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Audio Plug-Ins Guide v11.2

1. 227 Reel Tape Common Controls 228 Reel Tape Flanger Controls 228 Synchronizing Reel Tape Flanger to Session Tempo 230 Reel Tape Flanger Tips 231 Reel Tape Flanger Presets 231 Chapter 40 Sci Fi 232 Sci Fi Controls 232 Chapter 41 Voce Plug Ins 235 Voce Chorus Vibrato 235 Voce Spin 236 Contents x Part VIII Harmonic Plug Ins Chapter 42 Aphex Aural Exciter Type III
2. 333 Signal Generator Controls 333 AudioSuite Processing with Signal Generator 334 Chapter 58 SoundReplacer 335 Audio Replacement Techniques 335 SoundReplacer Controls 336 Using SoundReplacer 339 Getting Optimum Results with SoundReplacer 340 Using the Audio Files Folder for Frequently Used SoundReplacer Files 342 Chapter 59 Time Compression Expansion 343 Time Compression Expansion Controls 343 Chapter 60 Trim 345 Trim Controls
3. 81 D3 Limiter Controls 83 Using the Side Chain Input in D3 84 Chapter 15 Impact 85 Impact Controls 85 Using the Impact Compressor 87 Chapter 16 JOEMEEK SC2 Compressor 89 JOEMEEK Compressor Controls 89 JOEMEEK Compressor Tips and Tricks 90 Chapter 17 Maxim 91 About Peak Limiting 92 How Maxim Differs From Conventional Limiters 93 Ma
4. 345 Chapter 61 Other AudioSuite Plug In Utilities 346 DC Offset Removal 346 Duplicate 347 Gain 347 Invert 348 Normalize 348 Reverse 349 Index 350 Part I Introduction to Audio Plug Ins Chapter 1 Audio Plug Ins Overview 2 Chapter 1 Audio Plug Ins Overview Plug Ins are special purpose software components that provide additional signal processing and other functionality to Pro Tools HDX Pro Tools HD Native and Pro Tools Software systems These
5. 241 Meters 243 Rotary Controls 243 Switches 245 Using Aural Exciter III 247 Chapter 43 Aphex Big Bottom Pro 250 Meters 251 Rotary Controls 252 Switches 252 Using Big Bottom Pro 254 Chapter 44 Eleven 255 Eleven Input Calibration and QuickStart
6. 17 7 Band EQ III 20 Contents Contents iv Chapter 5 Focusrite D2 26 D2 Configurations 26 D2 Controls 27 Using D2 in Stereo 31 Chapter 6 JOEMEEK VC5 Meequalizer 32 JOEMEEK Meequalizer Controls 32 Chapter 7 Pultec Plug Ins 33 Pultec EQP 1A 33 Pultec EQH 2 34 Pultec MEQ 5
7. 122 AudioSuite Processing with Pro Limiter 123 AudioSuite Processing with Pro Limiter Loudness Analyzer 124 Chapter 21 Purple Audio MC77 125 Purple Audio MC77 Controls 125 Chapter 22 Smack 126 Smack Controls and Meters 127 Using the Smack Side Chain Input 131 Contents vii Part IV Pitch and Time Shift Plug Ins Chapter 23 Pitch II 133 Pitch II Controls 134 Chapter 24 Time Shift 137 Time Shift Controls
8. 43 Chapter 11 Channel Strip 44 Channel Strip Sections and Panes 45 Channel Strip Input Section 47 Channel Strip Output Section 48 Channel Strip FX Chain 49 Channel Strip Dynamics Section 49 Channel Strip EQ Filters Section 55 Contents v Chapter 12 Dynamics III 59 Dynamics III Common Controls 59 Compressor Limiter III 62 Expander Gate III 65 De Es
9. 34 Pultec Tips and Tricks 35 Part III Dynamics Plug Ins Chapter 8 BF 2A 37 BF 2A Controls 38 Using the BF 2A Side Chain Filter 38 BF 2A Tips and Tricks 39 Chapter 9 BF 3A 40 BF 3A Controls 40 BF 3A Tips and Tricks 41 Chapter 10 BF76 42 BF76 Controls 42 BF76 Tips and Tricks
10. 137 AudioSuite Input Modes and Time Shift 142 AudioSuite Preview and Time Shift 142 Time Shift as AudioSuite TCE Plug In Preference 142 Processing Audio Using Time Shift 142 Post Production Pull Up and Pull Down Tasks with Time Shift 144 Chapter 25 Vari Fi 145 Vari Fi Controls 145 Chapter 26 X Form 147 X Form Displays and Controls Overview 147 X Form AudioSuite Input Modes 151 AudioSuite TCE Plug In Preference 151 Processing Audio Using X Form
11. 107 Pro Expander Input Section 108 Pro Expander Output Section 109 Pro Expander Dynamics Graph 109 Pro Expander Controls 112 Pro Expander Side Chain Processing 114 Chapter 20 Pro Limiter 117 Pro Limiter Metering 118 Pro Limiter Input Section 118 Pro Limiter Output Section 120 Pro Limiter Controls 120 Pro Limiter Loudness Numeric Displays 122 Pro Limiter Histogram and Loudness Meters
12. 190 Space Snapshots 190 Space Controls and Displays 191 Space Display Area 192 Space IR Browser 195 Space Primary Controls 197 Space Group Selectors and Controls 198 Using Space 201 Space IR Library Categories 202 Part VI Delay Plug Ins Chapter 31 Mod Delay III 204 Mod Delay III Controls 204 Selections for Mod Delay III AudioSuite Processing 206 Chapter 32 Moogerfooger Ana
13. 152 Using X Form for Post Production Pull Up and Pull Down Tasks 153 Part V Reverb Plug Ins Chapter 27 D Verb 155 D Verb Controls 155 Selections for D Verb AudioSuite Processing 157 Chapter 28 Reverb One 158 About Reverb 159 Reverb One Controls 160 Reverb One Graphs 164 Other Reverb One Controls 166 Chapter 29 ReVibe II 167 Using ReVibe II 167 Dragging
14. 257 Using Eleven 261 Eleven Tips and Suggestions 278 Eleven Signal Flow Notes 280 Chapter 45 Lo Fi 281 Lo Fi Controls 281 Chapter 46 Recti Fi 283 Recti Fi Controls 283 Chapter 47 Reel Tape Saturation 286 Reel Tape Common Controls 286 Reel Tape Saturation Controls 287 Reel Tape Saturation Tips 288 Reel Tape Saturatio
15. 302 Chapter 52 Down Mixer 304 Source 305 Downmix 305 Part XI Instrument Plug Ins Chapter 53 Click II 307 Click II Controls and Displays 307 Creating a Click Track 309 Chapter 54 ReWire 310 ReWire Requirements 312 Using ReWire 313 MIDI Automation with ReWire 314 Quitting ReWire Client Applications
16. 92 Treadplate Vintage DC Modern Overdrive DC Vintage Crunch These models only appear in the full version of Eleven Choosing an amp from the Amp Type selector Eleven is not affiliated with or sponsored or endorsed by the makers of the amplifiers em ulated in the product Chapter 44 Eleven 266 Eleven Amp Controls Each Eleven amp provides a set of controls similar to and in some cases identical to those on the actual amp it models The following sections give a general overview of amp controls Amp Bypass The Amp Bypass switch or lamp lets you bypass just the amp model leaving the cab and mic set tings in effect The default setting is On When set to Bypass only the amp is bypassed Master sec tion cabinet and microphone settings remain ac tive Bright The Bright switch provides extra high frequency response to the input signal and alters the timbre of the distortion On some amp models the effect is most apparent at lower volume settings Gain 1 Gain 1 determines the overall gain amount and sen sitivity of the amp When Gain 1 is low it allows for cleaner brighter sounds with enhanced dynamic response When set high the entire personality of the amp changes becoming fatter and overdriven Gain 1 responds differently with each amp model and is designed to have a musical response that closely matches that of its original amp at all set tings The default setting is 5 0
17. Shower Stall Hallway Closet Classroom 1 Classroom 2 Large Concrete Room Medium Concrete Room Locker Room Muffled Room Very Small Room 1 Chapter 29 ReVibe II 180 Very Small Room 2 Very Small Room 3 Car 1 Car 2 Car 3 Car 4 Car 5 Phone Booth Metal Garbage Can Drain Pipe Tin Can Large Spaces Parking Garage 1 Parking Garage 2 Parking Garage 3 Warehouse 1 Warehouse 2 Stairwell 1 Stairwell 2 Stairwell 3 Stairwell 4 Stairwell 5 Gymnasium Auditorium Indoor Arena Stadium 1 Stadium 2 Tunnel Vintage Digital Large Hall Digital Medium Hall Digital Large Room Digital Medium Room Digital Small Room Digital Effects Mono Slapback 1 Mono Slapback 2 Mono Slapback 3 Wide Slapback 1 Wide Slapback 2 Wide Slapback 3 Multi Slapback 1 Multi Slapback 2 Multi Slapback 3 Multi Slapback 4 Spread Slapback 1 Spread Slapback 2 Mono Echo 1 Mono Echo 2 Mono Echo 3 Wide Echo 1 Wide Echo 2 Multi Echo 1 Multi Echo 2 Prism Prism Reverse Inverse Long Inverse Medium Inverse Short Stereo Enhance 1 Stereo Enhance 2 Stereo Enhance 3
18. Graph examples of hard knee left and soft knee right compression Chapter 12 Dynamics III 65 Compressor Limiter III Gain Control The Gain control lets you boost overall output gain to compensate for heavily compressed or limited signals This control ranges from 0 dB no gain boost to 40 dB loudest gain boost with the default value at 0 dB Compressor Limiter III Side Chain Section The side chain is the split off signal used by the plug in s detector to trigger dynamics processing The Side Chain section lets you toggle the side chain between the internal input signal or an exter nal key input and tailor the equalization of the side chain signal so that the triggering of dynamics processing becomes frequency sensitive See Dy namics III Side Chain Input on page 70 Expander Gate III The Expander Gate plug in applies expansion or gating to audio material depending on the ratio set ting About Expansion Expansion decreases the gain of signals that fall be low a chosen threshold They are particularly use ful for reducing noise or signal leakage that creeps into recorded material as its level falls as often oc curs in the case of headphone leakage About Gating Gating silences signals that fall below a chosen threshold To enable gating simply set the Ratio and Range controls to their maximum values Expanders can be thought of as soft noise gates since they provide a gentler way of
19. When set to Gain Reduction the meter needle moves backward from 0 to show the amount of compression being applied to the signal in dB When set to Output the needle indicates the out put level of the signal The meter is calibrated with 0 VU indicating 18 dBFS Using the BF 2A Side Chain Filter The BF 2A provides an extra a side chain filter that does not have a control on the plug in inter face but that can be accessed on screen through Pro Tools automation controls In addition the side chain filter can be adjusted directly from any supported control surface This side chain filter reproduces the effect of an ad justable resistor on the back panel of the LA 2A This control cuts the low frequencies from the side chain or control signal that determines the amount of gain reduction applied by the compressor By increasing the value of the side chain filter you filter out frequencies below 250 Hz from the con trol signal and decrease their effect on gain reduc tion A setting of zero means that the filter is not ap plied to the side chain signal A setting of 100 means that all frequencies be low 250 Hz are filtered out of the side chain signal To access the side chain filter on screen 1 Click the Plug In Automation button in the Plug In window to open the Automation Enable window 2 In the list of controls at the left click to select Side Chain Filter and click Add or just double
20. Chapter 11 Channel Strip 45 Channel Strip Sections and Panes The Channel Strip plug in window is organized in several sections Input FX Chain Output Dynam ics and EQ Filters The Dynamics and EQ Filters sections can be independently shown or hidden This lets you access controls or free up screen space depending on your needs When showing the Dynamics or EQ Filters sec tions several tabbed panes of controls are available for each section You can click a tab to show the controls for that tabbed pane For Expand Gate and Compressor Limiter and also for the For the EQ and Filter effects clicking the corresponding con trol point on the graph display automatically shows the tab for Expander Gate or the Compressor Lim iter or the corresponding EQ band or Filter Showing or Hiding the Dynamics and EQ Filters Sections You can independently show or hide the Dynamics and EX Filters sections of the Channel Strip plug in to use less screen space These sections are shown by default To hide or show the Dynamics or EQ Filters section of the plug in window Click the Show Hide triangle to the left of the section you want to show or hide Channel Strip Dynamics section hidden Chapter 11 Channel Strip 46 Disabling or Enabling Channel Strip Effects You can independently disable effects in the Dy namics and EX Filters sections of the Channel Strip plug in For example you may want to apply Comp Limit pr
21. Phase Switch The Phase switch allows you to alter the phase of the side chain signal which contains the Big Bot tom Pro effect before it is mixed with the original input signal This function is used as a optional way to change the quality of the Big Bottom Pro ef fect The switch illuminates when the Phase switch is activated Altering the side chain signal s phase dramatically effects the sound of the Big Bottom Pro enhance ment With the Phase switch turned Off you will recognize the Big Bottom Pro effect found in the Aphex Model 104 As an exclusive feature for this DSP plug in we have added the Phase switch When activated the Phase switch alters the Big Bottom Pro effect by setting the side chain in phase with the main sig nal This increases the output peak level Use the Mix or Level controls to restore the output peak level if the Drive meter indicates clipping AutoTrace Switch Activating the AutoTrace switch enables an auto matic threshold function for the compressor within the Big Bottom Pro side chain The AutoTrace function enables the dynamic processor to self op timize during normal operation The switch illumi nates when the AutoTrace switch is activated This control is particularly useful when you want a subtle Big Bottom Pro effect or when the peak level of the input material varies over time The AutoTrace feature is also ideal for changing the sound characteristics of the Big Bottom
22. SPECTRAL PHASE REFRACTOR PHASE Degrees TIME DELAY Milliseconds AUDIO FREQUENCY Hz PSYCHOACOUSTIC PHASE 400 350 300 250 200 150 100 50 0 50 10 100 1000 10000 12 0 2 4 6 8 10 LEADING PSYCHOACOUSTIC PHASE RESPONSE ACTUAL TIME DELAY Chapter 42 Aphex Aural Exciter Type III 247 LR Left Right Switches The LR switch is for stereo operation only It al lows you to view or change parameters on one channel at a time The switch for the currently displayed channel illu minates To edit both channels simultaneously click the Link switch Using Aural Exciter III In the recording studio post production suite or similar environment post processing of previously recorded audio tracks with Aural Exciter can re store lost vibrancy and realism even to the extent of saving dialogue or sound effects which were thought to be unusable Instruments and vocals can be made to stand out in the mix without substan tially increasing the mix levels or using equaliza tion The Pro Tools mixing environment is very flexible offering several ways to route and use Aural Ex citer in a session This section provides some sug gestions for efficient use of Aural Exciter in your Pro Tools sessions The exact steps you take to use Aural Exciter s capabilities will differ depending on the nature of your session and your specific Pro Tools mixer configuration Setting th
23. G Gain plug in 347 guitar amp simulators 255 289 I Impact plug in 85 Attack 85 clip indicator 87 compression ratio 87 External On Off 86 gain make up 86 Gain Reduction meter 86 Input Output Meter 87 Key Listen On Off 86 Make Up 86 meters 86 Ratio 85 Release 86 side chain processing 87 Threshold 85 impulse response IR 187 installing plug ins 8 internal side chain processing 84 InTune plug in 319 Automatic mode 320 Chromatic mode 322 creating presets 322 Display Flat Semitones option 323 factory presets 322 information display 322 Meter selector 321 Needle meter 321 note entry fields 323 Note Selection buttons 321 Octave range 321 reference frequency 321 Single Octave mode 322 Audio Plug Ins Guide 354 Strobe display 321 test tones 320 Tone Volume slider 322 tuner 319 Tuner Programming screen 320 322 Invert plug in 348 J JOEMEEK SC2 Compressor plug in compound release circuit 90 deadness avoiding 90 overshoot 90 K Key Listen parameter 80 84 Key On Off parameter 84 L limiter Dynamics III Compressor Limiter 62 Fairchild 660 75 Fairchild 670 77 Focusrite D3 78 Lo Fi plug in 281 adaptive quantization 282 aliasing artifacts 282 Anti Alias Filter control 282 Distortion Saturation controls 282 down processing audio 281 Linear Quantization control 282 Noise Generator 282 Output Meter 282 Quantization control 282 Sample Rate control 281 Sample Size control 282 M MasterMeter plug in 325 Clear
24. Release The Release control sets how long it takes for the expansion to close or the gate to open after the in put signal falls below the Threshold level and the Hold time has passed Depth The Depth control sets the depth of the processing expansion or gating when closed Setting the pro cessing to higher Depth levels allows more audio that falls below the threshold to remain audible Hold The Hold control specifies the duration in seconds or milliseconds during which Pro Expander stays in effect after the initial attack occurs This can keep processing in effect for longer periods of time with a single crossing of the threshold It can also be used to prevent gate chatter that may occur if varying input levels near the threshold cause the gate to close and open very rapidly Hysteresis The Hysteresis control lets you adjust whether or not the gate rapidly opens and closes when the in put signal fluctuates near the Threshold This can help prevent undesirably rapid gating of the signal This control is only available when Ratio is set to Gate Dry Mix The Dry Mix control sets the balance between the processed signal wet and the original signal dry The Dry Mix setting determines how much of the original signal is sent to the output rather than the processed signal For example at 30 the output will be 30 dry and 70 wet Turn the Dry Mix knob counterclockwise to 0 to pass only the pro cessed signal 100
