of audio signals These effects can also be used for creating unison or slightly thickened parts or even for creating harmony voices This chapter covers the following e Pitch Correction Effect p 203 e Pitch Shifter II p 207 e Vocal Transformer p 208 Pitch Correction Effect You can use the Pitch Correction effect to correct the pitch of incoming audio signals Improper intonation is a common problem with vocal tracks for example The sonic artifacts that can be introduced by the process are minimal and can barely be heard as long as your corrections are moderate Pitch correction works by accelerating and slowing down the audio playback speed ensuring that the input signal sung vocal always matches the correct note pitch If you try to correct larger int
197. ons These sections enable you to set the position and type of the microphone See Setting Guitar Amp Pro Microphone Parameters Building Your Guitar Amp Pro Model An amplifier model consists of an amplifier speaker cabinet EQ type and microphone type You can create your own hybrids of different amplifiers cabinets and so on using the pop up menus at the top center of the interface You choose the microphone position and type in the yellow areas to the left and right You can use the Settings menu to save your new hybrid amp combos as setting files which also include any parameter changes you may have made How to build your amplifier model is described in the following sections Choosing a Guitar Amp Pro Amplifier e Choosing a Guitar Amp Pro Speaker Cabinet Chapter 1 Amps and Pedals Choosing a Guitar Amp Pro Equalizer e Setting Guitar Amp Pro Microphone Parameters Choosing a Guitar Amp Pro Amplifier You can choose an amplifier model from the Amp pop up menu near the top of the interface UK Combo 30W Neutral sounding amp well suited for clean or crunchy rhythm parts e UKTop 50W Quite aggressive in the high frequency range well suited for classical rock sounds US Combo 40W Clean sounding amp model well suited for funk and jazz sounds US Hot Combo 40W Emphasizes the high mid frequency range making this model ideal for solo sounds US Hot Top 100W This amp produces very fat sounds even at l
198. ools found in the Utility category can help with routine tasks and situations that you may encounter during production such as the following Gain plug ins are used to adjust the level or phase of input signals I O Utility enables you to integrate external audio effects into your host application mixer Test Oscillator generates a static frequency or sine sweep This chapter covers the following Down Mixer p 259 Gain Plug in p 260 1 0 Utility p 261 e Multichannel Gain p 263 e Test Oscillator p 263 Down Mixer You can use the Down Mixer to adjust the input format of the surround master channel strip This allows you to quickly check a surround mix in stereo for example gt lt 4 pP View gt Show CS v Show Insert gt Bypass Down Mixer To Stereo 0 0dB 0 0dB 0 0dB 0 0dB 0 0dB Important The desired surround format is chosen from the Insert menu when you insert the plug in Choices include To Stereo To Quad To LCRS 259 260 Channel mapping panning and downmixing are automatically handled behind the scenes You do however have some control over the mix Use the Level sliders to control the respective channel levels The number and names of sliders is dependent on the chosen plug in format Gain Plug in Gain amplifies or reduces the signal by a specific decibel amount It is very useful for quick level adjustments when you are working with automated tracks during post proce
199. open chords one reason why they were embraced by the Brit pop bands of the 1990s The Stadium amps are famed for their ability to play ultra loud without dissolving into mushy distortion They retain crisp treble and superb note definition even at maximum gain settings Model Description Sunshine Stack A robust sounding head paired with a 4 x 12 cabinet It s a great choice for powerful pop rock chords Small Sunshine Combo A 1x 12 combo based on a modern amp known for a big amp sound It is brighter than the Sunshine Stack head with a touch of 1960s British Combo flavor Stadium Stack A classic head and 4x 12 cabinet configuration popular with 1970s arena rock bands Its tones are cleaner than other Amp Designer 4x 12 stacks while still retaining body and impact A good choice if you need power and clarity Stadium Combo A 2x 12 combo based on a modern amp The tone is a little smoother and rounder than that of the Stadium Stack Tip The tone of the Sunshine Stack can seem dark at times but a high Treble knob setting opens up the sound While the Small Sunshine Combo sounds great with its default 1x 12 cabinet it also shines through a 4 x 12 cabinet The Stadium amps can be slow to distort so most famous users have paired them with aggressive fuzz pedals Try combining it with Pedalboard s Candy Fuzz or Fuzz Machine stompboxes See Distortion Pedals and Pedalboard Metal Stacks The Metal Stack m
200. or arbitrarily placed or moved sources Speaker CTC Cross Talk Cancellation Setting for speaker playback allowing you to play back binaurally panned signals through stereo loudspeakers Good spatial reproduction is restricted to a limited range of listening positions on the symmetrical plane between the speakers e CTC Speaker Angle field and slider This parameter is effective only when the Speaker CTC compensation mode is chosen To achieve the best binaural effect enter the angle that your stereo speakers are turned towards the center the listening position For full details on using the Binaural Panner with the Binaural Post Processing plug in see the Logic Pro User Manual Direction Mixer You can use the Direction Mixer to decode middle and side audio recordings or to spread the stereo base of a left right recording and determine its pan position Chapter 7 Imaging Processors The Direction Mixer works with any type of stereo recording regardless of the miking technique used For information about XY AB and MS recordings see Getting to Know Stereo Miking Techniques e Input buttons Click the LR button if the input signal is a standard left right signal and click the MS button if the signal is middle and side encoded Spread slider and field Determines the spread of the stereo base in LR input signals Determines the level of the side signal in MS input signals See Using the Direction Mixer s Spread Param
201. or bell filter Frequencies close to the cutoff frequency are emphasized CryB This setting mimics the sound of the popular Cry Baby wah wah pedal Chapter 6 Filter Effects e Morl1 This setting mimics the sound of a popular wah wah pedal It features a slight peak characteristic e Morl2 This setting mimics the sound of a popular distortion wah wah pedal It has a constant Q uality Factor setting Auto Gain button The wah wah effect can cause wide variations in the output level Turning Auto Gain on compensates for this behavior and limits the output signal dynamics see Setting the Wah Wah Level with Auto Gain Wah Level knob Sets the amount of the wah filtered signal Relative Q slider Adjusts the main filter peak relative to the model setting thereby obtaining a sharper or softer wah wah sweep When set to a value of 0 the original peak level setting of the model is active Pedal Range slider Sets the sweep range of the Wah Wah filter when controlled with a MIDI foot pedal This parameter is designed to compensate for the differences in mechanical range between a MIDI foot pedal and a classic Wah Wah pedal see Setting the Pedal Range Auto Wah Parameters In addition to using MIDI foot pedals you can control the Wah Wah effect with the Auto Wah feature which continually performs a filter sweep across the entire range See Using the Fuzz Wah Depth knob Sets the depth of the Auto Wah effect When set to ze
202. osting the attack phase can add snap to a drum sound or it can amplify the initial pluck or pick sound of a stringed instrument Attenuating the attack causes percussive signals to fade in more softly You can also mute the attack making it virtually inaudible A creative use for this effect is alteration of the attack transients to mask poor timing of recorded instrument parts Boosting the release phase also accentuates any reverb applied to the affected channel strip Conversely attenuating 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 Only when you seriously raise the release phase which boosts the noise level of the original recording should you raise the Threshold slider a little This limits the Enveloper to affecting only the useful part of the signal Drastic boosting or cutting of either the release or attack phase may change the overall level of the signal You can compensate for this by adjusting the Out Level slider Generally you ll find that Attack Time values of around 20 ms and Release Time values of 1500 ms are good to start with Then adjust them for the type of signal that you re processing The Lookahead slider defines how far into the future of the incoming signal the Envelo
203. ound the center frequency that are affected Note The Q parameter of Band 1 and Band 8 has no effect when the slope is set to 6 dB Oct When the Q parameter is set to an extremely high value such as 100 these filters affect only a very narrow frequency band and can be used as notch filters Link button Activates Gain Q coupling which automatically adjusts the Q bandwidth when you raise or lower the gain on any EQ band to preserve the perceived bandwidth of the bell curve Analyzer Mode buttons Extended Parameters area Choose Peak or RMS Analyzer Decay slider and field Extended Parameters area Adjust the decay rate in dB per second of the Analyzer curve peak decay in Peak mode or an averaged decay in RMS mode Gain Q Couple Strength pop up menu Extended Parameters area Choose the amount of Gain Q coupling Set Gain Q Couple to strong to preserve most of the perceived bandwidth Light and medium settings allow some change as you raise or lower the gain The asymmetric settings feature a stronger coupling for negative gain values than for positive values so the perceived bandwidth is more closely preserved when you cut rather than boost gain Note If you play back automation of the Q parameter with a different Gain Q Couple setting the actual Q values will be different than they were when the automation was recorded Chapter5 Equalizers 119 120 Using the Linear Phase EQ The Linear Phase EQ
204. ow Master settings that result in broad sounds with a lot of oomph Custom 50W With the Presence parameter set to 0 this amp model is well suited for smooth fusion lead sounds British Clean GarageBand Simulates the classic British Class A combos used continuously since the 1960s for rock music without any significant modification This model is ideally suited for clean or crunchy rhythm parts British Gain GarageBand Emulates the sound of a British tube head and is synonymous with rocking powerful rhythm parts and lead guitars with a rich sustain American Clean GarageBand Emulates the traditional full tube combos used for clean and crunchy sounds American Gain GarageBand Emulates a modern Hi Gain head making it suitable for distorted rhythm and lead parts Clean Tube Amp Emulates a tube amp model with very low gain distortion only when using very high input levels or Gain Master settings Choosing a Guitar Amp Pro Speaker Cabinet The speaker cabinet can have a huge bearing on the type of tones you can extract from your chosen amplifier The speaker parameters are found near the top of the interface Speaker pop up menu You can choose one of the 15 speaker models UK 1x 12 open back Classic open enclosure with one 12 speaker neutral well balanced multifunctional UK2x 12 open back Classic open enclosure with two 12 speakers neutral well balanced multifunctional UK2x 12 closed
205. parameter to its default value for all selected taps In the Tap parameter bar Option click a parameter value to reset it to the default setting If multiple taps are selected Option clicking a parameter of any tap resets all selected taps to the default value for that parameter Synchronizing Taps in Delay Designer Delay Designer can either synchronize to the project tempo or run independently When you are in synchronized mode Sync mode taps snap to a grid of musically relevant positions based on note durations You can also set a Swing value in Sync mode which varies the precise timing of the grid resulting in a more laid back less robotic feel for each tap When you are not in Sync mode taps don t snap to a grid nor can you apply the Swing value When Sync mode is on a grid that matches the chosen Grid parameter value is shown in the Identification bar All taps are moved towards the closest delay time value on the grid Subsequently created or moved taps are snapped to positions on the grid When you save a Delay Designer setting the Sync mode status Grid and Swing values are all saved When you save a setting with Sync mode on the grid position of each tap is also stored This ensures that a setting loaded into a project with a different tempo to that of the project the setting was created in will retain the relative positions and rhythm of all taps at the new tempo Chapter 2 Delay Effects Note Delay Designer off
206. parameters are Threshold and Ratio The Threshold sets the floor level in decibels Signals that exceed this level are reduced by the amount set as the Ratio The Ratio is a percentage of the overall level the more the signal exceeds the threshold the more it is reduced A ratio of 4 1 means that increasing the input by 4 dB results in an increase of the output by 1 dB if above the threshold As an example with the Threshold set at 20 dB and the Ratio set to 4 1 a 16 dB peak in the signal 4 dB louder than the threshold is reduced by 3 dB resulting in an output level of 19 dB Chapter 4 Dynamics Processors Setting Suitable Compressor Envelope Times The Attack and Release parameters shape the dynamic response of the Compressor The Attack parameter determines the time it takes after the signal exceeds the threshold level before the Compressor starts reducing the signal Many sounds including voices and musical instruments rely on the initial attack phase to define the core timbre and characteristic of the sound When compressing these types of sounds you should set higher Attack values to ensure that the attack transients of the source signal aren t lost or altered When attempting to maximize the level of an overall mix it is best to set the Attack parameter to a lower value because higher values often result in no or minimal compression The Release parameter determines how quickly the signal is restored to its origi
207. pen back 1960s cabinet with glistening highs and surprising low mid body Brownface 1 x 12 A beautifully balanced 1960s open back cabinet It is smooth and rich sounding but with nice transparency Brownface 1 x 15 This early 1960s open back cabinet houses the largest speaker emulated by Amp Designer Its highs are clear and glassy and its lows are tight and focused Vintage British 4 x 12 This late 1960s closed back cabinet is synonymous with classic rock The tone is big and thick yet also bright and lively thanks to the complex phase cancellations between the four 30 watt speakers Modern British 4 x 12 A closed back 4 x 12 cabinet that is brighter and has a better low end than the Vintage British 4 x 12 with less mid range emphasis Brown 4 x 12 A closed back 4 x 12 cabinet with a great bottom end and complex mid range British Blues 2 x 12 A bright sounding open back cabinet with solid lows and highs that maintain their edge even at high gain settings Modern American 4 x 12 A closed back 4 x 12 cabinet that has a full sound The low mids are denser than the British 4 x 12 cabinets Studio 1 x 12 A compact sounding open back cabinet with full mids and shimmering highs British 2 x 12 A mid 1960s open back cabinet with an open smooth tone British 1 x 12 A small open back cabinet with crisp highs and nice low mid transparency Boutique British 2 x 12 A 2x 12 cabinet based on the British 2 x 1
208. per looks in order to anticipate future events You generally won t need to use this feature except when processing signals with extremely sensitive transients If you do raise the Lookahead value you may need to adjust the Attack Time to compensate In contrast to a compressor or expander the Enveloper operates independently of the absolute level of the input signal but this works only if the Threshold slider is set to the lowest possible value Chapter 4 Dynamics Processors 97 98 Expander The Expander is similar in concept to a compressor but increases rather than reduces the dynamic range above the threshold level You can use the Expander to add liveliness and freshness to your audio signals Threshold slider and field Sets the threshold level Signals above this level are expanded e Peak RMS buttons Determine whether the Peak or RMS method is used to analyze the signal e Attack knob and field Determines the time it takes for the Expander to respond to signals that exceed the threshold level Expansion display Shows the expansion curve applied to the signal Release knob and field Sets the time it takes for the Expander to stop processing the signal after it falls below the threshold level Ratio slider and field Sets the expansion ratio the ratio of signal expansion when the threshold is exceeded Note As the Expander is a genuine upward expander in contrast to a downward expander which incr