25. Tel Ray Variable Delay operates as a mono multi mono or stereo plug in Add delay or echo to any voice or instrument using the Tel Ray Variable Delay It provides lush delay amazing echo and warms up your tracks and mixes In the early 1960s a small company experimented with electronics and technology When they came up with something great they would Tell Ray the boss One invention involved a tuna can a motor and a few tablespoons of oil The creation an Electronic Memory Unit A technology they were sure that would be of great interest to companies like IBM and NASA Though it never made it to the moon most every major guitar amp manufacturer licensed the tech nology that gives Tel Ray its unique sound Tel Ray Variable Delay Space age technology in a can Chapter 34 Tel Ray Variable Delay 214 Tel Ray Controls Input Output Section Input Input sets the signal level to the tuna can echo unit Tone Tone is a standard tone control like those commonly found on guitar effects Mix Mix adjusts the amount of dry unprocessed signal relative to the amount of wet processed sig nal Full clockwise is 100 wet On original units this control is located deep inside the box typically soaked in carcinogenic PCB oil Output Output is a simple digital output trim con trol Echo Delay Section Variable Delay Variable Delay selects the delay time Delay times vary from 0 06 to 0 3 seconds Full clo
26. TimeAdjuster Controls The TimeAdjuster plug in provides the following controls Phase Invert This controls inverts the phase polar ity of the input signal While most Avid plug ins supply a phase invert button of their own some third party plug ins may not Phase inversion is also useful for performing delay compensation by tuning unknown delay factors by ear see Using TimeAdjuster for Manual Delay Compensation on page 216 Gain Provides up to 24 dB of positive or negative gain adjustment This control is useful for alter ing the gain of a signal by a large amount in real time For example when you are working with audio signals that are extremely low level you may want to adjust the channel gain to a reason able working range so that a fader is positioned at its optimum travel position Use the Gain control as an insert effect to make a wide range of gain adjustment in real time without having to perma nently process the audio files as you would with an AudioSuite plug in Delay Provides up to 8192 samples of delay com pensation adjustment or general adjustment of phase relationships of audio recorded with multiple microphones the amount of delay available de pends on the version of TimeAdjuster Short Me dium or Long It defaults to a minimum delay of four samples which is the delay created by use of the TimeAdjuster plug in itself TimeAdjuster Stereo For more information on Delay Compensa tio
27. With external key side chain processing you can trigger dynamics processing using an external sig nal such as a separate reference track or audio source instead of the input signal This external source is known as the key input With side chain filters you can make dynamics processing more or less sensitive to certain fre quencies For example you might configure the side chain so that certain lower frequencies on a drum track trigger dynamics processing Source The Source selector lets you set the source for side chain processing Int Stereo Pairs Ext All w LFE Int All no LFE or Int Front Rear Internal Source Stereo Pairs When Int StereoPairs is selected the plug in uses the amplitude of the input signal to trigger dynam ics processing based on stereo input With greater than stereo multichannel processing the input sig nal for each stereo pair affects only those same channels and likewise mono channels are affected only by their own input signal For example with an LCR multichannel format the processing for the Center channel is only triggered when the Center Side Chain section Selecting the Source for side chain processing Chapter 18 Pro Compressor 105 channel input signal reaches the threshold How ever when the input signal reaches the threshold on the Left or the Right channel processing is trig gered for both the Left and the Right channel External All with LFE When Ext All w
28. ant to understand that the amplitude of the wave form will invariably exceed the sample values Manifestation Today s recording environment demands that ses sions are mixed and mastered as hot as is possi ble pushing the levels up to the highest tolerable amount supposedly just short of clipping Sophis ticated digital tools allow music to be highly com pressed then recompressed compressed even more so with multi band compressors limited nor malized and maximized to get the audio to play as loud as possible out of a consumer s system Hence it is very common for popular music CDs to be full of digital samples that are at or nearly at full scale The problem is realized in that while going through these digital gyrations and utilizing digital tools to amplify the signal as much as possible both during mixing and during mastering the peak value of the sample points is closely watched to ensure that it does not get to full scale Since the peak meters in said DAW and digital mixing systems are inaccu rate and do not actually indicate the peak values of the resulting waveform the result is that while the samples themselves do not exceed full scale and are carefully monitored to ensure this the resulting waveforms represented by the samples may exceed full scale throughout any standard CD While the digital mixing system is not clipping the music or distorting the music the digital to analog converters
29. mat only output meters are displayed by default You can toggle the meter display to show only in put meters by clicking the blue green rectangle at the lower right of the meter display A red clip indicator appears at the top of each me ter Clicking a clip indicator clears it Alt clicking Windows or Option clicking Mac clears the clip indicators on all channels Using the Impact Compressor Compressors reduce the dynamic range of audio signals that exceed a user selectable threshold by a specific amount This is accomplished by reducing output levels as input levels increase above the threshold The amount of output level reduction that Impact applies as input levels increase is referred to as the compression ratio This control is adjustable in dis crete increments If you set the compression ratio to 2 1 for example for each 2 dB that the signal ex ceeds the threshold the output level will be re duced to 1 dB above the threshold With a com pression ratio of 4 1 an 8 dB increase in input will produce only a 2 dB increase in output Side Chain Processing Compressors generally use the detected amplitude of their input signal as a control source However you can also use other signals such as a separate reference track or an external audio signal as a con trol source This is known as side chain processing Side chain processing lets you control Impact com pression using an independent audio signal In this w
30. tent with the RMS Root Mean Square value or the effective average level of the selected clip Normalizing Multiple Clips When multiple clips are selected the Normalize plug in can search for peaks in two different modes Clip by Clip Searches for the peak level on a clip by clip basis Entire Selection Searches for the peak level of all clips in the selection To normalize an audio clip or selection 1 Select the audio you want to normalize 2 Choose AudioSuite gt Other gt Normalize 3 Adjust the Level slider 4 Ensure that Use In Playlist is enabled 5 Click Render Reverse The Reverse plug in replaces the audio with a re versed version of the selection This is useful for creating reverse envelope effects for foley special effects or musical effects To reverse an audio clip or selection 1 Select the clip you want to reverse 2 Choose AudioSuite gt Other gt Reverse 3 Ensure that Use In Playlist is enabled 4 Click Render Reverse Audio Plug Ins Guide 350 Numerics 1 2 Band EQ Focusrite D2 26 1 Band EQ EQ III 12 4 Band EQ Focusrite D2 26 660 Fairchild Limiter 75 670 Fairchild Limiter 77 6 Band EQ Focusrite D2 26 7 Band EQ EQ III 13 A AAX Avid Audio Extension plug ins 2 AudioSuite Input Mode selector 142 Preview 142 processing preferences 142 AudioSuite plug ins DC Offset Removal 346 Duplicate 347 Gain 347 Invert 348 Normalize 348 Reverse 349
31. 316 Session Tempo and Meter Changes and ReWire 316 Looping Playback with ReWire 317 Automating Input Switching with ReWire 317 Contents xii Part XII Other Plug Ins Chapter 55 InTune 319 InTune Controls and Displays 320 InTune Presets 322 Using InTune 323 Chapter 56 MasterMeter 325 MasterMeter Overview 326 Using MasterMeter 329 MasterMeter Controls and Displays 330 Chapter 57 Signal Generator
32. As you move the control up harmonic content in creases proportionally as it works in conjunction with the Timbre control Moreover the amount of harmonics generated is dependent on the input level The gain of the harmonics automatically in creases as the input level increases The generated harmonics are not products of har monic distortion since they are intelligently pro duced and formed into a power envelope that en hances rather than distorts the final audio signal Null Fill control TYPE III NULL FILLRANGE RESPONSE dB FREQUENCY Hz 7 0 5 0 3 0 1 0 1 0 20 0 100 0 1000 0 10000 0 PEAKING at midpoint for all curves NULL FILL at MAX NULL FILL at Midpoint NULL FILL at MIN Chapter 42 Aphex Aural Exciter Type III 245 Timbre Control The Timbre control sets the order or type of har monic signal being generated by way of the Har monics control The control can be varied from all Even harmonics in the Min position to all Odd har monics at the Max position Odd order harmonics will sound softer to your ear while even harmonics will sound harsher Varying the Timbre control between the two ex tremes will provide you with a mix of both Even and Odd harmonics in proportion to the control po sition To emphasize the effect of the Timbre con trol set the Density switch to High The Timbre control display ranges from 100 all Odd to 100 all Even Mix Control The Mix
33. Chapter 5 Focusrite D2 26 Chapter 5 Focusrite D2 Focusrite D2 is a high quality digital equalizer plug in for Pro Tools Developed in cooperation with Focusrite the D2 is based on the highly ac claimed Red Range 2 dual EQ designed by Ru pert Neve It provides up to six simultaneous bands of EQ including high pass low shelf low mid peak high mid peak high shelf and low pass fil ters D2 includes a highly accurate Cartesian graph that displays EQ curves in real time as EQ controls are adjusted D2 is available in DSP Native and AudioSuite formats D2 supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates D2 operates as a mono multi mono or stereo plug in D2 Configurations There are three configurations of the Focusrite D2 plug in 1 2 Band EQ D2 1 2 Band can use up to two filters simultane ously depending on which you enable The high pass low shelf and low pass filters each utilize the entire module and cannot be used in combination with another filter The low mid peak high mid peak or high shelf filters can be used in combina tion with each other up to two bands total 4 Band EQ D2 4 Band can use up to four filters simultane ously Any combination of filters can be engaged up to a total of four bands 6 Band EQ D2 6 Band can use up to six filters simultaneously Any combination of filters can be engaged up to a total of six bands By default the lo
34. Pro Limiter provides a Dim Input Meter option that dims the input meters while highlighting the atten uation meters This lets you visually focus more on the gain reduction applied rather than on the incom ing signal levels To toggle the Dim Input Meter option on or off Click the Dim Input Meter toggle in the upper left corner of the Input section Input and Gain Reduction Peak Hold Displays Pro Limiter displays the Input Sample Peak Hold value in dB at the upper left corner of the Input meters and the Gain Reduction Peak Hold value in dB at the upper right corner of the Input meters Input Sample Peak Hold Display Shows the last greatest sample peak value from the input signal on any channel Gain Reduction Peak Hold Display Shows the last greatest attenuation value applied to the input signal To reset either the Input or Gain Reduction Peak Hold display value Click the display value you want to reset Input Meters The Input meters show peak signal levels before processing The Input meter scaling is shown on the left side of the Input meters from 90 dB to 6 dB Gain Reduction Meters The Gain Reduction meters are interleaved with the Input meters Gain Reduction meters show the amount of gain reduction applied to the Input sig nal The gain reduction amount varies depending on the level of the Input signal as well as the Threshold and Character settings The Gain Reduction meter scali
35. Rescan for Files Forces Space to check the hard drive for new IRs This is typically required if new IR files have been copied to the hard drive Using the Rescan for Files command loads new IRs into Space without needing to close Space or the Pro Tools session Installing Space IR Packages Additional IR packages for Space are available for registered users to download from the Space On line IR Library at www avid com tlspace impulselibrary These package files are supplied in a lossless com pressed format To install a Space IR package 1 In the Space IR browser select Download IR Package from the Edit menu Your default Web browser launches and loads the Avid Space On line IR Library website 2 Click Download 3 Log in using your email address and password You may need to create a new account if you have not yet registered Space 4 Click Continue 5 Click Download for the IR package you want 6 In Space select Install Space IR Package from the Edit menu Space may pause briefly while it scans the hard drives to locate IRs or if all folders are opened at once The amount of time taken is proportional to the number of folders and IRs scanned To download IR packages from the Space Online IR Library you must first register with Avid and create an online profile Chapter 30 Space 197 7 In the resulting dialog locate and select the file you downloaded 8 Click Choose 9 Click Install to install the
36. Section Controls The Room Coloration controls work in conjunction with the selected Room Type Coloration takes the characteristic resonant frequencies or EQ traits of the room and allows you to apply this spectral shape to the reverb In addition to letting you adjust the overall sound of the room the high frequency and low frequency components are split to allow you to emphasize or de emphasize the low and high frequency response of the room Coloration Control Coloration adjusts how much of the EQ character istics of the selected Room Type are applied to the original signal The range of this control is from 0 to 200 A setting of 100 provides the optimum coloration for the room type Settings above 100 will tend to produce extreme and unnatural color ation Setting Spread to 100 produces widely spaced early reflections that may sound unnatural At 100 the early reflections have no spread at all and are heard as a single reflection Chapter 29 ReVibe II 173 High Frequency Color Control High Frequency Color HF Color adds or subtracts additional high frequency coloration or relative brightness to the acoustic model of the room The range of this control is from 50 0 to 50 0 Low Frequency Color Control Low Frequency Color adds or subtracts additional low frequency coloration or relative darkness to the acoustic model of the room The range of this control is from 50 0 to 50 0 ReVibe II
37. Slap Cathedral Plates Large Natural Plate Large Bright Plate Large Synthetic Plate Medium Natural Plate Medium Bright Plate Small Natural Plate Small Bright Plate Springs Guitar Amp Spring 1 Guitar Amp Spring 2 Guitar Amp Spring 3 Chapter 29 ReVibe II 179 Guitar Amp Spring 4 Guitar Amp Spring 5 Guitar Amp Spring 6 Studio Spring 1 Studio Spring 2 Studio Spring 3 Studio Spring 4 Dense Spring 1 Dense Spring 2 Resonant Spring Funky Spring 1 Funky Spring 2 Funky Spring 3 Funky Spring 4 Chambers Large Chamber 1 Large Chamber 2 Large Chamber 3 Large Chamber 4 Large Chamber 5 Large Chamber 6 Medium Chamber 1 Medium Chamber 2 Medium Chamber 3 Medium Chamber 4 Medium Chamber 5 Small Chamber 1 Small Chamber 2 Small Chamber 3 Small Chamber 4 Ambience Large Ambience 1 Large Ambience 2 Large Ambience 3 Large Ambience 4 Medium Ambience 1 Medium Ambience 2 Medium Ambience 3 Medium Ambience 4 Medium Ambience 5 Small Ambience 1 Small Ambience 2 Small Ambience 3 Very Small Ambience Film and Post Medium Kitchen Small Kitchen Bathroom 1 Bathroom 2 Bathroom 3 Bathroom 4 Bathroom 5
38. The Input Polarity button inverts the polarity of the input signal to help compensate for phase anoma lies occurring in multi microphone environments or because of mis wired balanced connections Input and Output Meters 7 Band EQ Only The plasma style Input and Output meters show peak signal levels before and after EQ processing and indicate them as follows Green Indicates nominal levels Yellow Indicates pre clipping levels starting at 6 dB below full scale Red Indicates full scale levels clipping When using the stereo version of EQ III the Input and Output meters display the sum of the left and right channels The Clip indicators at the far right of each meter in dicate clipping at the input or output stage of the plug in Clip indicators can be cleared by clicking the indicator I O controls and meters for 7 Band EQ top and 1 Band EQ bottom Input Output Gain Input and Output Meters Gain Input Polarity Clip Indicators Control Control Control Chapter 4 EQ III 16 EQ III EQ Band Controls Individual EQ bands on each EQ III configuration have a combination of controls EQ Type Selector On the 1 Band EQ the EQ Type selector lets you choose any one of six available filter types High Pass Notch High Shelf Low Shelf Peak and Low Pass On the 7 Band EQ the HPF LPF LF and HF sec tions have EQ Type selectors to toggle between the two available filter types in each se
39. Voce Spin The Voce plug ins are available in DSP Native and AudioSuite formats The Voce plug ins support 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates The Voce plug ins support mono mono to stereo Voce Spin only multi mono and stereo opera tion Voce Chorus Vibrato Voce Chorus Vibrato recreates the mechanical scanner vibrato found in the B 3 Organ Three set tings of chorus and three settings of vibrato pre sented on one cool knob Fun and easy to use it s a classic effect used for over sixty years In a large pipe organ ranks of pipes multiple pipes designed to emit the same frequency aren t perfectly in tune The effect goes by the name multirank or more commonly chorus Inside every B3 organ on the end of the driveshaft that spins the tonewheels you ll find a mechanical contraption that delays the sound of the organ Originally added to make the B3 sound more like a pipe organ it imparts frequency variation to the sound Although well received by churches the signature B 3 Chorus Vibrato graced jazz and rock record ings ever since Now you can use this beautiful ef fect on any instrument Voce Chorus Vibrato Controls Simply click the Big Knob to rotate between set tings of Vibrato and Chorus V1 provides the least amount of vibrato V2 slightly more and V3 the most Likewise the amount of Chorus increases from C1 to C3 Option click Mac
40. a unit of measurement that expresses loudness lev els on an absolute scale LU Loudness Unit is a unit of measurement for differences between loud ness levels loudness levels on a relative scale For example program material that has a loudness level of 23 LUFS will be 2 LU quieter than pro gram material that has a loudness level 21 LUFS LU can also be used as the units for loudness levels relative to the target level Note that since K weight ing has been adopted as the standard for loudness units LUFS is equivalent with LKFS in Pro Limiter Decibels dB are an expression of the ratio of two levels the level to be described or measured and a reference level Letters after dB such as dBm signify the reference level For example dBm is referenced to 1 milliwatt whereas dBu is refer ences to 0 775Vrms LUFS is a measurement on a decibel scale and is relative to the loudness level of a stereo front left and front right 1 kHz tone peak ing at 0 dBFS 0 decibels at full scale Pro Limiter Histogram and Loudness Meters Histogram Pro Limiter provides a histogram that shows a graphic representation of loudness over time within a window of 60 seconds The graph displays True Peak levels as a yellow line and the range of loud ness over time as a blue shadow around the peak level line Time Elapsed Displays the amount of time elapsed since the current analysis pass started in hours minutes and seconds 00 00
41. and non traditional match ups Some of the amps modeled in Eleven are combo amps Combo amps have both their amp and speaker housed in the same physical box meaning there is one and only one cabinet associated with the signature sound of a combo amp The Tweed Lux and AC Hi Boost are both examples of combo amps Other amps are amps only heads and were de signed to be run through a speaker cabinet Many amp cab pairings have become standards Using Settings for Realistic and Classic Pairings You can use Eleven s factory Settings files pre sets for combo amps and classic combinations Settings files store and recall all controls includ ing Amp and Cabinet Type For combo amps and default combinations Choose a factory Settings file for that amp from Eleven s Settings menu Using the Amp Type and Cabinet Type Selectors for Unlinked Pairing You can use the Amp Type and Cabinet Type se lectors to try your own unique combinations If you want to combine amps and cabs unlinked Click and choose from the Amp Type and Cabi net Type selectors to create new pairings Cabinet Type selector in the Master section Eleven is not affiliated with or sponsored or endorsed by the makers of the loud speakers and cabinets that are emulated in the product Visit the Avid website www avid com to learn about each of the cabinets used to cre ate Eleven Use the Settings menu to save new combina
42. ian menu SoundReplacer looks for the replace ment audio files associated with that preset Sound Replacer first looks in the audio file s original hard disk location at the time you saved the setting If it is not there SoundReplacer looks in a folder named Audio Files within SoundReplacer s Root Plug In Settings folder Plug In Settings Sound Replacer Audio Files If SoundReplacer finds the replacement audio file there the Settings file will load with the associated audio By always putting replacement audio files in this special folder you can freely exchange Sound Re placer settings and the audio files associated with them with other users Using a second higher threshold for the louder kick will make it trigger properly as shown by the now properly aligned trigger marker If only one replacement sample is loaded into SoundReplacer and it is loaded into Trigger threshold amplitude zone 1 yellow Soun dReplacer will let you use the red and blue Trigger Threshold sliders to set Amplitude Zones 2 and 3 without having to load the same sample again Do not create subfolders within SoundReplacer s Audio Files folder Files located within subfolders are not recognized Chapter 59 Time Compression Expansion 343 Chapter 59 Time Compression Expansion Time Compression Expansion is an AudioSuite only time processing plug in The Time Compres sion Expansion plug in adjusts the duration of se lecte
43. information in the display area shows how many inputs and outputs an IR has For example an IR listed as 2 input 4 output is a stereo to quad IR If an IR is loaded that doesn t match the current configuration Space will try to create the best pos sible match with the IR provided For example if a stereo IR is loaded into a mono instantiation of Space Space will sum the left and right channels in order to mimic a stereo reverb with both channels panned to mono If an IR is loaded that is missing a required channel Space will automatically create a phantom channel for the IR if needed For example if a stereo IR is loaded into a quad instantiation Space will com pute left and right surround channels automatically based on the existing channels If a quad IR is loaded into a 5 0 channel instantiation Space will compute a phantom center from the front left and right channels Phantom channels are indicated by comparing the IR information displayed in the dis play area to the number of channels in use For ex ample a 2 input 4 output IR used with a 5 0 output instantiation of Space will automatically have a phantom center channel created Chapter 30 Space 190 Space Presets Space supports the Pro Tools Plug In Librarian When an IR file is loaded all controls remain at their current positions as the IR file only contains the audio waveform By default presets contain both the IR waveform and control settings and can be
44. separating transient parts of the selection from non transient parts Transient material tends to change its content quickly in time as opposed to parts of the sound which are more sustained Adjust the Threshold control or click the Threshold field and type a value The default value for Threshold is 6 0 dB For highly percussive material lower the threshold for better transient detection especially with the Rhythmic audio setting For less percussive mate rial and for shifting with the Polyphonic audio set ting a higher setting can yield better results Exper iment with this control especially when shifting drums and percussive tracks to achieve the best re sults Window The Window control sets the analysis win dow length for processing audio You can set the Window from 6 0 milliseconds to 185 0 millisec onds Adjust the Window control or click the Win dow field and type a value The Window control is only available when Polyphonic is selected as the Audio Type The default for Window size is 18 0 milliseconds and works well for many applications but you may want to try different Window settings to get the best results Try larger window sizes for low frequency sounds or sounds that do not have many transients Try smaller window sizes for drums and percus sion 37 0 milliseconds tends to work well for poly phonic instruments such as piano or guitar A set ting as large as 71 0 milliseconds works well f
45. 3 If compressing the duration of the selection at tenuate the Gain control as necessary 4 Adjust the Transient controls 5 Enable a Range button 2x 4x or 8x to set the possible range for time change 6 Adjust the Time Shift control to set the amount of time change Time change is measured in terms of the target duration using the selected timebase or as a percentage of the original speed 7 Click Render To change the pitch of a selected audio clip 1 Select AudioSuite gt Pitch Shift gt X Form 2 Select the Audio Type appropriate to the type of material you are processing Monophonic or Polyphonic 3 If transposing the pitch of the selection up atten uate the Gain control as necessary 4 Adjust the Transient controls 5 Enable a Range button 2x 4x or 8x to set the possible range for pitch change 6 Adjust the Pitch Shift control to set the amount of pitch change Pitch change is measured in semitones and cents or as a percentage of the original pitch 7 If you want to enable formant processing click the IN button to enable Formant and adjust the Formant control 8 Click Render You can adjust both the Time Shift and Pitch Shift controls independently before processing Chapter 26 X Form 153 Using X Form for Post Production Pull Up and Pull Down Tasks The table below provides information on TCE settings for common post production tasks Type the corre sponding TCE represented to 10
46. 6 dB per oc tave High Frequency Boost Section Boost mid and high frequencies using the KCS kilocycles per second and Boost knobs on the second row High Frequency Attenuate Section Cut high fre quencies using the 10k Atten knob located at the right side of the plug in Pultec MEQ 5 The Pultec MEQ 5 is the most unique equalizer in the Pultec family It is particularly useful on indi vidual tracks during mixdown Pultec MEQ 5 Controls The Pultec MEQ 5 offers three equalization sec tions low frequency boost mid frequency boost and wide range attenuation Like all Pultecs it fea tures quality transformers and a tube gain stage Low Frequency Peak Boost low frequencies 200 300 500 700 1000 Hz using the upper left con trols Mid Frequency Peak Boost mid frequencies 1 5k 2k 3k 4k 5k using the controls at the upper right Wide Range Dip Cut frequencies using Dip con trols on the bottom row Pultec EQH 2 Pultec MEQ 5 Chapter 7 Pultec Plug Ins 35 Pultec Tips and Tricks Q and A You may wonder why the Pultec EQP 1A has sep arate knobs for boost and cut The short answer is that they connect to different circuitry in the unit You can use the extra knob to your advantage Because the filters are not phase perfect a Boost setting of 3 and an Atten setting of 3 can make a huge difference even though a frequency plot wouldn t show much difference in tone You re hearing the ph
47. Aphex Systems Inc first introduced Big Bottom Pro in 1992 as part of the Model 104 Since then Big Bottom Pro has become a standard in the professional audio industry and has been used on numerous albums CDs movies broadcast productions commercials and concerts The Big Bottom Pro plug in for Pro Tools continues this tradition of success and is ready for use with the latest cutting edge music productions Big Bottom Pro provides more energy to the bass increasing its sustain and density It dynamically contours the bass response of a complex range of shapes in the 40 to 400 Hz range isolating and en hancing the lowest frequencies to provide a deeper more resonant bass Big Bottom increases the per ception of low frequencies without significantly in creasing the maximum peak output Big Bottom Pro is a single ended process which can be inserted at any point within the audio chain The input signal is split into two parts One part goes to the output unmodified while the other part known as a side chain goes through Big Bottom Pro The side chain consists of a tunable low pass filter followed by a dynamic processor Big Bottom is available in DSP Native and Audio Suite formats Big Bottom supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Big Bottom Pro operates as a mono multi mono or stereo plug in Aphex Big Bottom Pro Chapter 43 Aphex Big Bottom Pro 251 Meters Drive Mete
48. Buffer size of 512 samples is recommended for using ReWire in sessions with sample rates above 48 kHz Chapter 54 ReWire 313 Using ReWire The ReWire plug in is installed when you install Pro Tools All inter application communications between Pro Tools and ReWire client software is handled automatically To use a ReWire client application with Pro Tools 1 In Pro Tools choose Track gt New and specify one Instrument track or audio or Auxiliary In put track and click Create 2 In the Mix window click the Insert selector on the track and assign the ReWire client plug in to the track insert The ReWire client application launches automati cally in the background if the client applications supports auto launch 3 Configure the ReWire client application to play the sounds you want 4 In Pro Tools set the output of the client applica tion in the ReWire plug in window This is the audio output of the ReWire client to Pro Tools If the client application does not support auto launch launch it manually Some Re Wire client applications may need to be launched and configured before launching Pro Tools such as Cycling 74 Max MSP Others may need to be launched after Pro Tools is launched such as Ableton Live For more information refer to the manufac turer s documentation for your ReWire client application Selecting the audio output from a ReWire client Chapter 54 ReWire 314 5 In the Mix window cl
49. Channel Strip Chain Listen mode enabled Control Shift click Mac or Start Shift click Windows and hold an EQ or Filter control point in the Frequency Graph to temporarily switch to Listen mode for that EQ band or Filter effect Chapter 11 Channel Strip 47 Channel Strip Input Section The Input section provides input metering and controls for trimming the input signal and inverting its phase It can also be toggled to show post pro cessing gain reduction meters Input Trim Control The Input Trim control sets the input gain of the plug in before EQ processing letting you make up gain or prevent clipping at the plug in input stage To Trim the input signal do one of the following Click in the Input Trim field and type a Trim value 36 0 dB to 36 0 dB Click Trim and drag up or down to adjust the Input Trim setting Phase Invert The Phase Invert button at the top of the Input sec tion inverts the phase polarity of the input signal to help compensate for phase anomalies that can occur either in multi microphone environments or because of mis wired balanced connections To enable or disable phase inversion on input Click the Phase Invert button so that it is high lighted Click it again so that it is not highlighted to disable it Input Meters The Input meters show peak signal levels before processing Dark Blue Indicates nominal levels from INF to 12 dB Light Blue Indic
50. D Verb Eleven EM Euphonix EUCON EveryPhase Expander ExpertRender Fairchild FastBreak Fast Track Film Cutter FilmScribe Flexevent FluidMotion Frame Chase FXDeko HD Core HD Process HDpack Home to Hollywood HyperSPACE HyperSPACE HDCAM iKnowledge Impact Improv iNEWS iNEWS Assign iNEWS ControlAir InGame Instantwrite Instinct Intelligent Content Management Intelligent Digital Actor Technology IntelliRender Intelli Sat Intelli Sat Broadcasting Recording Manager InterFX Interplay inTONE Intraframe iS Expander iS9 iS18 iS23 iS36 ISIS IsoSync LaunchPad LeaderPlus LFX Lightning Link amp Sync ListSync LKT 200 Lo Fi MachineControl Magic Mask Make Anything Hollywood make manage move media Marquee MassivePack MassivePack Pro Maxim Mbox Media Composer MediaFlow MediaLog MediaMix Media Reader Media Recorder MEDIArray MediaServer MediaShare MetaFuze MetaSync MIDI I O Mix Rack Moviestar MultiShell NaturalMatch NewsCutter NewsView NewsVision Nitris NL3D NLP NSDOS NSWIN OMF OMF Interchange OMM OnDVD Open Media Framework Open Media Management Painterly Effects Palladiium Personal Q PET Podcast Factory PowerSwap PRE ProControl ProEncode Profiler Pro Tools Pro Tools HD Pro Tools LE Pro Tools M Powered Pro Transfer QuickPunch QuietDrive Realtime Motion Synthesis Recti Fi Reel Tape Delay Reel Tape Flanger Reel Tape Saturation Reprise Res Rocket Surfer Re
51. Dragging a control point ReVibe II Part II EQ Plug Ins Chapter 4 EQ III 12 Chapter 4 EQ III The EQ III plug in provides high quality 1 Band and 7 Band EQ for adjusting the frequency spectrum of audio material EQ III is available in DSP Native and AudioSuite formats EQ III supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates EQ III operates as a mono multi mono or stereo plug in EQ III has a Frequency Graph display that shows the response curve for the current EQ settings on a two dimensional graph of frequency and gain The frequency graph display also lets you modify fre quency gain and Q settings for individual EQ bands by dragging their corresponding points in the graph EQ III Configurations The EQ III plug in appears as two separate choices in the plug in insert selector and in the AudioSuite menu EQ3 1 Band EQ3 7 Band 1 Band EQ The 1 Band EQ has its own window with six selectable filter types for a single band of EQ 1 Band EQ Chapter 4 EQ III 13 7 Band EQ The 7 Band EQ has its own window with up to seven separate bands each with it its own set of fil ter types Adjusting EQ III Controls In addition to dragging controls and typing control values there are other ways to adjust EQ III con trols Inverting Filter Gain Peak EQ Bands Only Gain values can be inverted on any Peak EQ band by Shift clicking its control dot in the Fr