209. point in the impulse response sample See Using Space Designer s IR Start Parameter Chapter 12 Space Designer Convolution Reverb Spread and Xover knobs synthesized IRs only Spread adjusts the perceived width of the stereo or surround field Xover sets the crossover frequency in Hertz Any synthesized impulse response frequency that falls below this value will be affected by the Spread parameter See Using Space Designer s Spread Parameters Using Space Designer s Input Slider The Input slider behaves differently in stereo or surround instances The Input slider does not appear in mono or mono to stereo instances In stereo instances the Input slider determines how a stereo signal is processed e In surround instances the Input slider determines how much LFE signal is mixed with the surround channels routed into the reverb Stereo Surround sien ii Space Designer Input Slider Stereo Mode Stereo setting top of slider The signal is processed on both channels retaining the stereo balance of the original signal Mono setting middle of slider The signal is processed in mono XStereo setting bottom of slider The signal is inverted with processing for the right channel occurring on the left and vice versa In between positions A mixture of stereo to mono crossfeed signals is produced Space Designer Input Slider Surround Mode Surround Max setting top of slider The maximum amount of LFE signal is mix
210. processed signal Chapter 6 Filter Effects b Use the High Level slider to blend frequencies above the defined frequency band with the processed signal You can modulate the defined frequency band using the Speed CF Modulation and BW Modulation parameters a Speed determines the modulation frequency b CF Center Frequency Modulation defines the intensity of the center frequency modulation c BW Band Width Modulation controls the amount of bandwidth modulation After making your adjustments you can use the Gain slider to adjust the final output level of the processed signal Chapter 6 Filter Effects 159 Imaging Processors Fi The Imaging processors included in Logic Pro are tools for manipulating the stereo image This enables 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 This chapter covers the following Binaural Post Processing p 161 e Direction Mixer p 162 Stereo Spread p 165 Binaural Post Processing Each channel strip in Logic Pro allows you to use a special version of the Pan knob known as the Binaural Panner This is a psychoacoustic processor that can simulate arbitrary sound source positions including up and down information when fed a standard stereo signal The output signal that results from Binaural Panner use is best suited for headphone pl
211. put signals Choose moderate monitoring levels and only use loudspeakers that are actually capable of reproducing the very low frequencies produced Never try to force a loudspeaker to output these frequency bands with an EQ SubBass Parameters The SubBass offers the following parameters e High Ratio knob and field Adjusts the ratio between the generated signal and the original upper band signal e High Center knob and field Sets the center frequency of the upper band Chapter 13 Specialized Effects and Utilities High Bandwidth knob and field Sets the width of the upper band Graphic display Shows the selected upper and lower frequency bands Freq Mix slider and field Adjusts the mix ratio between the upper and lower frequency bands Low Ratio knob and field Adjusts the ratio between the generated signal and the original lower band signal Low Center knob and field Sets the center frequency of the lower band Low Bandwidth knob and field Sets the width of the lower band Dry slider and field Sets the amount of dry non effect original signal Wet slider and field Sets the amount of wet effect signal Using SubBass Unlike a pitch shifter the waveform of the signal generated by SubBass is not based on the waveform of the input signal but is sinusoidal that is it uses a sine wave Given that pure sine waves rarely sit well in complex arrangements you can control the amount of and balance b
212. r LFO parameters The oscillator frequency and output signal can be modulated with an LFO See Modulating the Ringshifter with the LFO Output parameters The output section of the Ringshifter includes a feedback loop and controls to set the stereo width and amount of the dry and wet signals See Controlling the Ringshifter Output Parameters Chapter9 Modulation Effects Setting the Ringshifter Mode The four mode buttons determine whether the Ringshifter operates as a frequency shifter or as a ring modulator Freq Shift Ring Mod M 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 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 The oscillator Frequency control determines the shift direction in the left versus the right channel OSC Ring Modulator button The ring modulator uses the internal sine wave oscillator to modulate the input signal Side Chain Ring Modulator button The ring modulator modulates the amplitude of the input signal with the audio signal assigned via the side chain input The sine wave oscillator is switched off and the Frequency controls are not accessible when Side Chain mode is active Using the Ringshifter
213. r and field Determines the amount of the effect signal that is routed back to the input If you re going for radical flanging effects enter a high Feedback value If simple doubling is what you re after don t use any feedback Negative values invert the phase of the feedback signal resulting in more chaotic effects Flanger Chorus knob and field Sets the basic delay time Set to the far left position to create flanger effects to the center for chorus effects and to the far right to hear clearly discernible delays De Warble button Ensures that the pitch of the modulated signal remains constant Const Mod Constant Modulation button Ensures that the modulation width remains constant regardless of the modulation rate Note When Const Mod is enabled higher modulation frequencies reduce the modulation width Mod Intensity slider and field Sets the modulation amount LFO Mix slider and fields Determines the balance between the two LFOs LFO 1 and LFO 2 Rate knobs and fields The left knob sets the modulation rate for the left stereo channel and the right knob sets the modulation rate for the right stereo channel In surround instances the center channel is assigned the middle value of the left and right LFO Rate knobs The other channels are assigned values between the left and right LFO rates Note The right LFO Rate knob is available only in stereo and surround instances and it can be set separately only if the Left Rig
214. r banks You can use rhythmic material such as a drum loop as an input signal and set up tempo synchronized modulations with different rates for each LFO Feel free to try a tempo synchronized delay effect such as Tape Delay after the EVOC 20 Filterbank to produce unique polyrhythms EVOC 20 Filterbank Output Parameters The Output parameters provide control over the level and stereo width The Output section also incorporates an integrated overdrive distortion circuit Stereo Mode Stereo Width Overdrive button Click to turn 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 Level slider Controls the volume of the EVOC 20 Filterbank output signal Stereo Mode pop up menu Sets the input output mode of the EVOC 20 Filterbank The choices are m s mono input stereo output and s s stereo input stereo output In s s mode the left and right channels are processed by separate filter banks In m s mode a stereo input signal is first summed to mono before being routed to the filter banks Chapter 6 Filter Effects Stereo Width knob Distributes the output signals of the filter bands in the stereo field At the left position the outputs of all bands are centered At the centered position the outputs of all bands ascend from left to right At the right position the bands are output alternately to the left and r
215. r ears RMS instruments not peak reading instruments Therefore using RMS meters makes sense most of the time Alternatively you can use both RMS and Peak meters MultiMeter The MultiMeter provides a collection of professional gauge and analysis tools in a single window It includes An Analyzer to view the level of each 1 3 octave frequency band A Goniometer for judging phase coherency in a stereo sound field A Correlation Meter to spot mono phase compatibility An integrated Level Meter to view the signal level for each channel Chapter 8 Metering Tools 169 170 You can view either the Analyzer or Goniometer results in the main display area You switch the view and set other MultiMeter parameters with the controls on the left side of the interface Main display in Analyzer parameters Analyzer view Level Meter Left Right ERmax Contometer Peak parameters Goniometer parameters Correlation Meter While you can insert the MultiMeter directly into any channel strip it is more commonly used in the master channel strip of the host application when you are working on the overall mix There is also a surround version of the MultiMeter with parameters for each channel and a slightly different layout See Surround MultiMeter Chapter 8 Metering Tools Using the MultiMeter Analyzer In Analyzer mode the MultiMeter s main display shows the frequency spectrum of the input signal as 31 independent frequency
216. r entirely noise are reduced Changes to parameters are instantly reflected here so keep an eye on it Denoiser Smoothing Parameters The Denoiser offers the following smoothing parameters Frequency knob and field Transition knob and field Time knob and field Frequency knob and field Adjusts how smoothing is applied to neighboring frequencies If the Denoiser recognizes that only noise is present on a certain frequency band the higher you set the Frequency parameter the more it changes the neighboring frequency bands to avoid glass noise Time knob and field Sets the time required by the Denoiser to reach or release maximum reduction This is the simplest form of smoothing Transition knob and field 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 parameter the more similar level values are changed in order to avoid glass noise Enhance Timing Enhance Timing is designed to tighten up loose playing of recorded audio in a production It can be used on a variety of materials and works in real time Chapter 13 Specialized Effects and Utilities 251 252 While effective on suitable material this type of real time quantization has some limitations It does not work well on recordings of performances that have been played too far off the beat The same is true for very complex layered drum
217. r mix This curve approximates mirrors the average spectrum of your source audio file You can drag an audio file onto the Template Learn or Current Material Learn buttons for use as either the template or the current material A progress bar appears while the Match EQ is analyzing the file You can also load a previously saved plug in setting or you can import the settings of another unsaved Match EQ instance by copying and pasting When you click either of the Learn buttons the View parameter is set to Automatic and the graphic display shows the frequency curve for the function You can review any of the frequency curves when no file is being processed by choosing one of the other View options The filter curve is updated automatically each time a new template or current material spectrum is learned or loaded when the Match button is enabled You can alternate between the matched and possibly scaled and or manually modified filter curve and a flat response by activating deactivating the Match button Only one of the Learn buttons can be active at a time For example if the Learn button in the Template section is active and you press the Learn button in the Current Material section the analysis of the template file stops the current status is used as the spectral template and analysis of the incoming audio signal Current Material begins Note Each time you match two audio signals either by loading learning a new spectrum while Matc
218. r of the audio signal arriving at its analysis input and transfers it to the synthesizer s sound generators The result of this process is heard at the output of the vocoder The classic vocoder sound uses speech as the analysis signal and a synthesizer sound as the synthesis signal This sound was popularized in the late 1970s and early 1980s You ll probably know it from tracks such as O Superman by Laurie Anderson Funky Town by Lipps Inc and numerous Kraftwerk pieces such as Autobahn Europe Endless The Robots and Computer World In addition to these singing robot sounds vocoding has also been used in many films such as with the Cylons in Battlestar Galactica and most famously with the voice of Darth Vader from the Star Wars saga Chapter 6 Filter Effects 141 142 Vocoding as a process is not strictly limited to vocal performances You could use a drum loop as the analysis signal to shape a string ensemble sound arriving at the synthesis input How Does a Vocoder Work The speech analyzer and synthesizer features of a vocoder are actually two bandpass filter banks Bandpass filters allow a frequency band a slice in the overall frequency spectrum to pass through unchanged and cut the frequencies that fall outside the band s range In the EVOC 20 plug ins these filter banks are named the Analysis and Synthesis sections Each filter bank has a matching number of corresponding band
219. rb effect with a unique feature It allows you to freely adjust the envelope the shape of the diffuse reverb tail Time parameters Sound parameters Mix The interface can be broken down into three areas Time parameters These determine the delay time of the original signal and reverb tail and they change the reverb tail over time The graphic display visually represents the levels over time the envelope of the reverb See EnVerb Time Parameters Sound parameters This area allows you to shape the sound of the reverb signal You can also split the incoming signal into two bands with the Crossover parameter and set the level of the low frequency band separately See EnVerb Sound Parameters Mix parameter Determines the balance between the effect wet and direct dry signals Chapter 11 Reverb Effects 215 216 EnVerb Time Parameters EnVerb offers the following Time parameters Dry Signal Delay slider and field Determines the delay of the original signal You can hear the dry signal only when the Mix parameter is set to a value other than 100 Predelay knob and field Sets the time between the original signal and the starting point of the reverb attack phase the very beginning of the first reflection Attack knob and field Defines the time it takes for the reverb to climb to its peak level Decay knob and field Defines the time it takes for the level of the reverb to drop from its peak to the sustain
220. re that these overdriven signals continue to sound good Freeze button Captures the current delay repeats and sustains them until the Freeze button is turned off Delay field Sets the current delay time in milliseconds this parameter is dimmed when you synchronize the delay time to the project tempo Sync button Synchronizes delay repeats to the project tempo including tempo changes Tempo field Sets the current delay time in beats per minute this parameter is dimmed when you synchronize the delay time to the project tempo Chapter 2 Delay Effects 75 76 Groove slider and field Determines the proximity of every second delay repeat to the absolute grid position in other words how close every second delay repeat is A Groove setting of 50 means that every delay has the same delay time Settings below 50 result in every second delay being played earlier in time Settings above 50 result in every second delay being played later in time When you want to create dotted note values move the Groove slider all the way to the right to 75 For triplets select the 33 33 setting Note buttons Set the grid resolution for the delay time These are shown as note durations Low Cutand High Cut sliders and fields Frequencies below the Low Cut value and above the High Cut value are filtered out of the source signal You can shape the sound of the echoes with the highpass and lowpass filters The filters are located in the feedb
221. reely use the Stereo Delay on mono tracks or busses when you want to create independent delays for the two stereo sides Chapter 2 Delay Effects 74 Note If you use the effect on mono channel strips the track or bus will have two channels from the point of insertion all Insert slots after the chosen slot will be stereo As the parameters for the left and right delays are identical the descriptions below only cover the left channel the right channel information is provided in brackets if named differently Parameters that are common to both channels are shown separately Channel Parameters Left Right Input pop up menu Choose the input signal for the two stereo sides Options include OFF Left Right L R L R Left Right Delay field Sets the current delay time in milliseconds this parameter is dimmed when you synchronize the delay time to the project tempo Groove slider and field Determines the proximity of every second delay repeat to the absolute grid position in other words how close every second delay repeat is Note buttons Set the grid resolution for the delay time These are shown as note durations these are dimmed when the delay time is not synchronized with the project tempo Left Right Feedback knob and field Set the amount of feedback for the left and right delay signals Crossfeed Left to Right Crossfeed Right to Left knob and field Transfer the feedback signal of the left channel to the
222. rent sections of the Tap display Horizontally drag the middle of the bright rectangle in the Overview display The zoomed view in the Tap display updates as you drag Creating Taps in Delay Designer You can create new delay taps in three different ways by using the Tap pads by creating them in the Identification bar or by copying existing taps To create taps with the Tap pad Click the upper pad Start Note Whenever you click the Start pad it automatically erases all existing taps Given this behavior after you have created your initial taps you will want to create subsequent taps by clicking in the Identification bar The upper pad label changes to Tap and a red tap recording bar appears in the strip below the view buttons cry Tap Last Tap Click the Tap button to begin recording new taps Click the Tap button to create new taps These are created at the exact moments in time of each click adopting the rhythm of your click pattern To finish creating taps click the Last Tap button This adds the final tap ending tap recording and assigning the last tap as the feedback tap for an explanation of the feedback tap see Using Delay Designer s Master Section Note If you do not click the Last Tap button tap recording automatically stops after 10 seconds or when the 26th tap is created whichever comes first Chapter 2 Delay Effects To create taps in the Identification bar Click at the appropriate po
223. requency bands The upper section represents the left channel and the lower section represents the right channel The frequency scale displays frequencies in ascending order from left to right Upper and Lower Freq uency slider and fields Determine the highest and lowest frequencies that will be redistributed in the stereo image Order knob and field Determines the number of frequency bands that the signal is divided into A value of 8 is usually sufficient for most tasks but you can use up to 12 bands Chapter 7 Imaging Processors Metering Tools 8 You can use the Metering tools to analyze audio in a variety of ways These plug ins offer different facilities to the meters shown in channel strips They have no effect on the audio signal and are designed for use as diagnostic aids Each meter is specifically designed to view different characteristics of an audio signal making each suitable for particular studio situations As examples the BPM Counter displays the tempo the Correlation Meter displays the phase relationship and the Level Meter displays the level of an incoming audio signal This chapter covers the following BPM Counter p 167 e Correlation Meter p 168 e Level Meter Plug in p 168 e MultiMeter p 169 e Surround MultiMeter p 174 Tuner p 180 BPM Counter The BPM Counter is used to analyze the tempo of incoming audio in beats per minute bpm The detection circuit looks for any transients
224. requency range that is swept Order switches between different algorithms with higher even numbers resulting in a heavier phasing effect Odd order numbers result in more subtle comb filtering effects Feedback determines the amount of the effect signal that is routed back into the input This can change the tonal color can make the phasing effect more pronounced or can do both Tone works from the center position turn it to the left to increase the amount of lowpass filtering or turn it to the right to increase the amount of highpass filtering Mix sets the level ratio between each phaser Retro Chorus A subtle vintage chorus effect Rate sets the modulation speed and can run freely or be synchronized with the host application tempo by enabling the Sync button When synchronized you can specify bar beat and note values including triplets and dotted notes Depth sets the strength of the effect Robo Flanger Flexible flanging effect Rate sets the modulation speed and can run freely or be synchronized with the host application tempo by enabling the Sync button When synchronized you can specify bar beat and note values including triplets and dotted notes Depth sets the strength of the effect Feedback determines the amount of the effect signal that is routed back into the input This can change the tonal color can make the flanging effect more pronounced or can do both The Manual knob sets a delay time between the source a
225. reverb tail High Diffusion values represent a regular density with few alterations in level times and panorama position over the course of the diffuse reverb signal Low Diffusion values result in the reflection density becoming irregular and grainy This also affects the stereo spectrum As with Density find the best balance for the signal Chapter 11 Reverb Effects PlatinumVerb The PlatinumVerb allows you to edit both the early reflections and diffuse reverb tail separately making it easy 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 Early Reflections parameters Balance ER Reverb slider Output parameters Reverb parameters The interface is broken down into four parameter areas Early Reflections parameters Emulates the original signal s first reflections as they bounce off the walls ceiling and floor of a natural room See PlatinumVerb Early Reflections Parameters e Reverb parameters Controls the diffuse reverberations See PlatinumVerb Reverb Parameters Output parameters Determines the balance between the effected wet and direct dry signals See PlatinumVerb Output Parameters Balance ER Reverb slider Controls the balance between the Early Reflections and Reverb sections When you set the slider to either of its extreme positions the unused section is deactivated Chapter 11 Reverb Effect
226. right corner of the Amp Designer interface It serves as the final level control for Amp Designer and can be thought of as a behind the speaker volume control that sets the level of the output that is fed to the ensuing Insert slots in the channel strip or directly to the channel strip output Note This parameter is different from the Master control which serves the dual purpose of sound design as well as controlling the level of the Amp section Chapter 1 Amps and Pedals 27 28 Bass Amp Bass Amp simulates the sound of several famous bass amplifiers You can route bass guitar and other signals directly through the Bass Amp reproducing the sound of your musical part played through a number of high quality bass guitar amplification systems Model Mid Freq Treble Leeann Output Level Bass Amp offers the following parameters Model pop up menu Includes the following amplifier models American Basic 1970s era American bass amp equipped with eight 10 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 American Scoop Based on the American Basic amp but combines the frequency characteristics of the American Deep and American Bright with both low mid from 500 Hz and upper mid from 4 5 kHz frequencies emphasized Well suited for funk and fusion recordin
227. right channel and vice versa Feedback Phase button Use to invert the phase of the corresponding channel s feedback signal Crossfeed Phase button Use to invert the phase of the crossfed feedback signals Chapter 2 Delay Effects Common Parameters Beat Sync button Synchronizes delay repeats to the project tempo including tempo changes Output Mix Left and Right sliders and fields Independently control the left and right channel signals Low Cut and High Cut sliders and fields Frequencies below the Low Cut value and above the High Cut value are filtered out of the source signal Tape Delay Tape Delay simulates the warm sound of vintage tape echo machines with the convenience of easy delay time synchronization to your project tempo The effect is equipped with a highpass and lowpass filter in the feedback loop simplifying the creation of authentic dub echo effects Tape Delay also includes an LFO for delay time modulation which can be used to produce pleasant or unusual chorus effects even on long delays Feedback slider Determines the amount of delayed and filtered signal that is routed back to the input of the Tape Delay Set the Feedback slider to the lowest possible value to generate a single echo Turn Feedback all the way up to endlessly repeat the signal The levels of the original signal and its taps echo repeats tend to accumulate and may cause distortion You can use the internal tape saturation circuit to ensu
228. rmine phase differences between the left and right channels Phase problems are easily spotted as trace cancellations along the center line M mid mono The idea of the goniometer was born with the advent of early two channel oscilloscopes To use such devices as goniometers users would connect the left and the right stereo channels to the X and Y inputs while rotating the display by 45 to produce a useful visualization of the signal s stereo phase The signal trace slowly fades to black imitating the retro glow of the tubes found in older goniometers while also enhancing the readability of the display Analyzer Left Right Rmax Goniometer button Switches the main display to Goniometer mode Auto Gain field Sets the amount of display compensation for low input levels You can set Auto Gain levels in 10 increments or set it to off Note To avoid confusion with the Auto Gain parameter found in other effects and processors such as the compressors Auto Gain is only used as a display parameter in the meters It increases display levels to enhance readability It does not change the actual audio levels Decay field Determines the time it takes for the Goniometer trace to fade to black Chapter 8 Metering Tools Using the MultiMeter s Level Meter The Level Meter displays the current signal level on a logarithmic decibel scale The signal level for each channel is represented by a blue bar RMS and Peak le
229. rmines the distance of the virtual microphones the listening position from the emulated speaker cabinet Use higher values to make the sound darker and less defined This is typical of microphones when positioned further from the sound source Mic Angle slider Use to define the stereo image by changing the angle of the simulated microphones An angle of 0 results in a mono sound An angle of 180 causes phase cancellations Scanner Vibrato Effect Scanner Vibrato simulates the scanner vibrato section of a Hammond organ The Scanner Vibrato is based on an analog delay line consisting of several lowpass filters The delay line is scanned by a multipole capacitor which has a rotating pickup It is a unique effect that cannot be simulated with simple LFOs Chapter9 Modulation Effects 199 200 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 Vibrato knob Use to choose from three Vibrato positions V1 V2 and V3 or three Chorus positions C1 C2 and C3 In the Vibrato positions only the delay line signal is heard each with different intensities The three Chorus positions C1 C2 and C3 mix the signal
230. ro the automatic Wah Wah function is disabled Attack knob Sets the time it takes for the Wah Wah filter to fully open e Release knob Sets the time it takes for the Wah Wah filter to close Chapter 6 Filter Effects 155 156 A w N Fuzz Parameters These parameters control the integrated distortion and compression circuits The compressor always precedes the Fuzz effect Comp Compression Ratio knob Sets the compression ratio Fuzz Gain knob Sets the level of the Fuzz or distortion effect Fuzz Tone knob Adjusts the tonal color of the fuzz effect Low settings tend to be warmer and high settings are brighter and harsher Using the Fuzz Wah The following section provides practical tips for the Fuzz Wah parameters Setting the Wah Wah Level with Auto Gain The Wah 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 Switch Auto Gain to on Raise the effect level to a value just below the mixer s clipping limit Make a sweep with a high relative Q setting Switch Auto Gain to off and repeat the sweep Important Make sure to set a conservative master output level for your host application before trying this Failure to do so may result in damage to your hearing or speakers Setting the Pedal Range Common MIDI foot pedals have a much larger mechanical range
231. s Trem o Tone A tremolo effect modulation of the signal level Rate sets the modulation speed and can run freely or be synchronized with the host application tempo by enabling the Sync button When synchronized you can specify bar beat and note values including triplets and dotted notes Depth sets the strength of the effect Level sets the post tremolo gain 46 Chapter 1 Amps and Pedals Stompbox Description the Vibe A vibrato chorus effect based on the Scanner Vibrato unit found in the Hammond B3 organ You can choose from three vibrato V1 3 or chorus C1 3 variations with the Type knob Rate sets the modulation speed and can run freely or be synchronized with the host application tempo by enabling the Sync button When synchronized you can specify bar beat and note values including triplets and dotted notes Depth sets the strength of the effect See Scanner Vibrato Effect for background information on this effect Delay Pedals This section describes the Delay effects pedals Stompbox Description Blue Echo A delay effect Time sets the modulation speed and can run freely or be synchronized with the host application tempo by enabling the Sync button When synchronized you can specify bar beat and note values including triplets and dotted notes The Repeats knob determines the number of delay repeats Mix sets the balance between the delayed and source signals The Tone Cut switch controls a fixed frequency
232. s Dry slider Sets the overall level of the non effect signal for all channels Set this to a value of 0 mute when using Space Designer as a bus effect in an aux channel strip Use the Send knob of each bussed channel strip to control the wet dry balance Working with Pre Dly Predelay in Space Designer Predelay is the amount of time that elapses between the original signal and the initial early reflections of the reverberation For a room of any given size and shape predelay determines the distance between the listener and the walls ceiling and floor Space Designer allows you to adjust this parameter separately from predelay and over a greater range than what would be considered natural for predelay In practice an extremely short predelay tends to make it difficult to pinpoint the position of the signal source It can also color the sound of the original signal On the other hand an excessively long predelay can be perceived as an unnatural echo It can also divorce the original signal from its early reflections leaving an audible gap between the original and reverb signals The ideal predelay setting for different sounds depends on the properties of or more accurately the envelope of the original signal Percussive signals generally require shorter predelays than signals where the attack fades in gradually such as strings A good rule of thumb is to use the longest predelay possible before undesirable side effects such as