52. Dynamics Graph display also shows the threshold as an orange vertical line This control has an approximate range of 60 dB to 0 dB with a setting of 0 dB equivalent to no com pression or limiting The default value for the Threshold control is 24 dB Look Ahead control Threshold arrow on Input meter Threshold indicator on Dynamics Graph display Chapter 12 Dynamics III 67 Expander Gate III Ratio Control The Ratio control sets the amount of expansion For example if this is set to 2 1 it will lower sig nals below the threshold by one half At higher ra tio levels such as 30 1 or 40 1 the Expander Gate functions like a gate by cutting off signals that fall below the threshold As you adjust the ratio control refer to the built in graph to see how the shape of the expansion curve changes This control ranges from 1 1 no expansion to 100 1 gating Expander Gate III Attack Control The Attack control sets the attack time or the rate at which gain is reduced after the input signal crosses the threshold Use this along with the Ratio setting to control how soft the Expander s gain re duction curve is This control ranges from 10 s fastest attack time to 300 ms slowest attack time Expander Gate III Hold Control The Hold control specifies the duration in seconds or milliseconds during which the Expander Gate will stay in effect after the initial attack occurs This can be used as a function t
53. EQ Boosts or cuts a band of frequencies centered around the Frequency setting The width of the affected band is determined by the Q setting The High Shelf and High Peak Gain controls and their corresponding graph elements are displayed on screen in blue The following control values are available High Shelf EQ left and High Peak EQ right High Shelf EQ button Frequency control Band Enable button Gain control Q control High Peak EQ button Frequency control Band Enable button Gain control Q control Control Value Frequency Range 1 8 kHz to 20 kHz Frequency Default 6 kHz High Shelf Q Range 0 1 to 2 0 High Peak Q Range 0 1 to 10 0 Q Default 1 0 High Shelf Gain Range 12 dB to 12 dB High Peak Gain Range 18 dB to 18 dB Chapter 4 EQ III 25 Frequency Graph Display 7 Band EQ Only The Frequency Graph display in the 7 Band EQ shows a color coded control dot that corresponds to the color of the Gain control for each band The filter shape of each band is similarly color coded The white fre quency response curve shows the contribution of each of the enabled filters to the overall EQ curve Frequency Graph display for the 7 Band EQ High Pass control dot Low Mid control dot High Mid control dot Low Pass control dot gray brown green gray Low control dot red Mid control dot yellow High control dot blue Frequency response curve
54. Exciter operating theory As the time delay stretches transient waveforms to create a percep tion of louder sound a dip or null also occurs in the output equalization curve at the Tune fre quency As a result the null frequencies are de emphasized thus giving even more emphasis to the higher frequencies Although this often is a desir able effect the Null Fill control was created to al low the user to fill in the null by a selectable amount for any applications requiring less empha sis The following figure shows three different Null Fill settings with Tune set at the mid point position With the Null Fill control set at Min there is a no ticeable drop in the frequency response just before the start of the high pass shelf boost At this setting program material under enhancement would lose some presence When the Null Fill control is set at Max the frequency dip is filled but the frequencies associated with the shelf top become accentuated Also notice the shift in the Tune frequency 0 dB axis for the range of Null Fill settings Harmonics Control The Harmonics control adjusts the amount of har monics being generated which is displayed as a percentage underneath the controls The harmonic generator produces harmonic com ponents according to a complex set of laws includ ing considerations for transient and steady state qualities as well as relative amplitude of the origi nal audio signal
55. Hz You can set the LFO Rate control to synchronize to the current tempo of the Pro Tools session See Synchronizing Reel Tape Flanger to Session Tempo on page 230 Depth The LFO Depth control adjusts the amplitude of the change in variable delay A higher setting results in wider fluctuations in speed A lower setting results in narrower fluctuations in speed LFO Depth is ad justable from 0 to 100 percent with a default value of 65 percent Mix The Mix control adjusts the amount of fixed delay signal mixed with the variable delay signal in the fi nal output of the plug in The default Mix value is adjustable from 100 all fixed delay signal to 100 all variable delay signal percent with a de fault value of 0 50 fixed delay 50 variable de lay signals Invert Plug In Automation Playlist or Control Surface Access Only The Invert parameter inverts the polarity of the sig nal coming from the variable delay tape machine so that complete audio cancellation occurs when the flanger effect crosses the zero point The default setting for the Invert parameter is Off Operation with ADT Range setting positive offset When the LFO Depth control is set to zero you can still achieve a manual flanging or ADT effect by varying the Range control zero point LFO Depth LFO Rate Chapter 39 Reel Tape Flanger 230 This parameter is accessible only from the plug in automation playlist or from
56. IR package A window is displayed with the results of the installation The IR browser in Space updates to include the new IR If a problem occurs with the IR installation Space displays an error message Review the log file stored in the Space IR library for further details Each IR package has a version number and Space warns you if an IR package has already been in stalled The details of all installed IR packages can be re viewed using the Show Packages option in Prefer ences mode Space Primary Controls The primary control group is visible at all times and allows control of key reverb parameters This in cludes the wet and dry levels of the audio passing through Space Reset Resets all Space parameters except Wet Dry and Input and Output Level Wet Controls the level of wet or effected reverb signal from inf dB to 12 dB Dry Controls the level of dry or unaffected reverb signal from inf dB to 12 dB Decay Controls the overall decay of the IR wave form and is displayed as a percentage of the origi nal When Decay is adjusted the waveform is re calculated in real time Space primary controls Chapter 30 Space 198 Space Group Selectors and Controls Space presents reverb controls in five different groups Each group is activated by selecting the corresponding selector Space Level Controls The Levels group provides control of the overall in put and output of the reverb including indi
57. Indicates nominal levels Yellow Indicates pre clipping levels starting at 6 dB below full scale Red Indicates full scale levels clipping The Clip indicators at the top of each meter indicate clipping at the input or output stage of the plug in Clip indicators can be cleared by clicking the indi cator De Esser III The De Esser has no control to directly ad just the threshold level the level that an input signal must exceed to trigger de essing The amount of de essing will vary with the input signal De Esser III I O Meter display Gain Reduction meter Input meter Output meter Chapter 12 Dynamics III 69 De Esser III Gain Reduction Meter The Gain Reduction meter indicates the amount the input signal is attenuated in dB This meter shows different colors during de essing Light Orange Indicates that gain reduction is being applied but has not reached the maximum level set by the Range control Dark Orange Indicates that gain reduction has reached the maximum level set by the Range con trol De Esser III Frequency Control The Frequency Freq control sets the frequency band in which the De Esser operates When HF Only is disabled gain is reduced in frequencies within the specified range When HF Only is en abled the gain of frequencies above the specified value will be reduced This control ranges from 500 Hz lowest fre quency to 16 kHz highest frequency De Esser III Range
58. LFE is selected the plug in uses the amplitude of a separate reference track or exter nal audio source to trigger dynamics processing The reference track used is selected using the Plug In Key Input selector in the Plug In window header With greater than stereo multichannel pro cessing the key signal triggers dynamics process ing for all processed audio channels equally Internal All No LFE When Int All no LFE is selected dynamics pro cessing is applied equally to all channels when the input signal reaches the threshold on any input channel except for the LFE channel if present The LFE channel is processed independently based on its own input signal Internal Front Rear For LCRS or greater channel formats when Int Front Rear is selected dynamics processing is ap plied based on front channel inputs LCR and sur round channel inputs S independently For 1 for mats the LFE channel is processed independently based on its own input signal Side Chain Listen Mode Listen mode lets you hear the input signal for the side chain to the compressor This can be either the external key input or the internal side chain in cluding the applied filter To enable or disable Listen mode on the side chain Click the Listen button in the top right corner of the Side Chain section so that it is highlighted The icon flashes while Side Chain Listen mode is enabled To disable it click the button again so that
59. Meter option is enabled the Click Beat Value options grayed out To set the Click Beat Value independently of the Meter track 1 Ensure that the Follow Meter option is disabled 2 Click to select a rhythmic value for the down beat click whole half quarter eighth or six teenth note 3 Click to select or deselect the triplet or dot modifier for the beat value Click 2 The Click 2 section provides the same controls as the Click 1 section but for all beats other than the downbeat The Click Beat Value options can be auto mated to support the appropriate accent pat terns for different meters for the click Chapter 53 Click II 309 Creating a Click Track To create a click track with the Click II plug in 1 Ensure that the Options gt Click is enabled 2 Choose Track gt Create Click Track Pro Tools creates a new Auxiliary Input track named Click with the Click II plug in already in serted In the Edit window the track s Track Height is set to Mini To manually create a click track with the Click plug in 1 Select Options gt Click to enable the Click option or enable the Metronome button in the Trans port 2 Create new a mono Auxiliary Input track and in sert the Click II plug in 3 Select a click sound preset 4 Choose Setup gt Click Countoff and set the Click and Countoff options 5 Begin playback A click is generated according to the tempo and meter of the current sess
60. Moogerfooger 12 Stage Phaser offers 6 or 12 stages of MOOG resonant analog filters Unlike the Lowpass Filter however the filters are arranged in an allpass configuration A phaser works by sweeping the mid shift fre quency of the filters back and forth As this hap pens the entire frequency response of the output moves back and forth as well The result is the clas sic phaser whooshing sound as different fre quency bands of the signal are alternately empha sized and then attenuated Moogerfooger 12 stage Phaser Different types of filters Frequency Frequency Frequency Cutoff Center Mid Shift Low Pass Filter Resonant Filter 5 Stage Phaser Time Time Time 1 1 1 Chapter 37 Moogerfooger 12 Stage Phaser 223 A sweep control allows you to adjust the range of the frequency shift And keeping in the spirit of the MOOG modular synthesizers an integrated LFO allows you to modulate the sweep control allowing for extreme effects Moogerfooger 12 Stage Phaser Controls LFO Section Control the LFO using the Amount and Rate knobs and the Lo Hi selector switch Amount Amount varies the depth of phaser modu lation from barely perceptible at the full counter clockwise position to the full sweep range of the phaser full clockwise or Kill setting Rate Rate determines how fast the LFO oscillates The LFO light blinks to give a visual indication of the LFO rate Lo Hi T
61. Rate Rate determines how fast the LFO oscillates from 0 1 Hz one cycle every ten seconds to 25 Hz twenty five cycles per second The LFO light blinks to give a visual indication of the LFO rate Sine Square The Square Sine switch selects either a square or sine waveform The square wave pro duces trill effects whereas the sine waveform pro duces vibrato and siren effects Modulator Section The Carrier Oscillator is controlled by the Fre quency knob and the Low High switch Frequency Knob Operating at the selected fre quency the carrier oscillator provides one input to the ring modulator with the other coming from the input signal Lo In the Lo position the Frequency knob ranges from 0 5 Hz to 80 Hz Hi In the High position the Frequency knob ranges from 30 Hz to 4 kHz Mix The Mix control blends the input signal and the Ring Modulator output You hear only the input signal when the knob is counterclockwise and only the ring modulated signal with the knob fully clockwise Drive The Drive control sets the input gain LED Indicators Three LEDs provide visual feedback Level Level glows green when signal is present LFO LFO blinks to show the LFO rate Bypass Bypass glows either red bypassed or green not bypassed to show whether or not the ef fect is in the signal path Moogerfooger Ring Modulator Tips and Tricks You ll discover tons of great uses for the Mooger fooger Ring Modulator throu
62. Record enable the audio track Audio track for recording dry while hearing Eleven Eleven Guitar input Chapter 44 Eleven 272 5 Make sure you are not overloading your input signal by checking levels in all tracks and Eleven s Input LED 6 When you re ready arm Pro Tools and begin re cording The output from Eleven is recorded to disk If you need to conserve DSP or Native processing re sources you can remove or deactivate Eleven after recording Recording Dry and Eleven Simultaneously You can record a dry unprocessed track and an Eleven processed track simultaneously This method gets the best of both worlds by track ing dry to experiment with tones later and at the same time recording the tone used on the original tracking session It requires two audio track as fol lows To record guitar dry and with Eleven live 1 Choose Track gt New 2 Configure the New Tracks dialog to create two mono audio tracks then click Create 3 In the Mix or Edit window configure the first left most new audio track by doing the follow ing Click the Input selector and choose your guitar input the audio interface input your guitar is plugged in to Click the Output selector and choose Bus 1 Click the Insert selector and select Eleven Record enable the audio track 4 Configure the second audio track by doing the following Click the Input selector and choose Bus 1 Recor
63. Sample Size The Sample Size slider controls the bit resolution of the audio Like sample rate bit resolution affects audio quality and clarity The lower the bit resolu tion the grungier the quality The range of this con trol is from 24 bits to 2 bits Quantization Lo Fi applies quantization to impose the selected bit size on the target audio signal The type of quan tization performed can also affect the character of an audio signal Lo Fi provides you with a choice of Linear or Adaptive quantization Linear Linear quantization abruptly cuts off sample data bits in an effort to fit the audio into the selected bit resolution This imparts a characteristically raunchy sound to the audio that becomes more pro nounced as the sample size is reduced At extreme low bit resolution settings linear quantization will actually cause abrupt cut offs in the signal itself similar to gating Thus linear resolution can be used creatively to add random percussive rhythmic effects to the audio signal when it falls to lower lev els and a grungy quality as the audio reaches mid levels Adaptive Adaptive quantization reduces bit depth by adapting to changes in level by tracking and shifting the amplitude range of the signal This shifting causes the signal to fit into the lower bit range The result is a higher apparent bit resolution with a raunchiness that differs from the harsher quantization scheme used in linear resolution Noise Gen