233. s if the analysis filter bank has five bands 1 2 3 4 and 5 there will be a corresponding set of five bands in the synthesis filter bank Band 1 in the analysis bank is matched to Band 1 in the synthesis bank Band 2 to Band 2 and so on The audio signal arriving at the analysis input passes through the analysis filter bank where it is divided into bands An envelope follower is coupled to each filter band The envelope follower of each band tracks or follows any volume changes in the audio source or more specifically the portion of the audio that has been 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 voltage controlled amplifiers VCAs in analog vocoders This allows any volume changes to the bands in the analysis filter bank to be imposed on the matching bands in the synthesis filter bank These filter changes are heard as a synthetic reproduction of the original input signal or a mix of the two filter bank signals The more bands a vocoder offers the more precisely the original sound s character will be remodeled The EVOC plug ins offer up to 20 bands per bank To ensure their musical usefulness you have full control over the output level of each bandpass filter facilitating uniqu
234. s If the British Blues Combo is too clean for your needs combine it with Pedalboard s Hi Drive stompbox for an aggressive blues tone or the Candy Fuzz stompbox for an explosive rock tone See Distortion Pedals and Pedalboard British Combos The British Combos capture the brash treble rich sound that will forever be associated with 1960s British rock and pop The sonic signature of these amps is characterized by their high end response yet they are rarely harsh sounding due to a sweet distortion and smooth natural compression Model Description British Combo A 2x 12 combo based on the early 1960s amps that powered the British Invasion Perfect for chiming chords and stabbing solos Chapter 1 Amps and Pedals 15 16 Model Description Small British Combo A 1x 12 combo with half the power of the British Combo this amp offers a slightly darker less open tone Boutique British Combo A 2x 12 combo that is a modern take on the original 1960s sound The tone is thicker with stronger lows and milder highs than the other British Combos Tip Using high Treble and Presence knob settings that might become strident on other amp types can sound great with the British Combos British Alternatives The late 1960s amplifier heads and combos that inspired the Sunshine models are loud and aggressive with full bodied mid frequencies These amps are not just for single note solos and power chords as they can sound great with big
235. s Mention of third party products is for informational purposes only and constitutes neither an endorsement nor a recommendation Apple assumes no responsibility with regard to the performance or use of these products Preface Chapter 1 Chapter 2 Chapter 3 Chapter 4 o oNN 11 28 29 35 51 51 72 72 73 75 77 78 79 80 81 81 82 85 86 87 88 92 94 96 98 Contents An Introduction to the Logic Pro Effects About the Logic Pro Effects About the Logic Pro Documentation Additional Resources Amps and Pedals Amp Designer Bass Amp Guitar Amp Pro Pedalboard Delay Effects Delay Designer Echo Sample Delay Stereo Delay Tape Delay Distortion Effects Bitcrusher Clip Distortion Distortion Effect Distortion Il Overdrive Phase Distortion Dynamics Processors Types of Dynamics Processors Adaptive Limiter Compressor DeEsser Ducker Enveloper Expander Chapter 5 Chapter 6 Chapter 7 Chapter 8 Chapter 9 100 103 106 107 107 111 112 115 116 117 121 127 129 131 131 137 141 153 157 161 161 162 165 167 167 168 168 169 174 180 183 184 184 186 187 187 190 191 Limiter Multipressor Noise Gate Silver Compressor Silver Gate Surround Compressor Equalizers Channel EQ DJ EQ Fat EQ Linear Phase EQ Match EQ Single Band EQs Silver EQ Filter Effects AutoFilter EVOC 20 Filterbank EVOC 20 TrackOscillator Fuzz Wah
236. s These room characteristic sample recordings are known as impulse responses Convolution reverbs work by convolving combining an audio signal with the impulse response recording of a room s reverb characteristics See Space Designer Convolution Reverb AVerb AVerb is a simple reverb effect that employs a single parameter Density Time to control both the early reflections and diffuse reverb tail It is a quick and easy tool for creating a range of interesting space and echo effects The AVerb may not be the best choice for simulating real acoustic environments however Predelay slider and field Determines the time between the original signal and the early reflections of the reverb signal Reflectivity knob and field Defines how reflective the imaginary walls ceiling and floor are in other words how hard the walls are and what they re made of Glass stone timber carpet and other materials have a dramatic impact on the tone of the reverb Room Size knob and field Defines the dimensions of simulated rooms Chapter 11 Reverb Effects Density Time slider and field Determines both the density and duration of the reverb Low values tend to generate clearly discernible early reflection clusters generating something similar to an echo High values result in a more reverb like effect e Mix slider and field Sets the balance between the effect wet and direct dry signals EnVerb EnVerb is a versatile reve
237. s 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 Blues singers glide between these notes creating an uncertainty or tension between the scales Use of the bypass buttons allows you to exclude particular keys from changes leaving them as they were If you enable the Bypass All button the input signal is passed through unprocessed and uncorrected This is useful for spot corrections to pitch through use of automation Bypass All is optimized for seamless bypass enabling or disabling in all situations Tip You ll often find that it s best to correct only the notes with the most harmonic gravity For example choose sus4 from the Scale pop up menu and set the Root note to match the project key This will limit correction to the root note the fourth and the fifth of the key scale Activate the bypass buttons for all other notes and only the most important and sensitive notes will be corrected while all other singing remains untouched Setting the Pitch Correction Reference Tuning Choose File gt Project Settings gt Tuning to determine the tuning reference for all software instruments If you turn on the Use Globa
238. s 221 222 PlatinumVerb Early Reflections Parameters The PlatinumVerb offers the following Early Reflections parameters Predelay slider and field Determines the amount of time between the start of the original signal and the arrival of the early reflections Extremely short Predelay settings can color the sound and make it difficult to pinpoint the position of the signal source Overly long Predelay settings can be perceived as an unnatural echo and can divorce the original signal from its early reflections leaving an audible gap between them The optimum Predelay setting depends on the type of input signal or more precisely the envelope of the input signal Percussive signals generally require shorter predelays than signals where the attack fades in gradually A good working method is to use the longest possible Predelay value before you start to hear undesirable side effects such as an audible echo When you reach this point reduce the Predelay setting slightly Room Shape slider and field 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 slider and field Determines the dimensions of the room The numeric value indicates the length of the room s walls the distance between two corners Stereo Base slider and field Defines the distance between the two virtual microphones that are used to cap
239. s EQ shapes the sound of incoming audio by changing the level of specific frequency bands Equalization is one of the most commonly used audio processes both for music projects and in post production work for video You can use EQ to subtly or significantly shape the sound of an audio file instrument or project by adjusting specific frequencies or frequency ranges All EQs are specialized filters that allow certain frequencies to pass through unchanged while raising boosting or lowering cutting the level of other frequencies Some EQs can be used in a broad brush fashion to boost or cut a large range of frequencies Other EQs particularly parametric and multiband EQs can be used for more precise control The simplest types of EQs are single band EQs which include low cut and high cut lowpass and highpass shelving and parametric EQs Multiband EQs such as the Channel EQ Fat EQ or Linear Phase EQ combine several filters in one unit enabling you to control a large part of the frequency spectrum Multiband EQs allow you to independently set the frequency bandwidth and Q factor of each frequency spectrum band This provides extensive and precise tone shaping on any audio source be it an individual audio signal or an overall mix Logic Pro includes a variety of single band and multiband EQs This chapter covers the following e Channel EQ p 112 e DJEQ p 115 e Fat EQ p 116 e Linear Phase EQ p 117 e Mat
240. s Oscillator In both frequency shifter modes and the ring modulator OSC mode the internal sine wave oscillator is used to modulate the amplitude of the input signal In the frequency shifter modes the Frequency parameter controls the amount of frequency shifting up and or down applied to the input signal Chapter9 Modulation Effects 193 194 In the ring modulator OSC mode the Frequency parameter controls the frequency content timbre of the resulting effect This timbre can range from subtle tremolo effects to clangorous metallic sounds Freq Shift Ring Mod Frequency control Sets the frequency of the sine oscillator Lin ear and Exp onential buttons Switch the scaling of the Frequency control e Exp onential Exponential scaling offers extremely small increments around the 0 point which is useful for programming slow moving phasing and tremolo effects Lin ear Linear scaling resolution is even across the entire control range Env Follow slider and field Determines the impact of incoming signal levels on the oscillator modulation depth LFO slider and field Determines the amount of oscillator modulation by the LFO Using the Ringshifter s Delay The effect signal is routed through a delay following the oscillator Feedback Stereo Wi Time knob and field Sets the delay time This is in Hz when running freely or in note values including triplet and dotted notes when the Sync button is active
241. s brighter and more cutting than the Condenser model At the same time the lower mid frequency range is less pronounced making this model more suitable for miking rock guitar tones Tip Combining both microphone types can sound quite interesting Duplicate the guitar track and insert Guitar Amp Pro as an insert effect on both tracks Select different microphone types in each Guitar Amp Pro instance while retaining identical settings for all other parameters and mix the track signal levels You can of course choose to vary any other parameters Setting the Guitar Amp Pro Output Level The Output slider is found at the bottom below the Effects section It serves as the final level control for Guitar Amp Pro and can be thought of as a behind the speaker volume control that is used to set the level fed to the ensuing plug in slots on the channel strip or to Output channel strips Note This parameter is different from the Master control which serves the dual purpose of sound design as well as controlling the level of the Amp section Pedalboard The Pedalboard simulates the sound of a number of well loved and famous stompbox pedal effects You can process any audio signal with a combination of stompboxes You can add remove and reorder pedals The signal flow runs from left to right in the Pedal area The addition of two discrete busses coupled with splitter and mixer units enables you to experiment with sound design and pr
242. s can be the entire project or individual channel strips within it is analyzed These two spectra are then matched creating a filter curve This filter curve adapts the frequency response of the current material to match that of the template Before applying the filter curve you can modify it by boosting or cutting any number of frequencies or by inverting the curve The Analyzer allows you to visually compare the frequency spectrum of the source file and the resulting curve making it easier to make manual corrections at specific points within the spectrum You can use the Match EQ in different ways depending on your intended outcome and the audio you re working with In general you will want to make your mix sound similar to an existing recording either your own or that of another artist Note Although the Match EQ acoustically matches the frequency curve of two audio signals it does not match any dynamic differences between the two signals Chapter5 Equalizers 121 122 Match EQ Parameters The Match EQ offers the following parameters Match EQ Analyzer Parameters Analyzer button Turns the Analyzer function on or off Pre Post button Determines whether the Analyzer looks at the signal before Pre or after Post the filter curve is applied View pop up menu Sets the information shown on the graphic display Choices are Automatic Displays information for the current function as determined by the active button
243. s shown as a percentage value rather than in degrees The value 100 or 100 indicates the greatest possible distance between the modulation phases of all voices e Spread slider and field Distributes voices across the stereo or surround field Set a value of 200 to artificially expand the stereo or surround base Note that monaural compatibility may suffer if you choose to do this e Mix slider and field Determines the balance between dry and wet signals Effect Volume knob and field Determines the level of the effects signal This is a useful tool that compensates for changes in volume caused by changes to the Voices parameter Chapter9 Modulation Effects 185 186 Note When you are using the Ensemble effect in surround the input signal is converted to mono before processing In other words you insert the Ensemble effect as a multi mono instance Flanger Effect The Flanger effect works in much the same way as the Chorus effect but it uses a significantly shorter delay time In addition the effect signal can be fed back into the input of the delay line Flanging is typically used to create changes that are described as adding a spacey or underwater quality to input signals Feedback slider and field Determines the amount of the effect signal that is routed back into the input This can change the tonal color and or make the sweeping effect more pronounced Negative Feedback values invert the phase of the routed signal
244. s the Analyzer on or off Pre Post EQ button Determines whether the Analyzer shows the frequency curve before or after EQ is applied when Analyzer mode is active Resolution pop up menu Sets the sample resolution for the Analyzer with the following menu items low 1024 points medium 2048 points and high 4096 points Linear Phase EQ Graphic Display Section Band On Off buttons Click to turn the corresponding band on or off Each button icon indicates the filter type Band 1 is a highpass filter Band 2 is a low shelving filter Bands 3 through 6 are parametric bell filters Band 7 is a high shelving filter Band 8 is a lowpass filter Chapter 5 Equalizers Graphic display Shows the current curve of each EQ band Drag horizontally in the section of the display that encompasses each band to adjust the frequency of the band Drag vertically in the section of the display that encompasses each band to adjust the gain of each band except bands 1 and 8 The display reflects your changes immediately Drag the pivot point in each band to adjust the Q factor Q is shown beside the cursor when the mouse is moved over a pivot point Linear Phase EQ Parameter Section Frequency fields Adjust the frequency of each band Gain Slope fields Set the amount of gain for each band For bands 1 and 8 this changes the slope of the filter Q fields Adjust the Q or resonance for each band the range of frequencies ar
245. scale with the C above middle C displayed as C4 and middle C displayed as C3 Tuning Adjustment slider and field Sets the pitch of the note used as the basis for tuning By default the Tuner is set to the project s Tuning parameter value Drag the knob to the left to lower the pitch corresponding to A Drag the knob to the right to raise the pitch corresponding to A The current value is displayed in the field To use the Tuner Insert the Tuner into an audio channel strip Play a single note on the instrument and watch the display If the note is flat or sharp of the Keynote the segments to the left or right of center are illuminated indicating how far in cents the note is off pitch Adjust the tuning of your instrument until the indicator is centered in the graphic tuning display Chapter 8 Metering Tools 181 Modulation Effects 9 Modulation effects are used to add motion and depth to your sound Effects such as chorus flanging and phasing are well known examples Modulation effects typically delay the incoming signal by a few milliseconds and use an LFO to modulate the delayed signal The LFO may also be used to modulate the delay time in some effects A low frequency oscillator LFO is much like the sound generating oscillators in synthesizers but the frequencies generated by an LFO are so low that they can t be heard Therefore they are used only for modulation purposes LFO parameters include speed or frequenc
246. show the effective signal average and provide a representative overview of perceived volume levels The Peak mode shows level peaks accurately Scale shown in the main display Indicates the scale of levels Adjusting the scale is useful when analyzing highly compressed material as it makes it easier to identify small level differences Drag vertically on the scale to adjust it Using the Surround MultiMeter Goniometer A goniometer helps you to judge the coherence of the stereo image and determine phase differences between the left and right channels Phase problems are easily spotted as trace cancellations along the center line M mid mono The idea of the goniometer was born with the advent of early two channel oscilloscopes To use such devices as goniometers users would connect the left and the right stereo channels to the X and Y inputs while rotating the display by 45 to produce a useful visualization of the signal s stereo phase The signal trace slowly fades to black imitating the retro glow of the tubes found in older goniometers while also enhancing the readability of the display Chapter 8 Metering Tools Because the Surround MultiMeter Goniometer is dealing with multichannel signals the display is divided into multiple segments as shown in the image Each segment indicates a speaker position When the surround panner is moved in a channel strip the indicator changes accordingly This indicates not on
247. sition To copy taps in the Identification bar Option drag a selection of one or more taps to the appropriate position The delay time of copied taps is set to the drag position Delay Designer Tap Creation Suggestions The fastest way to create multiple taps is to use the Tap pads If you have a specific rhythm in mind you might find it easier to tap out your rhythm on dedicated hardware controller buttons instead of using mouse clicks If you have a MIDI controller you can assign the Tap pads to buttons on your device For information about assigning controllers see the Control Surfaces Support manual Note Whenever you click the Start Tap pad it automatically erases all existing taps Given this behavior after you create your initial taps you will want to create subsequent taps by clicking in the Identification bar After a tap has been created you can freely adjust its position or you can remove it if it was created accidentally For details see Moving and Deleting Taps in Delay Designer Identifying Taps in Delay Designer Taps are assigned letters based on their order of creation The first tap to be created is assigned as Tap A the second tap is assigned as Tap B and so on Once assigned each tap is always identified by the same letter even when moved in time and therefore reordered For example if you initially create three taps they will be named Tap A Tap B and Tap C If you then change the delay time of Tap B
248. so that it precedes Tap A it will still be called Tap B Chapter 2 Delay Effects 57 58 The Identification bar shows the letter of each visible tap The Tap Delay field of the Tap parameter bar displays the letter of the currently selected tap or the letter of the tap being edited when multiple taps are selected for details see Selecting Taps in Delay Designer Selecting Taps in Delay Designer There will always be at least one selected tap You can easily distinguish selected taps by color the toggle bar icons and the Identification bar letters of selected taps are white ey Tap SE To select a tap Do one of the following Click a tap in the Tap display Click the appropriate tap letter in the Identification bar Click one of the arrows to the left of the Tap name to select the next or previous tap Chapter 2 Delay Effects Open the pop up menu to the right of the Tap name and choose the appropriate tap letter 5750 0ms Delay To select multiple taps Do one of the following Drag across the background of the Tap display to select multiple taps Shift click specific taps in the Tap display to select multiple nonadjacent taps Moving and Deleting Taps in Delay Designer You can move a tap backward or forward in time or completely remove it Note When you move a tap you are actually editing its delay time To move a selected tap in time Select the tap in the Identification bar and dr
249. sound of early digital audio devices to create artificial aliasing by dividing the sample rate or to distort signals until they are unrecognizable Drive slider and field Sets the amount of gain in decibels applied to the input signal Note Raising the Drive level tends to increase the amount of clipping at the output of the Bitcrusher as well Resolution slider and field Sets the bit rate between 1 and 24 bits This alters the calculation precision of the process Lowering the value increases the number of sampling errors generating more distortion At extremely low bit rates the amount of distortion can be greater than the level of the usable signal e Waveform display Shows the impact of parameters on the distortion process Downsampling slider and field Reduces the sample rate A value of 1 x leaves the signal unchanged a value of 2 x halves the sample rate and a value of 10 x reduces the sample rate to one tenth of the original signal For example if you set Downsampling to 10x a 44 1 kHz signal is sampled at just 4 41 kHz Note Downsampling has no impact on the playback speed or pitch of the signal Mode buttons Set the distortion mode to Folded Cut or Displaced Signal peaks that exceed the clip level are processed Note The Clip Level parameter has a significant impact on the behavior of all three modes This is reflected in the Waveform display so try each mode button and adjust the Clip Level slid
250. speakers Don t be surprised if you get a deeper richer tone from a 10 speaker than from a large 4 x 12 cabinet Try several sizes and choose the one that works best for your music Single Speakers or Multiple Speakers Guitarists sometimes use cabinets with multiple speakers and not only for the larger sound they tend to provide Phase cancellations occur between the speakers adding texture and interest to the tone Much of the classic rock sound for example has to do with the tonal peaks and dips caused by this interaction between the speakers in a 4x 12 cabinet Chapter 1 Amps and Pedals 19 Amp Designer Cabinet Reference Table You can choose a cabinet model from the Cabinet pop up menu on the black bar at the bottom of Amp Designer s interface The table below covers the properties of each cabinet model available in Amp Designer Cabinet Description Tweed 1 x 12 A 12 open back cabinet from the 1950s with a warm and smooth tone Tweed 4 x 10 A 4x 10 open back cabinet that was originally conceived for bassists but guitarists love its sparkling presence An authentic late 1950s sound Tweed 1 x 10 A single 10 open back combo amp cabinet from the 1950s with a smooth sound Blackface 4 x 10 Classic open back cabinet with four 10 speakers Its tone is deeper and darker than the Tweed 4 x 10 Silverface 2 x 12 An open back model from the 1960s that provides great low end punch Blackface 1 x 10 An o
251. sponsive U V detection is When this knob is turned 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 sound source determined by the Mode menu as described in Mode menu below being used on the majority of the input signal including voiced signals Sonically this results in a sound that resembles a radio signal that is breaking up and contains a lot of static or noise Mode menu Sets the sound sources that can be used to replace the unvoiced content of the input signal You can choose between the following e Noise Uses noise alone for the unvoiced portions of the sound e Noise Synth Uses noise and the synthesizer for the unvoiced portions of the sound Blend Uses the analysis signal after it has passed through a highpass filter for the unvoiced portions of the sound The Sensitivity parameter has no effect when this setting is used Level knob Controls the volume of the signal used to replace the unvoiced content of the input signal Chapter 6 Filter Effects Important Take care with the Level knob particularly when a high Sensitivity value is used to avoid internally overloading the EVOC 20 TrackOscillator EVOC 20 TrackOscillator Synthesis In Parameters The Synthesis In section controls various aspects of the tracking signal for the synthesizer The tracking signal is used to
252. ssing for example when you have inserted an 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 slider and field Sets the amount of gain Phase Invert Left and Right buttons Invert the phase of the left and right channels respectively Balance knob and field Adjusts the balance of the incoming signal between the left and right channels Swap L R Left Right button Swaps the left and right output channels The swapping occurs after the Balance parameter in the signal path Mono button Outputs the summed mono signal on both the left and right channels Note The Gain plug in is available in mono mono to stereo and stereo instances In mono and mono to stereo modes only one Phase Invert button is available In the mono version the Stereo Balance Swap Left Right and Mono parameters are disabled Chapter 14 Utilities and Tools Using Phase Inversion Inverting phase is useful for dealing with time alignment problems particularly those caused by simultaneous recording with multiple microphones When you invert the phase of a signal heard in isolation it sounds identical to the original When the signal is heard in conjunction with other signals however phase inversion may have an audible effect For example if you place microphones above and below a snare drum you may find that inverting the phase of either microphone can improve or ruin the
253. t and s s stereo input to stereo output Note Set Stereo Mode to m s if the input signal is mono and to s s if the input signal is stereo In s s mode the left and right stereo channels are processed by separate filter banks When you use m s mode on a stereo input signal the signal is first summed to mono before it is passed to the filter banks Stereo Width knob Distributes the output signals of the Synthesis section s filter bands in the stereo field e At the left position the outputs of all bands are centered At the centered position the outputs of all bands ascend from left to right At the right position the bands are output alternately to the left and right channels Fuzz Wah The Fuzz Wah plug in emulates classic wah wah effects often used with a clavinet and it adds compression and fuzz distortion effects as well The name wah wah comes from the sound it produces It has been a popular effect usually a pedal effect with electric guitarists since the days of Jimi Hendrix The pedal controls the cutoff frequency of a bandpass lowpass or less commonly highpass filter Getting to Know the Fuzz Wah Interface The Fuzz Wah interface is broken down into the following sections Effect Order buttons Fuzz parameters Wah parameters Auto Wah parameters Effect Order buttons Select whether the wah wah effect precedes the fuzz effect in the signal chain Wah Fuzz or vice versa Fuzz Wah See Ef
254. t possible Predelay value before you start to hear undesirable side effects such as an audible echo When you reach this point reduce the Predelay setting slightly Room Shape slider and field 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 slider and field Determines the dimensions of the room The numeric value indicates the length of the room s walls the distance between two corners Stereo Base slider and field Defines the distance between the two virtual microphones that are used to capture the signal in the simulated room Note Spacing the microphones slightly farther apart than the distance between two human ears generally delivers the best and most realistic results This parameter is available only in stereo instances of the effect Chapter 11 Reverb Effects 219 220 GoldVerb Reverb Parameters The GoldVerb offers the following Reverb parameters Initial Delay slider and field Sets the time between the original signal and the diffuse reverb tail 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 parameter so that it is as long as possible without a noticeable gap between the early reflections and the reverb tail Spread slider and field
255. te them For example when you are recording a drum kit and using the Noise Gate to isolate the sound of the snare drum the hi hat may also open the gate in many cases To remedy this use the side chain controls to isolate the desired trigger signal with the High Cut and Low Cut filters Important The side chain signal is used only as a detector trigger in this situation The filters are used to isolate particular trigger signals in the side chain source but they have no influence on the actual gated signal the audio being routed through the Noise Gate To use the side chain filters Click the Monitor button to hear how the High Cut and Low Cut filters will affect the incoming trigger signal Drag the High Cut slider to set the upper frequency Trigger signals above this are filtered Drag the Low Cut slider to set the lower frequency Trigger signals below this are filtered Chapter 4 Dynamics Processors 106 The filters allow only very high loud signal peaks to pass In the drum kit example you could remove the hi hat signal which is higher in frequency with the High Cut filter and allow the snare signal to pass Turn monitoring off to set a suitable Threshold level more easily Silver Compressor The Silver Compressor is a simplified version of the Compressor for usage tips see Using the Compressor Gain Reduction meter Shows the amount of compression in real time Threshold slider and field Sets the threshold level
256. ted by Amp Designer can be combined in a number of ways to radically or subtly alter the tone Virtual microphones are used to pick up the signal of the emulated amplifier and cabinet You can choose from three different microphone types and you can reposition them 12 Amp Designer also emulates classic guitar amplifier effects including spring reverb vibrato and tremolo The Amp Designer interface can be broken down into four general sections in terms of different kinds of parameters Amp parameters Effects parameters Amp parameters name a L yee d Microphone parameters Model parameters The Model pop up menu is found at the left of the black bar at the bottom It is used to choose a preconfigured model consisting of an amplifier a cabinet an EQ type and a microphone type See Choosing an Amp Designer Model The model customizing parameters on the black bar allow you to independently choose the type of amplifier and cabinet See Building a Customized Amp Designer Combo The EQ type is chosen from the EQ pop up menu above the Bass Mids and Treble knobs in the knobs section See Using Amp Designer s Equalizer Amp parameters Located at each end of the knobs section these parameters are used to set an amp s input gain presence and output level See Using Amp Designer s Gain Presence and Master Controls Effects parameters Located in the center of the knobs section these parameters allow you to contr
257. tended Parameters area Introduces an additional allpass filter into the signal path An allpass filter shifts the phase angle of a signal influencing its stereo image All Pass Left and All Pass Right sliders and fields Extended Parameters area Determines the frequency at which the phase shift crosses 90 the half way point of the total 180 for each of the stereo channels In surround instances the other channels are automatically assigned values that fall between the two settings Chapter9 Modulation Effects 189 190 Phaser Effect The Phaser effect combines the original signal with a copy that is slightly out of phase with the original This means that the amplitudes of the two signals reach their highest and lowest points at slightly different times The timing differences between the two signals are modulated by two independent LFOs In addition the Phaser includes a filter circuit and a built in envelope follower that tracks volume changes in the input signal generating a dynamic control signal This control signal alters the sweep range Sonically phasing is used to create whooshing sweeping sounds that wander through the frequency spectrum It is a commonly used guitar effect but it is suitable for a range of signals Phaser Feedback Section Filter button Activates the filter section which processes the feedback signal LP and HP knobs and fields Set the cutoff frequency of the filter section s lowpass LP a