64. SansAmp s FET hybrid circuitry emulation cap tures the low order harmonics and sweet overdrive unique to tube amplifiers And pushed harder SansAmp also generates cool lo fi and grainy sound textures that still retain warmth SansAmp also features a proprietary speaker simu lator which emulates the smooth even response of a multiple miked speaker cabinet free of the harsh peaks valleys and notches associated with single miking or poor microphone placement Finally SansAmp provides two extremely sweet sounding tone controls high and low that sound great on most anything SansAmp PSA 1 Chapter 48 SansAmp PSA 1 290 PSA 1 Controls Use the eight knobs to dial in your tone or effect Pre Amp Determines the input sensitivity and pre amp dis tortion Increasing the setting produces an effect similar to putting a clean booster pedal ahead of a tube amp overdriving the first stage For cleaner sounds use settings below the unity gain point Buzz Controls low frequency break up and overdrive Boost the effect by turning clockwise from the cen ter point indicated by the arrows As you increase towards maximum the sound becomes you guessed it buzzy with added harmonic content For increased clarity and definition when using dis tortion position the knob at its midpoint or towards minimum Punch Sets midrange break up and overdrive Decreasing from the center produces a softer Fender style break up
65. Start Shift Win dows and drag any rotary control or control point horizontally or vertically When monitoring in Band Pass mode the Fre quency and Q controls function differently Frequency Sets the frequency above and below which other frequencies are cut off leaving a nar row band of mid range frequencies Q Sets the width of the narrow band of mid range frequencies centered around the Frequency setting To switch an EQ III control out of Band Pass mode Release Control Shift Mac or Start Shift Windows Controlling EQ III from a Control Surface EQ III can be controlled from any supported con trol surface including EUCON compatible control surfaces D Control D Command C 24 and 003 Refer to the guide that came with the control sur face for details Band Pass mode does not affect EQ III Gain controls EQ III interactive graph displaying Band Pass mode Chapter 4 EQ III 15 EQ III I O Controls Certain Input and Output controls are found on all EQ III configurations except where noted other wise Input Gain Control The Input Gain control sets the input gain of the plug in before EQ processing letting you make up gain or prevent clipping at the plug in input stage Output Gain Control 7 Band EQ Only The Output Gain control sets the output gain after EQ processing letting you make up gain or prevent clipping on the channel where the plug in is being used Input Polarity Control
66. The Release control sets how long it takes for the gate to close after the input signal falls below the threshold level and the hold time has passed Knee The Knee control sets the rate at which the Ex pander Gate reaches full effect once the threshold has been exceeded Hysteresis The Hysteresis Hyst control lets you adjust whether or not the gate rapidly opens and closes when the input signal is fluctuating near the Threshold This can help prevent undesirably rapid gating of the signal This control is only available when Ratio is set to Gate otherwise it is greyed out Dynamics section Expander Gate tab Chapter 11 Channel Strip 52 Compressor Limiter Controls Threshold The Threshold control sets the level that an input signal must exceed to trigger compression or limit ing Signals that exceed this level will be com pressed Signals that are below it will be unaf fected Attack The Attack control sets the attack time or the rate at which gain is reduced after the input signal crosses the threshold The smaller the value the faster the attack The faster the attack the more rapidly the Compres sor Limiter applies attenuation to the signal If you use fast attack times you should generally use a proportionally longer release time particularly with material that contains many peaks in close proximity Ratio The Ratio control sets the compression ratio or the amount of compression applied as the
67. TimeAdjuster Controls 215 Using TimeAdjuster for Manual Delay Compensation 216 When to Compensate for Delays 217 Part VII Modulation Plug Ins Chapter 36 Moogerfooger Lowpass Filter 219 Moogerfooger Lowpass Filter Controls 220 Moogerfooger Lowpass Filter Tips and Tricks 221 Chapter 37 Moogerfooger 12 Stage Phaser 222 Moogerfooger 12 Stage Phaser Controls 223 Moogerfooger 12 Stage Phaser Tips and Tricks 224 Chapter 38 Moogerfooger Ring Modulator 225 Moogerfooger Ring Modulator Controls 226 Moogerfooger Ring Modulator Tips and Tricks 226 Chapter 39 Reel Tape Flanger
68. To use the TimeAdjuster plug in to flip phase when blending amps or cabinets 1 Configure your audio track and Aux Inputs as instructed in Blending Eleven Cabinets and Amps on page 274 Make sure each Aux Input has an Eleven plug in followed by a TimeAd juster plug in 2 Open the plug in window for each of the Time Adjuster plug ins click the first one to open it then Shift click each of the other TimeAdjuster plug ins 3 Click the Phase switch in the first TimeAdjuster plug in to invert the polarity Listen to the effect it has on the combined signal Click it again to disengage flip back 4 Click the Phase switch on the next channel s TimeAdjuster plug in listen then disengage 5 Repeat for additional Eleven TimeAdjuster channels 6 Try combinations of flipped and non flipped Phase settings to find the ideal relationship for the currently blended amps and cabinets Chapter 44 Eleven 278 Tweaking Phase If each of the mics used on a single cabinet are not positioned carefully comb filtering and other fre quency anomalies can occur With real amps the engineer moves one or more mics to find their op timal positions relative to the source and to each other To hear the effect of small adjustments to the phase relationships of signals do the following To use the Time Adjuster plug in to control phase 1 Configure your audio track and Aux Inputs as instructed in Blending Eleven Cabinets and A
69. Type selector and select the au tomation control you just enabled from the Voce Spin sub menu 5 Edit the breakpoint automation for the enabled control Accessing Voce Spin Controls on a Control Surface When using a control surface all Voce Spin con trols are available whenever the plug in is focused You only need to enable plug in automation if you want to record your adjustments as breakpoint au tomation To access additional Voce controls from a control surface 1 Focus the Voce Spin plug in on your control sur face All available controls are mapped to en coders faders and switches 2 Adjust the corresponding control 3 Use the previous next Page controls to access additional controls To automate your adjustments enable automation for that control For more in formation on plug in automation see the Pro Tools Reference Guide Chapter 41 Voce Plug Ins 239 Voce Spin Tips and Tricks The One Mic Way Back In The Corner Of The Room Trick Spin isn t designed to sound like a rotating speaker spinning all by itself in a large room Spin provides the sound of a miked rotating speaker the sound the producer and engineer hear in the control room But don t let that stop you from getting the sound you want To achieve the sound of a distant microphone cap turing the rotating speaker run Spin using the wide stereo preset Now apply a room reverb remove any pre delay and adjust the wet dry reve
70. VU indicating 18 dBFS BF 3A Tips and Tricks AudioSuite Processing When using the AudioSuite version of the BF 3A be sure to select an auxillary side chain input nor mally the track you are processing The default is None and if you leave it set like this there is nothing feeding the detector and you will not hear any compression action Line Amp Turn the Peak Reduction knob full counterclock wise off and use the Gain control to increase the signal level Although the BF 3A does not com press the sound with these settings it still adds its unique character to the tone Chapter 10 BF76 42 Chapter 10 BF76 BF76 is a vintage style compressor plug in that is available in DSP Native and AudioSuite formats BF76 supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates BF76 operates as a mono multi mono or stereo plug in Modeled after the solid state 1176 studio compres sor BF76 preserves every sonic subtlety of this classic piece of studio gear The 1176 Compressor originally introduced in the late 1970s uses a FET field effect transistor The 1176 also uses solid state amplification The 1176 still provides an extremely high quality audio sig nal path but because of these internal differences offers a much different compression sound than other compressors Four selectable compression ratios are provided along with controls allowing variable attack and re leas
71. Voce Spin Additional Controls on page 237 Chapter 41 Voce Plug Ins 238 Lower Slow Speed Lower Accel Rate Lower Decel Rate Lower Mic Angle You can adjust and automate controls such as input trim from 24 dB to 24 dB set the rotor balance the mix between the upper and lower speakers specify acceleration and deceleration times in sec onds for both the upper and lower speakers tweak the fast and slow speeds of each speaker and spec ify the microphone angle for each stereo pair of mi crophones You can access these additional controls through Pro Tools plug in automation and or from a com patible control surface Accessing Voce Spin Controls Accessing Voce Spin Controls On Screen All Voce Spin controls can be adjusted on screen by editing Pro Tools breakpoint automation data To access additional Voce Spin controls on screen 1 Click the Plug In Automation button in the Plug In window to open the Plug In Automation window 2 In the list of controls at the left click to select a control and click Add or just double click a control in the list Repeat to access and enable additional controls 3 Click OK to close the Plug In Automation win dow 4 In the Edit window do one of the following Click the Track View selector and select the au tomation control you just enabled from the Voce Spin sub menu Reveal an Automation lane for the track click the Automation
72. Windows or Option click Mac anywhere to toggle between Chorale and Tremolo speeds Spin Presets 122 Model 122 speaker medium pulleys 122 Small Pulley Small pulleys fastest rotation 122 Large Pulley Large pulleys slowest rotation 122 Wide Stereo Middle pulleys wide stereo mi crophone placement 122 Mono Middle pulleys one mic each top and bottom 21H Model 21H speaker Foam Drum Middle pulleys microphones close Memphis Lower drum slow motor unplugged mi crophones close Steppenwolf Lower drum only loose belts micro phones close Rover Slow to Fast Guitar rotating speaker max imum speed differential Rover Slow to Medium Guitar rotating speaker slower variation Rover Medium to Fast Guitar rotating speaker faster variation Phaser Medium rotation rate microphones very close Watery Guitar Fast rotation rate microphones close Speed Options Chorale Slow rotation Tremolo Fast rotation Off No rotation but still through the crossover and speakers wherever the speakers comes to rest rela tive to the microphones Voce Spin Additional Controls Though the Voce Spin plug in window contains only the Chorale Off Tremolo control the follow ing controls are also available Input Trim Speed Switch Rotor Balance Upper Slow Speed Upper Accel Rate Upper Decel Rate Upper Mic Angle Lower Fast Speed See also
73. a 12 note chro matic scale The note entry fields are disabled when Chromatic Mode is selected Single Octave Mode When selected Single Octave Mode disables the display of octave information with each note on the main InTune screen When tuning in this mode In Tune ignores the octave of the note being tuned The octave information entered in the Edit screen is used only for generating test tones Selecting a InTune preset For more information on using plug in presets in Pro Tools see the Pro Tools Reference Guide Tuner Programming Chapter 55 InTune 323 Single Octave Mode is typically used for instru ments which generate harmonics in multiple oc taves such as bass guitars Because of the low fre quency waveform generated by a bass guitar it is easier for InTune to tune to a higher harmonic of the note instead Display Flat Semitones InTune will display all semitones entered into note fields as sharp by default For example a guitar tuned to E flat is usually represented by the follow ing Eb2 Ab2 Db3 Gb3 Bb3 Eb4 By default if these notes are entered in the Edit screen InTune will display these same notes in the following way D 2 G 2 C 3 F 3 A 3 D 4 The Display Flat Semitones option overrides the default behavior and displays semitones as flats not sharps It is not possible to display both sharp and flat semitones in the same tuning preset Note Entry Fields The twelve note entry
74. a supported control surface Noise Plug In Automation Playlist or Control Surface Access Only The Noise parameter controls the level of simu lated tape hiss that is added to the processed signal Noise is adjustable from Off INF to 24 dB with a default value of Off This parameter is accessible only from the plug in automation playlist or from a supported control surface Synchronizing Reel Tape Flanger to Session Tempo You can set the LFO Rate in Reel Tape Flanger to synchronize to the session tempo in beats per min ute To synchronize the LFO Rate control setting to the session tempo 1 In the BPM Sync section click the On button The Tempo Rate display changes to synchronize with the current session tempo 2 To set a rhythmic LFO rate click the Note Value to choose from the available note values whole half quarter eighth sixteenth or thirty second note 3 To adjust the rhythm further do any of the fol lowing To enable triplet rhythm delay timing click the Triplet 3 button so that it is lit To set a dotted rhythm delay value click the Dot button so that it is lit Settings for this parameter are saved with plug in presets If you use a preset for the DSP Native or AudioSuite version of this plug in any settings for this parameter will be active Settings for this parameter are saved with plug in presets If you use a preset for the DSP Native or Audi
75. adjust high frequency cut or damp Drag the HF Cut HF Damp control point right or left Reverb EQ Graph You can use this 3 band equalizer to shape the tonal spectrum of the reverb The EQ is post reverb and affects both the reverb tail and the early reflections Frequency Sliders Sets the frequency boundaries between the low mid and high band ranges of the EQ The low frequency slider 60 0 Hz 22 5 kHz sets the frequency boundary between low and mid cut boost points in the EQ The high frequency slider 64 0 Hz 24 0 kHz sets the frequency boundary between the mid and high cut boost points in the EQ Band Breakpoints Control cut and boost values for the low mid and high frequencies of the EQ To cut a frequency band drag a breakpoint downward To boost drag upward The adjustable range is from 24 0 dB to 12 0 dB HF Cut Breakpoint Sets the frequency above which a 6 dB octave low pass filter attenuates the processed signal It removes both early reflections and reverb tails affecting the overall high fre quency content of the reverb Use the HF Cut con trol to roll off high frequencies and create more nat ural sounding reverberation The adjustable range is from 120 0 Hz to 24 0 kHz Reverb One EQ graph controls Reverb One AAX version EQ icon selected Adjusting graph controls HF Cut HF Damp Band Cut Boost Frequency Crossover Frequency Crossover Chapter 28 Reverb One 165 Reverb Color
76. aliasing For this reason digital recording has a maximum level at which signals can be recorded Anything exceeding this level full scale has unde sirable consequences The method used for computing the peak value in side the system however is not particularly accu rate DAW systems typically take the amplitude of the samples and use these as the basis for the peak meter The problem with this approach is easily identified the samples themselves do not represent the peak value of the waveform The waveform is only complete after the reconstruction process Un Chapter 56 MasterMeter 328 til this process has been completed the waveform is inaccurately represented by the samples This is the reason that in most DAWs the waveform is rep resented on the screen as a dot to dot connection between sample points They do not undergo the reconstruction process inside the system so all that can be represented is the sample points and for the sake of visual ease they connect the dots between them with straight lines They save the reconstruc tion process for the digital to analog converters The consequence of the way in which DAWs treat waveforms is that the meter inside the DAW or other digital mixers inevitably shows inaccurate in formation It is virtually a mathematical certainty that the waveform will exceed the amplitude of the samples in any sampling system The samples themselves only represent a waveform It is import
77. and signal routing as explained in the previous workflow see To blend multiple cabinets on page 274 2 Remove or simply bypass the Eleven plug in on the source input track 3 Solo the first Aux Input track 4 Click to open the Eleven plug in window on the soloed Aux Input and do any of the following Make sure the amp and cabinet are active not bypassed Choose a preset Settings file Pair any amp with any cabinet Choose a mic and its position Adjust Speaker Breakup 5 Solo the next Aux Input track and repeat to con figure its settings for a different tone To maximize processing resources remove the Eleven plug in on the source track or make the plug in Inactive See the Pro Tools Reference Guide for more information Setup for blending amps No insert Amps and Cabs on Chapter 44 Eleven 277 6 Repeat for other Aux Input tracks to configure their settings 7 When you have set your tones make sure to un solo all the Aux Inputs 8 Continue playing so you can hear the combined tone of all the amps 9 Do the following to continue Balance the tracks using the volume faders on the Aux Input tracks Try different pan positions for each Auxiliary Input track 10 Evaluate the phase relationships of the com bined signals and adjust accordingly see Phase Considerations with Blending in Eleven on page 277 Phase Considerations with Blending in E