258. ter 6 Filter Effects e Resonance knob Resonance is responsible for the basic sonic character of the vocoder low settings result in a soft character whereas high settings lead to a more snarling sharp character Technically increasing the Resonance value emphasizes the middle frequency of each frequency band Using Formant Stretch and Formant Shift Formant Stretch and Formant Shift are significant Formant Filter parameters that you can use separately or in combination see EVOC 20 TrackOscillator Formant Filter Parameters When Formant Stretch is set to 0 the width and distribution of the bands in the Synthesis filter bank at the bottom matches the width of the bands in the Analysis filter bank at the top Low values narrow the width of each band in the Synthesis bank whereas high values widen the bands The control range is expressed as a ratio of the overall bandwidth When Formant Shift is set to 0 the positions of the bands in the Synthesis filter bank match the positions of the bands in the Analysis filter bank Positive values move the Synthesis filter bank bands up in frequency whereas negative values move them down in respect to the Analysis filter bank band positions When combined Formant Stretch and Formant Shift alter the formant structure of the resulting vocoder sound which can lead to some interesting timbre changes For example using speech signals and tuning Formant Shift up results in Mickey Mouse effects
259. ter settings of pedals added in import mode are also imported To replace a pedal setting in the Pedal area with an imported pedal setting Click the pedal you want to replace in the Pedal area It becomes highlighted with a blue outline Click the stompbox in the Pedal Browser to replace the selected pedal or pedal setting in the Pedal area The blue outlines of the selected pedal in the Pedal area and Pedal Browser blink on and off to indicate an imported setting The setting name area at the bottom of the Pedal Browser displays Click selected item again to revert Note If you want to make your replacement permanent click the background in the Pedal Browser or click the Import Mode button To restore the selected pedal s previous setting click the highlighted stompbox in the Pedal Browser The Import Mode button and the outline of the selected pedal in the Pedal area become solidly highlighted indicating that the original setting has been restored Using Pedalboard s Pedal Area Pedalboard s stompbox effect pedals not only resemble their physical counterparts they are also used in much the same way without the inconvenience of patch cords power supplies and screws or locking mechanisms The Pedal area layout mirrors a traditional pedalboard with signals running from left to right RATE DEPTH N Tone Fat Dive 6 Level Fast Slow SQUASH COMPRESSOR O Chapter 1 Amps and Pedals 39 40 T
260. tereo effect Correlation values lower than 0 indicate that out of phase material is present which can lead to phase cancellations if the stereo signal is combined into a monaural signal Level Meter Plug in The Level Meter displays the current signal level on a decibel scale The signal level for each channel is represented by a blue bar When the level exceeds 0 dB the portion of the bar to the right of the 0 dB point turns red Chapter 8 Metering Tools Stereo instances of the Level Meter show independent left and right bars whereas mono instances display a single bar Surround instances display a bar for each channel in a vertical rather than horizontal orientation The current peak values are displayed numerically superimposed over the graphic display You can reset these values by clicking in the display The Level Meter can be set to display levels using Peak RMS or Peak amp RMS characteristics Choose the appropriate setting in the pop up menu below the graphic display RMS levels appear as dark blue bars Peak levels appear as light blue bars You can also choose to view both Peak and RMS levels simultaneously Peak and RMS Explained The peak value is the highest level that the signal will reach The RMS root mean square value is the effective value of the total signal In other words it is a measurement of the continuous power of the signal Human hearing is optimized for capturing continuous signals making ou
261. ters The main filter parameters are found at the lower left corner of the interface Filter On Off button Switches the filter section on and off Filter Mode knob Determines the filter mode 6 dB LP Bright good general purpose filter mode It can be used to retain the top end of most material while still providing some filtering 12 dB LP Useful where you want a warmer sound without drastic filter effects It is handy for smoothing out bright reverbs BP 6 dB per octave design Reduces the lower and high end of the signal leaving the frequencies around the cutoff frequency intact HP 12 dB per octave two pole design Reduces the level of frequencies that fall below the cutoff frequency Reso nance knob Emphasizes frequencies above around or below the cutoff frequency The impact of the resonance knob on the sound is highly dependent on the chosen filter mode with steeper filter modes resulting in more pronounced tonal changes Using Space Designer s Filter Envelope The filter envelope is shown in the main display when the Filter Env button is active It provides control of the filter cutoff frequency over time All filter envelope parameters can be adjusted either numerically in the parameter bar or graphically in the main display using the techniques discussed in Setting Space Designer s Envelope Parameters Chapter 12 Space Designer Convolution Reverb Note Activation of the filter envelope autom
262. than AB recordings so they can lack asense of perceived width when played back XY recordings can be mixed down to mono Understanding MS Miking To make a Middle Side MS 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 that directly faces the sound source you want to record in a straight alignment The other is a bidirectional microphone with its axes pointing to the left and right of the sound source at 90 angles The cardioid microphone records the middle signal to one side of a stereo recording The bidirectional microphone records the side signal to the other side of a stereo recording MS recordings made in this way can be decoded by the Direction Mixer When MS recordings are played back the side signal is used twice As recorded Panned hard left and phase reversed panned hard right MS is ideal for all situations where you need to retain absolute mono compatibility The advantage of MS recordings over XY recordings is that the stereo middle is positioned on the main recording direction on axis of the cardioid microphone This means that slight fluctuations in frequency response that occur off the on axis as is the case with every microphone are less troublesome because the recording always retains mono compatibility Stereo Spread Stereo Spread is typically used when mastering There are se
263. the same value Settings below 50 result in every second increment being shorter in time Settings above 50 result in every second grid increment being longer in time Chapter 2 Delay Effects 69 70 Use subtle variations of the grid position of every second increment values between 45 and 55 to create a less rigid rhythmic feel This can deliver very human timing variations Use of extremely high Swing values are unsubtle as they place every second increment directly beside the subsequent increment Make use of higher values to create interesting and intricate double rhythms with some taps while retaining the grid to lock other taps into more rigid synchronization with the project tempo Using Delay Designer s Master Section The Master section incorporates two global functions delay feedback and dry wet mix In simple delays the only way for the delay to repeat is to use feedback Because Delay Designer offers 26 taps you can use these taps to create repeats rather than requiring discreet feedback controls for each tap Delay Designer s global Feedback parameter does however allow you to send the output of one user defined tap back through the effect input to create a self sustaining rhythm or pattern This tap is known as the feedback tap Tap A 6 0dB Mix wet dy Feedback button Enables or disables the feedback tap Feedback Tap pop up menu Used to choose a tap as the feedback tap Feed
264. the bottom Reverb can be added to either the Tremolo or Vibrato effect or used independently Reverb pop up menu Choose one of the three types of spring reverb Level knob Sets the amount of reverb applied to the pre amplified amp signal Setting Guitar Amp Pro Microphone Parameters After choosing a speaker cabinet from the Speaker menu you can set the type of microphone you want to be emulated and where the microphone is placed in relation to the speaker The Microphone Position parameters are available in the yellow area to the left and the Microphone Type parameters in the yellow area to the right Microphone Position Parameters Centered button 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 Off Center button Places the microphone on the edge of the speaker also referred to as off axis This placement produces a tone that is brighter and sharper but also thinner suitable for cutting rock or R amp B guitar parts When you select either button the graphic speaker display reflects your choice Microphone Type Parameters Condenser button Emulates the sound of a studio condenser microphone The sound of condenser microphones is fine transparent and well balanced Chapter 1 Amps and Pedals Dynamic button Emulates the sound of a dynamic cardioid microphone This microphone type sound
265. the duration of the reverb Low Cut slider and field Frequencies below the set value are filtered out of the reverb signal This affects only the tone of the reverb signal not the original signal High Cut slider and field Frequencies above the set value are filtered out of the reverb signal This affects only the tone of the reverb signal not the original signal Mod ulation Rate knob and field Sets the frequency the speed of the LFO Mod ulation Phase knob and field Defines the phase of the modulation between the left and right channels of the reverb signal At 0 the extreme values minimum or maximum of the modulation are achieved simultaneously on both the left and right channels Ata value of 180 the extreme values opposite each other left channel minimum right channel maximum or vice versa are reached simultaneously Mod ulation Intensity slider and field Sets the modulation amount A value of 0 turns the delay modulation off Mix slider and field Sets the balance between the effect wet and original dry signals Chapter 11 Reverb Effects 225 Space Designer Convolution Reverb Space Designer is a convolution reverb effect You can use it to place your audio signals in exceptionally realistic recreations of real world acoustic environments Space Designer generates reverb by convolving or combining an audio signal with an impulse response IR reverb sample An impulse response is a recording
266. the signal and it can add movement to low or sustained sounds Intensity slider and field Sets the modulation amount e Rate knob and field Defines the frequency and therefore the speed of the LFO e Mix slider and field Determines the balance of dry and wet signals Ensemble Effect The Ensemble combines up to eight chorus effects Two standard LFOs and one random LFO which generates random modulations enable you to create complex modulations The Ensemble s graphic display visually represents what is happening with the processed signals Chapter9 Modulation Effects The Ensemble effect can add a great deal of richness and movement to sounds particularly when you use a high number of voices It is very useful for thickening parts but it can also be used to emulate more extreme pitch variations between voices resulting in a detuned quality to processed material Intensity sliders and fields Set the amount of modulation for each LFO Rate knobs and fields Control the frequency of each LFO Voices slider and field Determines how many individual chorus instances are used and therefore how many voices or signals are generated in addition to the original signal Graphic display Indicates the shape and intensity of the modulations e Phase knob and field Controls the phase relationship between the individual voice modulations The value you choose here is dependent on the number of voices which is why it i
267. the volume envelope begins Decay Time field Sets the length of the decay phase Volume decay mode buttons Set the volume decay curve type Exp The output of the volume envelope is shaped by an exponential algorithm to generate the most natural sounding reverb tail Lin The volume decay will be more linear and less natural sounding End Level field Sets the end volume level It is expressed as a percentage of the overall volume envelope If set to 0 you can fade out the tail If set to 100 you can t fade out the tail and the reverb stops abruptly if the end point falls within the tail If the end time falls outside the reverb tail End Level has no effect Chapter 12 Space Designer Convolution Reverb Using Space Designer s Density Envelope The density envelope allows you to control the density of the synthesized impulse response over time You can adjust the density envelope numerically in the parameter bar and you can edit the Init Level Ramp Time and End Level parameters using the techniques described in Setting Space Designer s Envelope Parameters Note The density envelope is available only in Synthesized IR mode Weed definition 100 2 005 A Density Envelope init level ramp time end level reflection shape tJ 33 O8is 100 S LIIN Init Level field Sets the initial density the average number of reflections in a given period of time of the reverb Lowering the density l
268. ther end of the spectrum extreme processing is possible when you need to radically alter your sound All effects and processors are highly optimized for efficient CPU usage About the Logic Pro Documentation Logic Pro comes with various documents that will help you get started as well as provide detailed information about the included applications Logic Pro User Manual This manual provides comprehensive instructions for using Logic Pro to set up a recording system compose music edit audio and MIDI files and output audio for CD productions Exploring Logic Pro This booklet provides a fast paced introduction to the main features and tasks in Logic Pro encouraging hands on exploration for new users Logic Pro Control Surfaces Support This manual describes the configuration and use of control surfaces with Logic Pro Logic Pro Instruments This manual provides comprehensive instructions for using the powerful collection of instruments included with Logic Pro e Logic Pro Effects This manual provides comprehensive instructions for using the powerful collection of effects included with Logic Pro Working with Apogee Hardware This manual describes the use of Apogee hardware with Logic Pro Impulse Response Utility User Manual This manual provides comprehensive instructions for using Impulse Response Utility to create your own mono stereo and surround impulse responses for Space Designer the Logic Pro convolution based r
269. tion click a parameter value to reset it to the default setting If multiple taps are selected Option clicking a parameter of any tap resets all selected taps to the default value for that parameter filter HP cutoff LP slope eso d pitch transp flip pan spread mute level 20Hz 20000H2 MEM de CZ 7so oms ff E os so s EI Da centers 100 S M 0 648 Filter On Offbutton Enables or disables the highpass and lowpass filters for the selected tap HP Cutoff LP fields Sets the cutoff frequencies in Hz for the highpass and lowpass filters Slope buttons Determines the steepness of the highpass and lowpass filter slope Click the 6 dB button for a gentler filter slope or click the 12 dB button for a steeper more pronounced filtering effect Note You cannot set the slope of the highpass and lowpass filters independently Reso nance field Sets the amount of filter resonance for both filters Tap Delay fields Shows the number and name of the selected tap in the upper section and the delay time in the lower section e Pitch On Off button Enables or disables pitch transposition for the selected tap Transp ose fields The left field sets the amount of pitch transposition in semitones The right field fine tunes each semitone step in cents 1 100th of a semitone Flip buttons Swaps the left and right side of the stereo or surround
270. tions that you can clearly discern before the diffuse reverb tail builds up These early reflections are essential in human perception of spatial characteristics such as the size and shape of a room Signal Discrete Diffuse reverb tail reflections Reflection pattern reverberation This chapter covers the following e Plates Digital Reverb Effects and Convolution Reverb p 214 e AVerb p 214 e EnVerb p 215 e GoldVerb p 218 e PlatinumVerb p 221 e SilverVerb p 225 213 214 Plates Digital Reverb Effects and Convolution Reverb 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 metal plates and springs were also 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 The time differences between the original signal and the arrival of the early reflections can be adjusted by a parameter commonly known as predelay The average number of reflections in a given period of time is determined by the density parameter The regularity or irregularity of the density is controlled with the diffusion parameter Today s computers make it possible to sample the reverb characteristics of real spaces using convolution reverb
271. tle of the appropriate signal is reduced 3 If you encounter artifacts use the smoothing parameters Denoiser Main Parameters The Denoiser offers the following main parameters Threshold slider and field Noise Type slider and field Graphic display Reduce slider and field Threshold slider and field Sets the threshold level Signals that fall below this level are reduced by the Denoiser Reduce slider and field Sets the amount of noise reduction applied to signals that fall 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 250 Chapter 13 Specialized Effects and Utilities Note If the noise floor of your recording is very high more than 68 dB reducing it to a level of 83 to 78 dB should be sufficient provided this doesn t introduce any audible side effects This effectively reduces the noise by more than 10 dB to less than half of the original noise volume Noise Type slider and field Determines the type of noise that you want to reduce A value of 0 equals white noise equal frequency distribution Positive values change the noise type to pink noise harmonic noise greater bass response e Negative values change the noise type to blue noise hissy tape noise Graphic display Shows how the lowest volume levels of your audio material which should be mostly o
272. to stereo channels each tap contains a line showing its pan position For stereo to stereo channels each tap contains a dot showing its stereo balance A line extending outwards from the dot indicates the tap s stereo spread For surround channels each tap contains a line representing its surround angle for details see Working with Delay Designer in Surround e Level button Shows the relative volume level of each tap Tip You can temporarily switch the Tap display to Level view from one of the other view modes by pressing Command Option Chapter 2 Delay Effects Zooming and Navigating Delay Designer s Tap Display You can use Delay Designer s Overview display to zoom and to navigate the Tap display area Overview display Tip If the Overview display is hidden behind a tap you can move it to the foreground by holding down Shift To zoom the Tap display Do one of the following Vertically drag the highlighted section the bright rectangle of the Overview display Horizontally drag the highlighted bars to the left or right of the bright rectangle in the Overview display Note The Autozoom button needs to be disabled when manually zooming with the Overview display When you zoom in on a small group of taps the Overview display continues to show all taps The area shown in the Tap display is indicated by the bright rectangle in the Overview display Chapter 2 Delay Effects 55 56 To move to diffe
273. tone stepped glissandi but the intonation will be perfect If the Response value is too low the pitch of the output signal won t change quickly enough The optimum setting for this parameter depends on the singing style tempo vibrato and accuracy of the original performance Detune slider and field Detunes the output signal by the set value Defining the Pitch Correction Effect s Quantization Grid Use the Pitch Correction effect s 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 be used only for audio material that contains extremely low frequencies below 100 Hz which may result in inaccurate pitch detection These parameters have no effect on the sound they are simply optimized tracking options for the chosen target pitch range The Scale pop up menu allows you to choose different pitch quantization grids The scale that is set manually with the keyboard graphic in the plug in window is called the User Scale The default setting is the chromatic scale If you re unsure of the intervals used in any given scale choose it in the Scale menu and look at the keyboard graphic You can alter any note in the chosen scale by clicking the keyboard keys Any such adjustments overwrite the existing user scale settings There is only one user scale per project You can however create multiple user scales and sa
274. tons e Inv inverse mode In Tremolo mode the bass compartment rotates at a fast speed while the horn compartment rotates slowly This is reversed in Chorale mode In Brake mode both rotors stop Chapter9 Modulation Effects 910 The 910 or Memphis mode stops the bass drum rotation at slow speed while the speed of the horn compartment can be switched This may be desirable if you re after a solid bass sound but still want treble movement e Sync The acceleration and deceleration of the horn and bass drums are roughly the same This sounds as if the two are locked but the effect is clearly audible only during acceleration or deceleration Rotor Fast Rate slider Adjust to set the maximum possible rotor speed Tremolo The Tremolo rotation speed is displayed in Hertz Acc Dec Scale slider The Leslie motors need to physically accelerate and decelerate the speaker horns in the cabinets and their power to do so is limited Use the Acc Dec Scale parameter to determine the time it takes to get the rotors up to a determined speed and the length of time it takes for them to slow down Set the slider to the far left to switch to the preset speed immediately As you drag the slider to the right it takes more time to hear the speed changes e At the default position 1 the behavior is Leslie like Rotor Cabinet Microphone Parameters The Rotor Cabinet offers the following microphone parameters Mic Distance slider Dete
275. tor Output Parameters Chapter 6 Filter Effects 143 144 EVOC 20 TrackOscillator Analysis In Parameters The parameters in the Analysis In section determine how the input signal is analyzed and used by the EVOC 20 TrackOscillator You should be as precise as possible with these parameters to ensure the best possible speech intelligibility and accurate tracking A A Freeze Attack Release Analysis in Bands Syathess i a Attack knob Determines how quickly each envelope follower coupled to each analysis filter band reacts to rising signals Release knob Determines how quickly each envelope follower coupled to each analysis filter band reacts to falling signals Freeze button When enabled holds or freezes the current analysis sound spectrum indefinitely While Freeze is enabled the analysis filter bank ignores the input source and the Attack and Release knobs have no effect Bands field Determines the number up to 20 of frequency bands used by the EVOC 20 TrackOscillator Analysis In pop up menu Sets the analysis signal source The choices are Track Uses the input audio signal of the channel strip in which the EVOC 20 TrackOscillator is inserted as the analysis signal Side Chain Uses a side chain as the analysis signal You choose the side chain source channel strip from the Side Chain pop up menu at the top of the plug in window Note If Side Chain is chosen and no Side Chain channel s
276. tors or digital circuits Vacuum tubes were used in audio amplifiers before the development of digital audio technology and they are still used in musical instrument amplifiers today When overdriven they produce a type of distortion that many people find musically pleasing and which has become a familiar part of the sound of rock and pop music Analog tube distortion adds a distinctive warmth and bite to the signal There are also distortion effects that intentionally cause clipping and digital distortion of the signal These can be used to modify vocal music and other tracks to produce an intense unnatural effect or to create sound effects Distortion effects include parameters for tone which let you shape the way the distortion alters the signal often as a frequency based filter and for gain which let you control how much the distortion alters the output level of the signal Warning When set to high output levels distortion effects can damage your hearing and your speakers When you adjust effect settings it is recommended that you lower the output level of the track and raise the level gradually when you are finished This chapter covers the following e Bitcrusher p 78 e Clip Distortion p 79 e Distortion Effect p 80 e Distortion Il p 81 e Overdrive p 81 e Phase Distortion p 82 77 78 Bitcrusher Bitcrusher is a low resolution digital distortion effect You can use it to emulate the
277. toward the left end of the knobs section EQ pop up menu Click the word EQ or CUSTOM EQ above the Bass Mids and Treble knobs to open the EQ pop up menu which contains the following EQ models British Bright Vintage U S Classic Modern and Boutique Each EQ model has unique tonal qualities that affect the way the Bass Mids and Treble knobs respond See Amp Designer Equalizer Type Reference Table Bass Mids and Treble knobs Adjust the frequency ranges of the EQ models similar to the tone knobs on a hardware guitar amplifier The behavior and response of these knobs changes when different EQ models are chosen Amp Designer Equalizer Type Reference Table You can choose an Equalizer type by clicking the word EQ or CUSTOM EQ above the Bass Mids and Treble knobs in the knobs section The table below covers the properties of each EQ type available in Amp Designer EQ type Description British Bright Inspired by the EQ of British combo amps of the 1960s It is loud and aggressive with even bolder highs than the Vintage EQ This EQ is useful if you want more treble definition without an overly clean sound Vintage Emulates the EQ response of American Tweed style amps and the vintage British stack amps that used a very similar circuit It is loud and somewhat distortion prone This EQ is useful if you want to roughen the sound U S Classic Derived from the EQ circuit of the American Blackface style amps The tone is of hig
278. tracks It will however provide noticeable timing improvements on reasonably tight percussive and melodic material played in an eighth or quarter note feel If a large amount of timing correction is needed and transients are shifted too far you may notice a number of audio artifacts Therefore you should try to strike a balance between sound quality and timing enhancement Important For technical reasons the Enhance Timing plug in works only on audio channel strips and must be inserted in the top Insert slot x Audio 2 P View Show CS v Show Insert Bypass lt defaultdef Enhance Timing 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 Note Grid pop up menu are corrected Note Grid pop up menu Provides a choice of four grid divisions The grid divisions serve as reference points for the timing correction process As a tip for eighth note triplets try the 1 12 note setting Exciter The Exciter generates high frequency components that are not part of the original signal It does this by employing a nonlinear distortion process that resembles overdrive and distortion effects Unlike these effects however the Exciter passes the input signal through a highpass filter before feeding it into the harmonics distortion generator This results in artificial harmonics being added to the original
279. trigger the internal synthesizer Analysis ia Synthesis in Synthesis In pop up menu Sets the tracking signal source The choices are Oscillator Osc Sets the tracking oscillator as the synthesis source The oscillator mirrors or 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 Track Uses the input audio signal of the channel strip in which the EVOC 20 TrackOscillator is inserted as the synthesis signal which drives the internal synthesizer Side Chain Uses a side chain as the synthesis signal You choose the side chain source channel from the Side Chain pop up menu at the top of the EVOC 20 TrackOscillator window Note If you choose Side Chain and no Side Chain channel is assigned the EVOC 20 TrackOscillator reverts to Track mode operation Bands field Determines the number of frequency bands used by the Synthesis In section Basic Tracking Oscillator Parameters The tracking oscillator follows the pitch of incoming monophonic audio signals and mirrors these pitches with a synthesized sound The FM tone generator for the tracking oscillator consists of two oscillators each of which generates a sine wave The frequency of Oscillator 1 the carrier is modulated by Oscillator 2 the modulator which deforms the sine wave of Oscillator 1 This results
280. trip is assigned the EVOC 20 TrackOscillator reverts to Track mode operation Using EVOC 20 TrackOscillator Analysis In Parameters This section outlines some settings and approaches for the parameters of the Analysis In section Setting the Attack Time Longer attack times result in a slower tracking response to transients level spikes of the analysis input signal A long attack time on percussive input signals such as a spoken word or hi hat part will translate into a less articulated vocoder effect Therefore you should set the Attack parameter to the lowest possible value to enhance articulation Chapter 6 Filter Effects Setting the Release Time Longer release times cause the analysis input signal transients to sustain for a longer period at the vocoder s output A long release time on percussive input signals such as a spoken word or hi hat part will translate into a less articulated vocoder effect Use of extremely short release times results in rough grainy vocoder sounds Release values of around 8 to 10 ms are useful starting points Using Freeze The frozen analysis signal can capture a particular characteristic of the source signal which is then imposed as a complex sustained filter shape on the Synthesis section The following are examples of when this could be useful If you are using a spoken word pattern as a source the Freeze button could capture the attack or tail phase of an individual word within the patter
281. ts the amount of saturation Tone sets the overall color of the distortion Higher Tone values increase the treble content of the signal but there is a corresponding decrease in overall volume Texture can smooth out or roughen up the distortion Grain sets the amount of nonlinear distortion applied to the signal The effect output is controlled with the Level knob Octafuzz A fat fuzz effect that can deliver a soft saturated distortion Fuzz controls the input gain Level sets the ratio between the distorted and source signals The Tone knob sets the cutoff frequency of the highpass filter Rawk Distortion A metal hard rock distortion effect Crunch sets the amount of saturation applied to the input signal Output gain is set with Level Tonal color is set with the Tone knob making the sound brighter at higher values Vintage Drive Overdrive effect that emulates the distortion produced by a field effect transistor FET which is commonly used in solid state amplifiers When saturated FETs generate a warmer sounding distortion than bipolar transistors such as those emulated by Grinder Drive sets the saturation amount for the input signal Tone sets the frequency for the high cut filter resulting in a softer or harsher tone The Fat switch when at the top position enhances lower frequency content in the signal Level sets the overall output level of the effect Modulation Pedals This section describes the modulation eff
282. ture the signal in the simulated room Note Spacing the microphones slightly farther apart than the distance between two human ears generally delivers the best and most realistic results This parameter is available only in stereo instances of the effect ER Scale slider and field Extended Parameters area Scales the early reflections along the time axis influencing the Room Shape Room Size and Stereo Base parameters simultaneously Chapter 11 Reverb Effects PlatinumVerb Reverb Parameters The PlatinumVerb offers the following Reverb parameters Initial Delay slider and field Sets the time between the original signal and the diffuse reverb tail Spread slider and field Controls the stereo image of the reverb At 0 the effect generates a monaural reverb At 200 the stereo base is artificially expanded Crossover slider and field Defines the frequency at which the input signal is split into two frequency bands for separate processing Low Ratio slider and field Determines the relative reverb times of the bass and high bands It is expressed as a percentage At 100 the reverb time of the two bands is identical At values below 100 the reverb time of frequencies below the crossover frequency is shorter At values greater than 100 the reverb time for low frequencies is longer Low Freq Level slider and field Sets the level of the low frequency reverb signal At 0 dB the volume of the two bands is equal