78. and stereo formats only In addition to standard controls in each module Dynamics III also provides a graph to track the gain transfer curve in the Compressor Limiter and Ex pander Gate plug ins and a frequency graph to dis play which frequencies trigger the De Esser and which frequencies will be gain reduced Dynamics III Common Controls The Levels section the LFE Enable button and the Dynamics Graph display of the user interface are shared between the Compressor Limiter Ex pander Gate and De Esser plug ins Dynamics III Levels Section The indicators and controls in the Dynamic III Lev els section let you track input output and gain re duction levels as well as work with phase invert and the threshold setting Greater than stereo formats are only available with Pro Tools HD See De Esser III Level Meters on page 68 for more information on De Esser III In put Output Level controls I O Meter display stereo instance shown Gain Reduction meter Threshold arrow Phase Invert Input meter Output meter Peak hold indicators Peak hold indicators Chapter 12 Dynamics III 60 Input and Output Meters The Input In and Output Out meters show peak signal levels before and after dynamics processing Green Indicates nominal levels Yellow Indicates pre clipping levels starting at 6 dB below full scale Red Indicates full scale levels clipping The clip indicators at the t
79. are the same digital sound re peated over and over whereas a bucket brigade de vice itself imparts a warm organically evolving timbre to the echoes Avid s digital replica re creates all the warm natu ral sounds of its analog counterpart The Mooger fooger Analog Delay plug in was enhanced to be even more useful for digital recording An inte grated Highpass Filter allows you to remove un Moogerfooger Analog Delay Chapter 32 Moogerfooger Analog Delay 208 wanted bass buildup from the feedback loop al lowing you to have warmer more controllable echo swarms while minimizing the potential for digital clipping Moogerfooger Analog Delay Controls The Moogerfooger Analog Delay provides the fol lowing controls Delay Time Delay Time allows you to select the length of delay between the original and the de layed signal Used with Feedback it also affects how long apart the echoes are Short Long The Short Long switch sets the range of the Delay Time control Set to Short the Delay Time ranges from 0 04 to 0 4 seconds Set to Long it ranges from 0 08 to 0 8 seconds Feedback Feedback determines how much signal is fed back to the delay input affecting how fast the echoes die out Highpass The Highpass knob removes low fre quencies from the feedback loop It removes unde sirable low frequency mud common when mix ing with delays and also allows the creation of amazing echo swarms that won t clip the out
80. as 1 8 1 16 or 1 32 notes Chapter 28 Reverb One 163 ER Settings Selects an early reflection preset These range from realistic rooms to unusual reflective effects The last five presets Plate Build Spread Slapback and Echo feature a nonlinear response Early reflection presets include Room Simulates the center of a small room without many reflections Club Simulates a small clear natural sounding club ambience Stage Simulates a stage in a medium sized hall Theater Simulates a bright medium sized hall Garage Simulates an underground parking garage Studio Simulates a large live empty room Hall Places the sound in the middle of a hall with reflective hard bright walls Soft Simulates the space and ambience of a large concert hall Church Simulates a medium sized space with natural clear sounding reflections Cathedral Simulates a large space with long smooth reflections Arena Simulates a big natural sounding empty space Plate Simulates a hard bright reflection Use the Spread control to adjust plate size Build A nonlinear series of reflections Spread Simulates a wide indoor space with highly reflective walls Slapback Simulates a large space with a long delayed reflection Echo Simulates a large space with hard unnatu ral echoes Good for dense reverb Level Controls the output level of
81. at different frequen cies depending on the space Cavernous spaces of ten produce a booming bassy reverb whereas other spaces may have reverb tails which taper off to pri marily high frequencies Space allows for equaliza tion of the frequencies of the reverb tail in order to adjust the tonal characteristics of the reverb sound A reverb tail is often described by the time it takes for the sound pressure level of the reverb to decay 60 decibels below the direct sound and is known as RT60 Overall Space lets you adjust the decay as desired For surround processing decay can be ad justed for individual channel groups Space Convolution Reverb Convolution reverb goes beyond traditional analog and synthetic digital reverb techniques to directly model the reverb response of an actual reverb space First an impulse response IR is taken of an actual physical space or a traditional reverb unit An IR can be captured in mono stereo surround or any combination The IR as displayed by Space clearly shows the early reflections and the long de cay of the reverb tail Space uses a set of mathematical functions to con volve an audio signal with the IR creating a reverb effect directly modeled on the sampled reverb space By using non reverb impulse responses Space expands from reverb applications to a gen eral sound design tool useful for many types of au dio processing The downside of traditional software based convo lution
82. audio spectrum becomes de layed compared to mid and high frequencies SPR corrects the bass delay anomaly to restore clarity and openness and significantly increases the appar ent bass energy level without adding any amplitude equalization or bass boost To audition the effect of SPR on the audio signal turn the Ax switch Off and turn SPR On Then al ternately turn the Bypass switch On and Off to hear the SPR effect on incoming audio The SPR function is shown in the following figure which shows the frequency dependent time delay that is produced Note that this is not the same as a group delay Group delay is a constant time delay at all frequencies Bypass Switch The Bypass switch allows the main audio signal to bypass the Aural Exciter plug in completely The indicator switch illuminates when Bypass is engaged The Bypass switch on the plug in provides the same function as the Bypass button in the Pro Tools Plug In window on a per channel basis Link Switch The Link switch is for stereo operation only It links the left and right controls so they work as one Grab the control on one page with the cursor and move it to the required position The control on the other page automatically updates In this way both controls can be set to the exact same position Ste reo controls may be linked temporarily by holding down the Shift key while adjusting the control The switch illuminates when Link is activated SPR switch
83. button 174 Chorus section 174 clip indicator 169 Color graph display 175 Coloration control 172 Contour display 176 Depth control 174 Diffusion control 171 Early Reflection Level control 172 Early Reflection Pre Delay control 172 Early Reflection section 171 Early Reflection Spread control 172 Early Reflections button 176 Early Reflections On button 172 EQ graph display 175 Front button 177 Front control 173 High Frequency Color control 173 High Frequency control 175 High Frequency Crossover 176 High Frequency Ratio 176 High Frequency Rear Cut control 175 High Gain control 175 Input control 173 Input Output meter 169 Level control 170 Levels section 173 Low Frequency Color control 173 Low Frequency control 175 Low Frequency Crossover 176 Low Frequency Ratio 176 Low Gain control 175 Mix section 174 Next and Previous Room Type buttons 170 Pre Delay control reverb tail 171 Pre Delay Link button 172 Rate control 174 Rear button 177 Rear ER control 173 Rear Level Link button 173 Rear Reverb control 173 Rear Shape control 171 Reverb Contour button 176 Reverb section 170 reverb tail controls 170 Reverb tail type 170 Reverb Type 169 Reverb Type menu 170 Room Coloration section 172 Room Type 169 Room Type Category menu 170 Room Type Name menu 170 Size control 170 Spread control 171 Stereo Width control 174 supported channel formats 167 Time control 170 Wet Dry control 174 ReWire plug in and voices 312 automating input sw
84. button 331 Clip field 332 Export button 331 historical metering 329 Offset field 332 Oversampled Clip Events browser 330 Oversampled Level meter 331 real time metering 329 Signal Clip Events browser 330 Signal Level meter 331 View Time menu 331 Maxim plug in 91 Attenuation control 94 Ceiling control 94 controls 93 Dither control 95 drum limiting 92 dynamic range of a mix 92 dynamic range of individual instruments 92 Histogram 91 93 Input Level control 93 Limiting 92 Link button 95 96 Mix control 95 Noise Shaping control 95 Online Help 91 Output control 94 peak levels 91 Peak limiting 91 92 quantization noise 95 Release control 94 signal delay 93 signal peaks 92 Threshold control 94 96 X axis of histogram 93 Y axis of histogram 93 Mod Delay III plug in 204 Delay Time 205 Feedback setting 205 Groove control 206 Input meters 204 Link button 205 Low Pass Filter 205 Meter setting 205 Mix control 206 Output Gain control 206 Output meters 206 Phase Invert 204 Sync option 205 Tempo control 205 Moogerfooger 12 Stage Phaser plug in 222 allpass 222 analog filters 222 modulation 222 resonant filters 222 tremolo 222 whooshing 222 Moogerfooger Analog Delay plug in 207 Bucket Brigade Analog Delay Chips 207 echo analog delay 207 Audio Plug Ins Guide 355 lo fi 208 sound generator 208 Moogerfooger Lowpass Filter plug in 219 auto wah 221 envelope follower 220 filters 220 resonance filter 219 whistling 220 Moogerf
85. by generating less com pression in the dynamic processor resulting in a more powerful side chain signal If you need more headroom when adjusting the Mix control lower the input Level and re tune the Mix control Drive Control The Drive control sets the sensitivity to the bass generating side chain The corresponding Drive meter shows the actual peak level of the side chain input There is a boost in the side chain signal of 12 dB when the Drive control is set to Max The Drive control needs to be set at a point where the dynamic processor receives the optimum level required for Big Bottom Pro to work effectively To find the optimum level adjust the Drive control until the Comp meter displays in the yellow area Make sure the Drive meter does not indicate clip ping If the Comp meter is not showing any activity the input level is too low Adjust the Level control ac cordingly When the AutoTrace switch is set to the On position the setting of the Drive control is less sensitive and the Big Bottom Pro side chain af fects a wider input range In general higher Drive settings to the side chain provide better control over peaks while lower Drive settings tend to produce a more open sound By adjusting both the Drive and Mix controls you can experiment with the different colors or tim bral modifications Big Bottom Pro is able to gener ate Tune Control The Tune control sets the bandwidth corner fre quency
86. compressor The Output control has no effect on the level of distortion applied to the signal VU Meter Output meter Input meter Input Clipping indicator Meter Mode button Output Clipping indicator Internal Clipping indicator Gain meter Chapter 22 Smack 131 Input and Output Meters The Input and Output meters indicate input and output signal levels in dBFS dB relative to full scale or maximum output The Internal Clipping indicator labelled INT CLIP turns red when the signal exceeds the avail able headroom Clicking the Internal Clipping indi cator clears it Alt clicking Windows or Option clicking Mac clears the clip indicators on all channels Using the Smack Side Chain Input Smack provides side chain processing capabilities Compressors typically use the detected amplitude of their input signal to cause gain reduction This split off signal is called the side chain However an external signal referred to as the Key Input can be used to trigger compression A typical use for external side chain processing is to control the dynamics of one audio signal using the dynamics of another signal For example you could use a lead vocal track to duck the level of a background vocal track so that the background vo cals do not interfere with the lead vocals To use an external Key Input to trigger compression 1 Insert Smack on a track you want to compress using external side chain
87. delay values and use TimeAdjuster to compensate for the delay 1 Do one of the following Control click Windows or Command click Mac the Track Level Indicator to toggle be tween level that appears on the display as vol headroom pk and channel delay dly indications Delay values are shown in samples Select the Delay Compensation view from the Mix Window View selector This view will ap pear below the track name Delay User Offset Compensation Amount values are shown in samples 2 Apply the TimeAdjuster plug in to the track whose delay you want to increase and Control click Windows or Command click Mac its Track Level indicator until the channel delay value is displayed for that track 3 Change the delay time in TimeAdjuster by mov ing the Delay slider or entering a value in the Delay field until the channel delay value matches that of the first track 4 Test the delay values by duplicating an audio track and reversing its phase while compensat ing for delay When to Compensate for Delays If you want to compensate for delays across your entire system with Time Adjuster you will want to calculate the maximum delay incurred on any channel and apply the delays necessary to each channel to match this channel However this may not always be necessary You may only really need to compensate for delays be tween tracks where phase coherency must be main tained as with instru
88. ethereal reverb character It is of ten used for creative effects rather than to simulate a realistic acoustic environment Depth Control Depth controls the amplitude of the sine wave gen erated by the LFO low frequency oscillator and the intensity of the chorusing The higher the set ting the more intense the modulation The range of this control is from 0 to 100 Rate Control Rate controls the frequency of the LFO The higher the setting the more rapid the chorusing The range of this control is from 0 1 Hz to 30 0 Hz Setting the Rate above 20 Hz can cause frequency modulation to occur This will add side band har monics and change the reverb s tone color produc ing interesting effects Typical settings are between 0 2 Hz and 1 0 Hz Chorus On Button This button toggles the chorus effect on or off ReVibe II Mix Section Controls The Mix section has controls for adjusting the rela tive levels of the source signal and the reverb ef fect Wet Control The Wet control adjusts the mix between the dry unprocessed signal and the reverb effect If you in sert the ReVibe II plug in directly onto an audio track settings from 30 to 60 are a good starting point for experimenting with this control The range of this control is from 0 to 100 Stereo Width Control Stereo Width controls the stereo field spread of the front reverb channels A setting of 0 produces a mono reverb but leaves the panning of th
89. format see Channel Compatibility and Space on page 189 An IR can be loaded by double clicking with the mouse or using the Load button displayed at the top of the IR browser drawer The currently loaded IR is highlighted with a small dot next to the file name in the browser The IR browser can be operated using the follow ing shortcuts When the IR browser has keyboard focus a blue highlight is displayed around the edge of the browser window The IR browser lets you install and import new IRs Each IR folder reflects a folder on the hard drive When importing a new IR folder a standard file di alog will be displayed to enable the user to choose the folder that contains the desired IR The IR browser also provides a Favorites folder which is a user defined group of links to IRs in the IR browser Favorites can be sorted in any desired order by dragging and dropping them as required In addition folders can be created in Favorites us ing the New Folder in Favorites function in the Edit menu To add an IR file or folder to the Favorites folder 1 In the IR browser select the desired IR file or folder 2 From the IR browser s Edit menu select Add to Favorites IR Browser IR Browser Shortcuts Browser Navigation Arrow Keys Load IR Enter Windows Return Macintosh Open close all folders Alt click Windows Option click Macintosh Edit menu Right click Windows or Macintosh Control clic
90. graphical and numeric representations for the amount of compression The numerical display for the Peak value turns orange when the signal exceeds 0 dB on the meters You can click the numerical display to reset the dis played value Pro Expander Input Section The Input section provides input metering and con trols for adjusting the level of the input signal Input Level The Input Level control sets the input gain of the plug in before processing letting you boost or at tenuate gain at the plug in input stage To adjust the level of the input signal do one of the following Click in the Input Level field and type a value 36 0 dB to 36 0 dB Click the Input Level control and drag up or down to adjust the Input Level setting Input Meters The Input meters show peak signal levels before processing Peak indicators in the Output meters Peak Hold value Peak Hold indicator Sample Peak indicator Average meter Input section with Meters and Input Level control Chapter 19 Pro Expander 109 Pro Expander Output Section The Output section provides output metering and controls for adjusting the level of the output signal The Output meters can also be toggled to show post processing gain attenuation meters Output Level The Output Level control sets the output level after processing letting you boost or attenuate gain on the channel where the Pro Expander plug in is being used To adjus