283. ty slider Controls the amount of Formant Shift modulation by the Shift LFO Rate knobs and fields Determine the speed of modulation Values to the left of the center positions are synchronized with the host application tempo and include bar values triplet values and more Values to the right of the center positions are non synchronized and are displayed in Hertz cycles per second Note The ability to use synchronous bar values could be used to perform a formant shift every four bars on a cycled one bar percussion part for example Alternately you could perform the same formant shift on every eighth note triplet within the same part Either method can generate interesting results and can lead to new ideas or add new life to old audio material Chapter 6 Filter Effects 139 Waveform buttons Set the waveform type used by the Shift LFO on the left side or Fade LFO on the right side You can choose between triangle falling and rising sawtooth square up and down around zero bipolar good for trills square up from zero unipolar good for changing between two definable pitches a random stepped waveform S amp H and a smoothed random waveform for each LFO LFO Fade Intensity slider Controls the amount of Fade AB modulation by the Fade LFO Tip LFO modulations are the key to some extraordinary effects that can be obtained with the EVOC 20 Filterbank Set up either completely different or complementary filter curves in both filte
284. u to make precise adjustments To adjust Match EQ curve values Drag horizontally to shift the peak frequency for the band over the entire spectrum Drag vertically to adjust the gain of the band Shift drag vertically to adjust the Q Factor Option drag to reset the gain to 0 dB Note If you manually modify the filter curve you can restore it to the original or flat curve by Option clicking on the background of the Analyzer display Option click the background again to restore the most recent curve The Q factor of the filter is determined and set by the vertical distance between the clicked position and the curve To set the Match EQ Q factor Click directly on the curve to set the maximum Q value of 10 for notch like filters Click above or below the curve to decrease the Q value The farther you click from the curve the smaller the value down to the minimum of 0 3 The colors and modes of the dB scales on the left and right of the display are automatically adapted to the active function If the Analyzer is active the left scale displays the average spectrum in the signal while the right scale serves as a reference for the peak values of the Analyzer A dynamic range of 60 dB is shown by default If this is not precise enough for your edits you can increase the range To change the Match EQ scale range Drag either scale to set values of up to 20 dB and 100 cB To change Match EQ gain with the scales Drag either s
285. uld set in order to effectively pitch shift the signal Crossfade slider and field Extended Parameters area Sets the range expressed as a percentage of the original signal used to analyze the input signal Stereo Link radio buttons Extended Parameters area Select Inv to invert the stereo channel s signals with processing for the right channel occurring on the left and vice versa Select Normal to leave the signal as it is Follow these steps when pitch shifting Set the Semi Tones slider for the amount of transposition or pitch shift Set the Cents slider for the amount of detuning Click the Drums Speech or Vocals button to select the algorithm that best matches the material you are working with If you are working with material that doesn t fit any of these categories experiment with each of the algorithms starting with Speech compare the results and use the one that best suits your material Tip While auditioning and comparing different settings it s often a good idea to temporarily set the Mix parameter to 100 as Pitch Shifter Il artifacts are easier to hear Vocal Transformer The Vocal Transformer can be used 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 that mirrors the pitches of a melody No matter how you change the pitches of the melody the constituent parts of the signal formants remain the same You can shift th
286. unique tool that can be used to achieve results that differ from other dynamic processors Threshold slider and field Sets the threshold level Signals that exceed the threshold have their attack and release phase levels altered Attack Gain slider and field Boosts or attenuates the attack phase of the signal When the Gain slider is set to the center position 0 the signal is unaffected Lookahead slider and field Sets the pre read analysis time for the incoming signal This enables the Enveloper to know in advance what signals are coming enabling accurate and fast processing Attack Time knob and field Determines the amount of time it takes for the signal to increase from the threshold level to the maximum Gain level e Display Shows the attack and release curves applied to the signal Release Time knob and field Determines the amount of time it takes for the signal to fall from the maximum gain level to the threshold level Release Gain slider and field Boosts or attenuates the release phase of the signal When the Gain slider is set to the center position 0 the signal is unaffected Out Level slider and field Sets the level of the output signal Using the Enveloper The most important parameters of the Enveloper are the two Gain sliders one on each side of the central display These govern the Attack and Release levels of each respective phase Chapter 4 Dynamics Processors Bo
287. us The ring modulator was used extensively on jazz rock and fusion records in the early 1970s The frequency shifter moves the frequency content of the input signal by a fixed amount and in doing so alters the frequency relationship of the original harmonics The resulting sounds range from sweet and spacious phasing effects to strange robotic timbres Note Frequency shifting should not be confused with pitch shifting Pitch shifting transposes the original signal leaving its harmonic frequency relationship intact Chapter9 Modulation Effects 192 Getting to Know the Ringshifter Interface The Ringshifter interface consists of six main sections Mode buttons Delay parameters Output parameters EEE S Freq Shift TErequenc Oscillator parameters Envelope follower LFO parameters parameters Mode buttons Determine whether the Ringshifter operates as frequency shifter or ring modulator See Setting the Ringshifter Mode Oscillator parameters Use these to configure the internal sine wave oscillator which modulates the amplitude of the input signal in both frequency shifter modes and the ring modulator OSC mode See Using the Ringshifter s Oscillator Delay parameters Use these to delay the effect signal See Using the Ringshifter s Delay Envelope follower parameters The oscillator frequency and output signal can be modulated with an envelope follower See Modulating the Ringshifter with the Envelope Followe
288. usically This is because human hearing is more responsive to the overall power of the signal than to single peaks As a basic setting for most applications the centered position is recommended Setting Multipressor Output Parameters The Out slider sets the overall output level Set Lookahead to higher values when the Peak RMS fields are set to higher values farther towards RMS Set Auto Gain to On to reference the overall processing to 0 dB making the output louder Noise Gate The Noise Gate is commonly used to suppress unwanted noise that is audible when the audio signal is at a low level You can use it to remove background noise crosstalk from other signal sources and low level hum among other uses The Noise Gate works by allowing signals above the threshold level to pass unimpeded while reducing signals below the threshold level This effectively removes lower level parts of the signal while allowing the desired parts of the audio to pass Chapter 4 Dynamics Processors 103 104 Noise Gate Parameters The Noise Gate has the following parameters Threshold slider and field Sets the threshold level Signals that fall below the threshold will be reduced in level Reduction slider and field Sets the amount of signal reduction Attack knob and field Sets the amount of time it takes to fully open the gate after the signal exceeds the threshold Hold knob and field Sets the amount of time the gate is kept open after the
289. v Bypass lt gt default Bypass lt gt default High Pass Filter Low Pass Filter Frequency 1000Hz Frequency 2000Hz amp Order 3n 4 Order Smoothing Frequency slider and field Sets the cutoff frequency Order slider and field Sets the filter order The more orders used the stronger the filtering effect Smoothing slider and field Adjusts the amount of smoothing in milliseconds High Shelving and Low Shelving EQ The Low Shelving EQ affects only the frequency range that falls below the selected frequency The High Shelving EQ affects only the frequency range above the selected frequency Inst 1 Inst 1 gp View y ShowCS v Show insert v p View ShowCS v Show insert v Bypass ki gt default Bypass lt gt default High Shelving EQ Low Shelving EQ Gain 0 0dB a Gain 0 0dB a Frequency 10000Hz Frequency 100Hz a Gain slider and field Sets the amount of cut or boost Frequency slider and field Sets the cutoff frequency 128 Chapter5 Equalizers Parametric EQ The Parametric EQ is a simple filter with a variable center frequency It can be used to boost or cut any frequency band in the audio spectrum either with a wide frequency range or as a notch filter with a very narrow range A symmetrical frequency range on either side of the center frequency is boosted or cut x Inst 1 gp View ShowCS v Show insert v Bypass lt gt default Param
290. ve them as Pitch Correction plug in settings files Tip The drone scale uses a fifth as a quantization grid and the single scale defines a single note Neither of these scales is meant to result in realistic singing voices so if you re after interesting effects you should give them both a try Open the Root pop up menu to choose the root note of the scale If you chose user scale or chromatic in the Scale pop up menu the Root pop up menu is non functional You may freely transpose the major and minor scales and scales named after chords Chapter 10 Pitch Effects 205 206 Excluding Notes from Pitch Correction You can use the Pitch Correction effect s onscreen keyboard to exclude notes from the pitch quantization grid When you first open the effect all notes of the chromatic scale are selected This means that every incoming note will be altered to fit the next semitone step of the chromatic scale If the intonation of the singer is poor this might lead to notes being incorrectly identified and corrected to an unwanted pitch For example the singer may have intended to sing an E but the note is actually closer to a D If you don t want the D in the song the D key can be disabled on the keyboard Because the original pitch was sung closer to an E than a D it will be corrected to an E Note The settings are valid for all octave ranges Individual settings for different octaves aren t provided Use of the small bypass button
291. vels are shown simultaneously with RMS levels appearing as dark blue bars and Peak levels appearing as light blue bars When the level exceeds 0 dB the portion of the bar above the 0 dB mark turns red Current peak values are displayed numerically in dB increments above the Level Meter Click in the display to reset peak values Using the MultiMeter s Correlation Meter The Correlation Meter gauges the phase relationship of a stereo signal The Correlation Meter s scale values indicate the following A 1 correlation value indicates that the left and right channels correlate 100 In other words the left and right signals are in phase and are the same shape Correlation values in the blue zone between 1 and the middle position indicate that the stereo signal is mono compatible e The middle position indicates the highest allowable amount of left right divergence which is often audible as an extremely wide stereo effect Chapter 8 Metering Tools 173 174 e When the Correlation Meter moves into the red area to the left of the center position out of phase material is present This will lead to phase cancellations if the stereo signal is combined into a mono signal Using the MultiMeter Peak Parameters The MultiMeter Peak parameters are used to enable disable the peak hold function and to reset the peak segments of all meter types You can also determine a temporary peak hold duration e Hold button Activates peak
292. veral ways to extend the stereo base or perception of space including use of reverbs or other effects and altering the signal s phase These options can all sound great but may also weaken the overall sound of your mix by ruining transient responses for example Chapter 7 Imaging Processors 165 166 Stereo Spread extends the stereo base by distributing a selectable number of frequency bands from the middle frequency range to the left and right channels This is done alternately middle frequencies to the left channel middle frequencies to the right channel and so on This greatly increases the perception of stereo width without making the sound totally unnatural especially when used on mono recordings Lower Int ensity slider and field Sets the amount of stereo base extension for the lower frequency bands Upper Int ensity slider and field Sets the amount of stereo base extension for the upper frequency bands Note When setting the Lower and Upper Int sliders be aware that the stereo effect is most apparent in the middle and higher frequencies so distributing low frequencies between the left and right speakers can significantly alter the energy of the overall mix For this reason use low values for the Lower Int parameter and avoid setting the Lower Freq parameter below 300 Hz Graphic display Shows the number of bands the signal is divided into and the intensity of the Stereo Spread effect in the upper and lower f
293. vert buttons Invert the phase of the selected channel e Mute buttons Mute the selected channel Test Oscillator The Test Oscillator is useful for tuning studio equipment and instruments and can be inserted as both an instrument or effect plug in It operates in two modes generating either a static frequency or a sine sweep Chapter 14 Utilities and Tools 263 264 In the first mode default mode it starts generating the test signal as soon as it is inserted You can switch it off by bypassing it In the second mode activated by clicking the Sine Sweep button Test Oscillator generates a user defined frequency spectrum tone sweep when triggered with the Trigger button 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 If the Sine Sweep button is active the fixed oscillator settings in the Waveform section are disabled Frequency knob and field Determines the frequency of the oscillator default is 1 kHz Sine Sweep button Generates a sine wave sweep of the frequency spectrum you set with the Start Freq and End Freq fields Time field Sets the duration of the sine wave sweep Start Freq and End Freq fields Drag vertically to define the oscillator frequency
294. y and depth also called intensity controls You can also control the ratio of the affected wet signal and the original dry signal Some modulation effects include feedback parameters which add part of the effect s output back into the effect input Other modulation effects involve pitch The most basic type of pitch modulation effect is vibrato It uses an LFO to modulate the frequency of the sound Unlike other pitch modulation effects vibrato alters only the delayed signal More complex modulation effects such as Ensemble mix several delayed signals with the original signal This chapter covers the following e Chorus Effect p 184 e Ensemble Effect p 184 Flanger Effect p 186 Microphaser p 187 Modulation Delay p 187 Phaser Effect p 190 Ringshifter p 191 Rotor Cabinet Effect p 197 Scanner Vibrato Effect p 199 183 184 Spreader p 201 e Tremolo Effect p 202 Chorus Effect 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 incoming signal and create the impression that multiple instruments or voices are being played in unison The slight delay time variations generated by the LFO simulate the subtle pitch and timing differences heard when several musicians or vocalists perform together Using chorus also adds fullness or richness to

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