91. in Moogerfooger Analog Delay provides a warm sounding delay in the digital domain A delay cir cuit produces a replica of an audio signal a short time after the original signal Mixed together the delayed signal sounds like an echo of the original If this mixture is fed back to the input of the delay circuit the delayed output provides a string of echoes that repeat and die out gradually a classic musical effect The Moogerfooger Analog Delay uses Bucket Bri gade Analog Delay Chips to achieve its delay These analog integrated circuits function by pass ing the audio waveform down a chain of thousands of circuit cells just like water being passed by a bucket brigade to put out a fire Each cell in the chip introduces a tiny time delay The total time de lay depends on the number of cells and on how fast the waveform is clocked or moved from one cell to the next With the advent of digital technology these and similar analog delay chips have gradually been phased out of production In fact Bob Moog se cured a supply of the last analog delay chips ever made and used them to build a Limited Edition of 1 000 real world Moogerfooger Analog Delay units Compared to digital delays the frequency and overload contours of well designed analog delay devices generally provide smoother more natural series of echoes than digital delay units Another difference is that the echoes of a digital delay are static because they
92. information on the concepts of reverb and convolution reverb Reverb Basics Reverberation is an essential aspect of the sound character of any space in the real world Every room has a unique reverb sound and the qualities of a reverb can make the difference between an or dinary and an outstanding recording The same re verb principles responsible for the sound of a ma jestic soaring symphony in a concert hall also produce the booming unintelligible PA system at a train station Recordings of audio in the studio con text have traditionally been captured with a mini mum of real reverb and engineers have sought to create artificial reverbs to give dry recorded mate rial additional dimension and realism The first analog reverbs were created using the echo chamber method which consists of a speaker and microphone pair in a quiet closed space with hard surfaces often a tiled or concrete room built in the basement of a recording studio Chamber reverbs offered a realistic complex re verb sound but provided very little control over the reverb as well as requiring a large dedicated room Plate reverbs were introduced by EMT in the 1950s Plate reverbs provide a dense reverb sound with more control over the reverb characteristics Although bulky by modern standards plate reverb units did not require the space needed by a chamber reverb Plate reverbs function by attaching an elec trical transducer to the center of a thin
93. initial reverb contour Size Determines the rate of diffusion buildup and acts as a master control for Time and Spread within the reverberant space Size values are given in meters and can be used to approximate the size of the acoustic space you want to simulate When considering size keep in mind that the size of a reverberant space in meters is roughly equal to its longest dimension Diffusion Controls the degree to which initial echo density increases over time High Diffusion set tings result in high initial buildup of echo density Low Diffusion settings cause low initial buildup After the initial echo buildup Diffusion continues to change by interacting with the Size control and affecting the overall reverb density Use high Dif fusion settings to enhance percussion Use low or moderate settings for clearer more natural sound ing vocals and mixes Pre Delay Determines the amount of time that elapses between the original audio event and the onset of reverberation Under natural conditions the amount of Pre delay depends on the size and construction of the acoustic space and the relative position of the sound source and the listener Pre delay attempts to duplicate this phenomenon and is used to create a sense of distance and volume within an acoustic space Long Pre Delay settings place the reverberant field behind rather than on top of the original audio signal Reverb One Early Reflection Controls The Early Refl
94. key input to trigger its effect 5 Adjust the plug in Threshold control to fine tune external key input triggering External Key Key Listen Selecting a key input Chapter 15 Impact 85 Chapter 15 Impact Impact is available in DSP Native and AudioSuite formats Impact plug in provides critical control over the dynamic range of audio signals with the look and sound of a mixing console s stereo bus compressor Impact supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Impact supports mono stereo and greater than stereo multichannel formats up to 7 1 Impact Controls Impact Ratio Control Ratio sets the compression ratio If the ratio is set to 2 1 for example it will compress changes in signals above the threshold by one half This control pro vides four fixed compression ratios 2 1 4 1 10 1 and 20 1 Selecting 2 1 applies very light compres sion selecting 20 1 applies heavy compression bordering on limiting Impact Attack Control Attack sets the compressor attack time To use compression most effectively the attack time should be set so that signals exceed the threshold level long enough to cause an increase in the aver age level This helps ensure that gain reduction does not decrease the overall volume The range of this control is from 0 1 ms to 30 0 ms Impact Threshold Control Threshold sets the decibel level that a signal must exceed for Impact to begin app
95. lected as the side chain input click Listen To stop listening to the side chain input click Lis ten again 6 Adjust Impact s Threshold control to fine tune Key Input triggering Remember to disable Listen to resume normal plug in monitoring Chapter 16 JOEMEEK SC2 Compressor 89 Chapter 16 JOEMEEK SC2 Compressor The JOEMEEK SC2 Compressor is a dynamics processing plug in that is available in DSP Native and AudioSuite formats SC2 supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates SC2 operates as a mono multi mono or stereo plug in Legendary producer Joe Meek used to say If it sounds right it is right Nowhere is this more ap parent than in Joe Meek s masterful use of non lin ear sometimes severe compression in his produc tions In use by top producers the world over JOEMEEK compression is the secret weapon that gives your sound the character and excitement it deserves The JOEMEEK Compressor is designed purely as an effects compressor Its purpose is to change the way the ear perceives sound its action changes the clarity balance and even rhythmic feel of music JOEMEEK Compressor Controls The SC2 Compressor provides the following con trols Input Gain Input Gain adjusts the input level to the compressor Compression The Compression control affects the gain to the side chain of the compressor Use it along with Slope to adjust the amount of compres
96. lowest gain to 0 dB highest gain Compressor Limiter III Ratio Control The Ratio control sets the compression ratio or the amount of compression applied as the input signal exceeds the threshold For example a 2 1 compres sion ratio means that an input level that is 2 dB above the threshold will be attenuated resulting in an output level that is 1 dB over the threshold This control ranges from 1 1 no compression to 100 1 hard limiting Threshold arrow on input meter Threshold indicator on Dynamics Graph display Chapter 12 Dynamics III 64 Compressor Limiter III Attack Control The Attack control sets the attack time or the rate at which gain is reduced after the input signal crosses the threshold The smaller the value the faster the attack The faster the attack the more rapidly the Compres sor Limiter applies attenuation to the signal If you use fast attack times you should generally use a proportionally longer release time particularly with material that contains many peaks in close proximity This control ranges from 10 s fastest attack time to 300 ms slowest attack time Compressor Limiter III Release Control The Release control sets the length of time it takes for the Compressor Limiter to be fully deactivated after the input signal drops below the threshold Release times should be set long enough that if sig nal levels repeatedly rise above the threshold the gain reduction re
97. most prominent audio peaks 3 Loop playback and look at the data displayed by the Histogram and Attenuator meter 4 Select the Link button to link the Threshold and Ceiling controls You can then adjust these con trols together proportionally and using the By pass button compare the audio with and without limiting 5 Adjust the Threshold downwards until you hear and see limiting occur then bring the Threshold back up slightly until you have roughly the amount of limiting you want 6 Periodically click and clear the Attenuation me ter to check attenuation In general applying 2 dB to 4 dB of attenuation to occasional peaks in pop oriented material is appropriate 7 Use the Bypass button to compare the processed and unprocessed sound and to check if the re sults are acceptable 8 Avoid pumping effects with heavier limiting by setting the Release control to longer values 9 When you get the effect you want deselect the Link button and raise the output level with the Ceiling slider to maximize signal levels without clipping In general a value of 0 5 dB or so is a good maxi mum ceiling Don t set the ceiling to zero since the digital to analog converters on some DATs and CD players will clip at or slightly below zero Maxim and Mastering If you intend to deliver audio material as a 32 bit floating point or 24 bit audio file on disk for profes sional mastering be aware that many mastering en gineers prefer mat
98. negative Pre Delay setting can be used to eliminate the early portion of an IR A large neg ative Pre Delay setting lets you use the very end of a reverb tail for creative sounds not possible with standard reverbs Group Selectors Level controls 5 0 shown Delay controls 5 0 shown Chapter 30 Space 199 Late Delay Adjusts length of the Late Delay from zero to 200 ms The Late Delay is the time be tween the Early Reflections and the Late Reflec tions or tail of the reverb Increasing the Late Delay control from zero allows the reverb tail to be delayed so that it does not start immediately after the early portion of the IR As Late Delay is increased the reverb tail starts later in time and makes the reverb space sound larger Large amounts of late delay can be used to achieve creative effects not possible with standard reverbs Front Rear Center Delay In quad and 5 0 channel output modes adjusts length of the Front Rear and Center Delays independently from zero to 200 ms Space Early Section Controls The Early group controls the character of the early portion of the IR and the early reflections The pri mary control is Early Length which defines the size of the early portion of the IR waveform When loading an IR from an audio file Space relies on the user to define which part of the IR is the early por tion of the waveform By default the Early length is set to 20 ms The early portion of the IR waveform i
99. of the low pass filter in the side chain prior to the dynamics processor The range of the Tune control is from 40 400 Hz Aside from the Mix control this is the most import ant control on the Big Bottom Pro plug in The Tune control is used to isolate the range of fre quencies being enhanced by Big Bottom Pro Mix Control The Mix control adjusts the amount of the Big Bot tom Pro enhancement signal being added to the original signal The lower the setting the subtler the effect The higher the setting the more dramatic the effect It s important to note that higher settings may increase the peak output Switches In Out Switch The In Out switch gives you the choice of turning the Big Bottom Pro process On or Off When the switch is set to the On position Big Bottom Pro en hancement is sent to the outputs The switch illuminates when the Big Bottom Pro effect is activated Unlike system bypass the audio from the input travels through the DSP algorithm on the way to the output whenever the In Out switch is set to Off Chapter 43 Aphex Big Bottom Pro 253 Switching back and forth from On to Off provides a quick A B comparison allowing you to hear the enhancements from the Big Bottom Pro effect in your program content Solo Switch When engaged the Solo switch allows you to audi tion the Big Bottom Pro side chain effect without the main audio signal The switch illuminates when the Solo switch is activated
100. past Application Most contemporary audio recording is done with Digital Audio Workstations DAWs although digital mixing systems in the form of outboard dig ital mixers are also very popular To the user these digital systems appear similar to traditional audio tools and are designed order to emulate the opera tion of a conventional analog recording system One familiar analog tool that has been carried over to the digital realm is a peak meter that tells the amplitude of the waveform s peaks In the analog realm peak signal was an indicator that would alert the audio engineer when the peak signal level was getting too high A peak signal in analog recording would cause the tape to saturate creating distor tion In an analog system however this type of dis tortion was often deliberately engineered into tracks in order to achieve a certain sound In the digital realm this type of meter is important and more vital because if the amplitude of a wave form exceeds the top of the measurable scale full scale or full code the signal will clip causing unwanted and unpleasant distortion rather than the traditional distorted sound of analog This digital clipping occurs because the waveform is lopped off and the data is changed When the waveform is reconstructed it cannot be accurately done in order to represent the original waveform Instead it has a significant amount of inharmonic distortion caused by
101. possible level without clipping RMS mode adjusts the input signal to a level consistent with the RMS Root Mean Square value or the effective average level of the selected clip To change the gain of an audio clip 1 Select the clip whose gain you want to change 2 Choose AudioSuite gt Other gt Gain 3 Adjust the Gain slider 4 Click Preview to audition your changes 5 Ensure that Use In Playlist is enabled 6 Click Render Duplicate Gain Chapter 61 Other AudioSuite Plug In Utilities 348 Invert The Invert plug in reverses the polarity of selected audio Positive sample amplitude values are made negative and all negative amplitudes are made pos itive This process is useful for altering the phase or po larity relationship of tracks The Invert plug in is useful during mixing for modifying frequency re sponse between source tracks recorded with multi ple microphones You can also use it to correct au dio recorded out of phase with an incorrectly wired cable To invert the phase an audio clip or selection 1 Select the clip whose phase you want to invert 2 Choose AudioSuite gt Other gt Invert 3 Ensure that Use In Playlist is enabled 4 Click Render Normalize The Normalize plug in optimizes the volume level of an audio selection Use it on material recorded with too little amplitude or on material whose vol ume levels are inconsistent as in a poorly recorded narration Unli
102. processing 2 On the audio track or Auxiliary Input that you want to specify as the Key Input the signal that will be used to trigger compression click the Send button and select the bus path to the track that will use side chain processing 3 In the track that you are compressing click the instance of Smack in the Inserts pop up menu 4 In the Smack plug in window click the Key In put menu and select the input or bus path that you have designated as the Key Input 5 Begin playback Smack uses the input or bus that you selected as a Key Input to trigger its ef fect 6 To fine tune the amount of compression adjust the send level from the Key Input track 7 To tailor the side chain signal so that the detec tor is frequency sensitive use the Side Chain EQ filter see Smack Side Chain EQ Filter on page 129 for more information The Side Chain EQ filter lets you tailor the equalization of the side chain signal so that the compression becomes frequency sensi tive See Smack Side Chain EQ Filter on page 129 for more information The Key Input must be monophonic When you are using a Key Input to trigger compression the Input control has no effect on the amount of compression Part IV Pitch and Time Shift Plug Ins Chapter 23 Pitch II 133 Chapter 23 Pitch II Pitch II is a pitch shifting plug in that is available in DSP Native and AudioSuite formats Pitch II is designed for a variety of a
103. processing 70 Audio Plug Ins Guide 352 E Eleven Free plug in 255 Eleven plug in 255 advanced applications 278 Amp Bypass 266 Amp Type 265 navigating 262 amps 268 controls 266 list of 265 AudioSuite 279 Axis On Off 270 beat clock see Tempo Sync 267 blending amps and cabinets 274 and cab resonance 275 and phase 277 buffer 259 bypass amp only 266 cabinet and mic 269 Cab Type 268 navigating 262 Cabinet bypass 269 cabinets 268 Category 261 close mic 270 comb filtering 277 combining 274 combo amps 268 condenser 269 cone breakup 269 control surfaces and unused controls 262 controls 262 CPU Usage 269 Depth 267 DSP 279 dynamic 269 flip phase 277 format mono or multi mono 261 Gain 1 266 Gain 2 266 gate 265 Hardware Buffer for Input Calibration 259 Harmonic 261 humbucker 260 impedance 258 input about guitar amps and levels 257 Input Trim 264 input calibration 260 HW Buffer 259 Input LED 264 inserting Eleven 261 intensity Tremolo 267 lamp bypass 266 LED colors for input calibration 260 load 275 Manufacturer 261 Master 267 Master section 264 mic placement 270 Mic Type 269 navigating 262 mics microphones 270 on and off axis 270 MIDI 262 Learn 263 tempo sync 267 mono multi mono 261 multichannel 261 multiple cabinets 274 noise gate 265 Output 264 phase 277 278 polarity see phase 277 Presence 267 presets 263 Previous Next arrows 262 pure excess 274 recording 271 dry 270 dry and Elev
104. rides the Release setting and automatically adjusts the limiter release time based on changes in the pro gram material When the Auto Release toggle is disabled you can set the limiter release time man ually using the Release control Listen Enable Listen to isolate the processed part of the audio signal This can help you hear what parts of the input signal are triggering limiting which in turn can help you better understand the character istics of the current Threshold Character and Release settings Insert a multi mono instance of Pro Limiter to ensure no linking between channels Each channel will trigger its own processing inde pendently of the other channels Chapter 20 Pro Limiter 122 Pro Limiter Loudness Numeric Displays Pro Limiter provides numeric displays that show the current loudness or peak level of the processed signal Note that Pro Limiter conforms to the ITU R BS 1770 3 standard for loudness metering The numeric loudness values generated by Pro Limiter can be used for both EBU R128 and ATSC A 85 CALM Act loudness compliance Integrated Displays the current integrated level of the processed signal level in LUFS Range Displays the range of the processed signal level over time in LU True Peak Displays the true peak hold value of the Output signal in dB Short Term Displays the short term output signal level in LUFS About LUFS LU and dB LUFS Loudness Unit Referenced to Full Scale is
105. s detector to trigger dynamics processing D3 lets you switch between internal and external side chain processing With external side chain processing a plug in s de tector is triggered by an external signal such as a separate reference track or audio source known as the key input A typical use for this feature is to use a kick drum track to gate and tighten up a bass track or a rhythm guitar track to gate another instrument External Key External Key toggles external side chain process ing on or off When this button is enabled the plug in uses the amplitude of an external signal the key input to trigger compression or limiting When this button is disabled the plug in uses the amplitude of the input signal to trigger dynamics processing Key Listen Key Listen enables and disables auditioning of the key input controlling the external side chain This is useful for fine tuning the compressor s settings to the key input Using a Key Input for External Side Chain Processing To use a key input to trigger dynamics processing 1 Click the Key Input selector and select the input or bus carrying the audio from the reference track or external audio source 2 Click External Key to activate external side chain processing 3 To listen to the key input that will be used to control side chain processing click Key Listen to enable it 4 Begin playback The plug in uses the input or bus that you chose as an external
106. saved as required so that specific control settings can be retained for future sessions If you save pre sets without embedding the IR waveform be sure that you include the IR waveform with the session when transferring the session between different Pro Tools systems There are two important items to note about using presets in Space Space presets do not store information for the Wet and Dry level controls This is to enable you to change presets without losing level informa tion Likewise the Pro Tools Compare function is not enabled for these controls A Space preset only includes the currently se lected snapshot Space Snapshots In addition to presets Space lets you manage a group of settings called snapshots that can be switched quickly using a single automatable con trol Each snapshot contains a separate IR and set tings for all Space controls IRs in a snapshot have been pre processed by the impulse computer and can be loaded instantly into the convolution processor Snapshots are useful for example in post production mixes when the re verb is changed for different scenes via automation as the picture moves from one scene to another Embedding IRs in Sessions Presets and Snapshots By default all IR and snapshot info used by Space including up to ten IRs is saved in the Pro Tools session file Likewise plug in presets contain a saved copy of the IR and settings in the currently selected snaps
107. selected and in circuit and the Low Mid Peak Mid Peak High Mid Peak bands are in circuit 7 Band EQ High Pass Low Pass Low Shelf Peak Mid Peak High Shelf Peak Low Mid Peak High Mid Peak Input Output Level meters Frequency Graph Display Input Output Level and Polarity controls Low Notch High Notch Chapter 4 EQ III 21 7 Band EQ III High Pass Low Notch The High Pass Notch band is switchable between high pass filter and notch EQ functions By default this band is set to High Pass Filter High Pass Filter Attenuates all frequencies below the Frequency setting at the selected slope while letting all frequencies above pass through Low Notch EQ Attenuates a narrow band of fre quencies centered around the Frequency setting The width of the attenuated band is determined by the Q setting The High Pass and Low Notch EQ controls and their corresponding graph elements are displayed on screen in gray The following control values are available 7 Band EQ III Low Pass High Notch The Low Pass Notch band is switchable between low pass filter and notch EQ functions By default this band is set to Low Pass Filter Low Pass Filter Attenuates all frequencies above the Frequency setting at the selected slope while letting all frequencies below pass through High Notch EQ Attenuates a narrow band of fre quencies centered around the Frequency setting The width of the attenuated band is determined b
108. sound itself Room 1 A medium sized natural rich sounding room that can be effectively varied in size between very small and large with good results Room 2 A smaller brighter reverberant character istic than Room 1 with a useful adjustment range that extends to very small Ambient A transparent response that is useful for adding a sense of space without adding a lot of depth or density Extreme settings can create inter esting results Nonlinear Produces a reverberation with a natural buildup and an abrupt cutoff similar to a gate This unnatural decay characteristic is particularly useful on percussion since it can add an aggressive char acteristic to sounds with strong attacks Size Control The Size control in conjunction with the Algo rithm control adjusts the overall size of the rever berant space There are three sizes Small Medium and Large The character of the reverberation changes with each of these settings as does the rel ative value of the Decay setting The Size buttons can be used to vary the range of a reverb from large to small Generally you should select an algorithm first and then choose the size that approximates the size of the acoustic space that you are trying to cre ate Chapter 27 D Verb 157 Diffusion Control Diffusion sets the degree to which initial echo den sity increases over time High settings result in high initial build up of echo density Low settings cause l
109. started MasterMeter Meters Signal Level Meters The Signal Level meter shows the instantaneous signal level of the current audio signal The clip light at the top of the meter can be cleared by click ing on it or by using the Clear button Oversampled Level Meter The Oversampled Level meter shows the instanta neous signal level of the current audio signal after it has been oversampled As the oversampling pro cess can create levels above 0 dB this meter shows an expanded scale from 6 dB to 0 dB and from 0 dB to 6 dB The clip light at the top of the meter can be cleared by clicking on it or using the Clear button MasterMeter Clear Button The Clear button clears all of the historical infor mation displayed in Signal Clip Events browser and the Oversampled Clip Events browser It also click the clip lights at the top of the Signal Level and Oversampled Level meters This information is also cleared when the Pro Tools transport is acti vated by pressing Play or Record MasterMeter Export Button The Export button exports all of the information displayed in the two browsers to the clipboard as tab delimited text It can then be pasted into any text or spreadsheet application MasterMeter View Time Menu The View Time menu lets you select the way in which timing information is displayed in either minutes and seconds format or in samples format This affects the timecode display in both the data browsers and t
110. subharmonics To create classic subhar monic synthesis effects set the Pre Filter and Post Filter to a relatively low frequency The range of the Post Filter control is 43 Hz to 22 kHz with a maximum value of Thru which ef fectively means bypass Recti Fi Mix Control Mix adjusts the mix of the rectified waveform with the original unprocessed waveform Recti Fi Output Meter The Output Meter indicates the output level of the processed signal Note that this meter indicates the output level of the signal not the input level If this meter clips the signal may have clipped on in put before it reached Recti Fi Monitor your send or insert signal levels closely to prevent this from happening Chapter 47 Reel Tape Saturation 286 Chapter 47 Reel Tape Saturation Reel Tape Saturation is part of the Reel Tape suite of tape simulation effects plug ins Reel Tape Sat uration simulates the saturation effect of an analog tape machine modeling its frequency response noise and distortion characteristics but without any delay or wow and flutter effects Reel Tape Saturation is available in DSP Native and AudioSuite formats Reel Tape Saturation supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Reel Tape Saturation operates as a mono multi mono or stereo plug in For years engineers have relied on analog tape to add a smooth warm sound to their recordings When driven hard tape r
111. tfx 104 1666666667 24 30 to 25 Film to NTSC 0 1 tfx 99 9000999001 24 30 to 23 976 29 97 NTSC to Pal 4 2667 tfx 104 2708333333 23 976 29 97 to 25 NTSC to Film 0 1 tfx 100 10 23 976 29 97 to 24 30 Chapter 25 Vari Fi 145 Chapter 25 Vari Fi Vari Fi is an AudioSuite plug in that provides a pitch change effect similar to a tape deck or record turntable speeding up from or slowing down to a complete stop Vari Fi preserves the original dura tion of the audio selection Vari Fi provides a pitch change effect similar to a tape deck or record turntable speeding up from or slowing down to a complete stop Features include Speed up from a complete stop to normal speed Slow down to a complete stop from normal speed Vari Fi Controls Change Controls Slow Down When selected Slow Down applies a pitch change effect to the selected audio similar to a tape re corder or record turntable slowing down to a com plete stop Speed Up When selected Speed Up applies a pitch change effect to the selected audio similar to a tape re corder or record turntable speeding up from a com plete stop This effect does not change the duration of the audio selection Selection Controls The Selection setting determines the duration of the rendered clip in relation to the processing Fit To When the Fit To option is selected the length of the audio selection is retained when rendering the Au dioSuit
112. the early reflections Turning the Early Reflections Level slider com pletely off produces a reverb made entirely of re verb tail Spread Globally adjusts the delay characteristics of the early reflections moving them closer together or farther apart Use Spread to vary the size and char acter of an early reflection preset Setting the Plate preset to a Spread value of 50 for example will change the reverb from a large smooth plate to a small tight plate Delay Master Determines the amount of time that elapses be tween the original audio event and the onset of early reflections Early Reflect On Toggles early reflections on or off When early re flections are off the reverb consists entirely of re verb tail Chapter 28 Reverb One 164 Reverb One Graphs The reverb graphs display information about the tonal spectrum and envelope contour of the reverb The Reverb EQ and Reverb Color graphs provide graphic editing tools for shaping the harmonic spectrum of the reverb Editing Graph Values In addition to the standard slider controls the Re verb EQ and Reverb Color graph settings can be adjusted by dragging elements of the graph display To select the EQ or Color graph for editing Select the EQ icon or the Color icon To cut or boost a particular band Drag a Band Cut Boost breakpoint up or down To adjust frequency or crossover Drag a Frequency Crossover slider right or left To
113. the highpass filter As the control is set to a higher value the corner fre quency of the highpass filter is increased Use this control to reduce boom and low frequency cancel lations that can happen when mixing the reverb output with a dry signal Balance Early Balance controls the left right gain balance of the early portion of the IR as specified by the Early Length control Adjust the Balance to control the apparent position of the reverb input in the stereo image A negative value reduces the right channel gain A positive value reduces the left channel gain Space Reverb Section Controls The Reverb group offers a low and high shelf EQ in addition to width and balance controls The EQ op erates prior to convolution processing Lo Freq Adjusts the frequency of a low frequency filter from 20 to 500 Hz Lo Gain Cuts or boosts the frequency set in Lo Freq from 15 dB to 15 dB Hi Freq Adjusts the frequency of a high frequency filter from 500 Hz to 20 kHz Hi Gain Cuts or boosts the frequency set in Hi Freq from 15 dB to 15 dB Width Increase or reduces the stereo spaciousness of the reverb Use this control to tailor the reverb s character in a mix Keep in mind that an IR that has little stereo separation to begin with may have lim ited results Balance Controls the balance of the reverb output Use this control to balance a reverb from an IR that has been captured without a centered stereo image or for c
114. the overall character of the origi nal unprocessed signal In addition Maxim provides a histogram that dis plays the distribution of waveform peaks in the au dio signal This provides a convenient visual refer ence for comparing the density of waveform peaks at different decibel levels and choosing how much limiting to apply to the material Maxim Controls and Meters Maxim Input Level Meter This meter displays the amplitude of input signals prior to limiting Unlike conventional meters Maxim s Input meter displays the top 24 dB of dy namic range of audio signals which is where limit ing is typically performed This provides you with much greater metering resolution within this range so that you can work with greater precision Maxim Histogram The Histogram displays the distribution of wave form peaks in the audio signal This graph is based on audio playback If you select and play a short loop the histogram is based on that data If you se lect and play a longer section the Histogram is based on that Maxim holds peak data until you click the Histogram to clear it The Histogram provides a visual reference for comparing the density of waveform peaks at differ ent decibel levels You can then base limiting deci sions on this data The X axis of the Histogram shows the number of waveform peaks occurring at specific dB levels The Y axis shows the specific dB level at which these peaks occur The more waveform p
115. threshold To use compression most effectively the attack time should be set so that signals exceed the thresh old level long enough to cause an increase in the average level This helps ensure that gain reduction does not decrease the overall volume too drasti cally or eliminate desired attack transients in the program material Of course compression has many creative uses that break these rules About Limiting Limiting prevents signal peaks from ever exceed ing a chosen threshold and is generally used to pre vent short term peaks from reaching their full am plitude Used judiciously limiting produces higher average levels while avoiding overload clipping or distortion by limiting only some short term transients in the source audio To prevent the ear from hearing the gain changes extremely short at tack and release times can be used Limiting is used to remove only occasional peaks because gain reduction on successive peaks would be noticeable If audio material contains many peaks the threshold should be raised and the gain manually reduced so that only occasional extreme peaks are limited Cursor Color Compression Amount white no compression light orange below full ratio dark orange full ratio amount See De Esser III Frequency Graph Display on page 69 for information on using the De Esser graph display Compressor Limiter III Chapter 12 Dynamics III 63 Limiting generally begins wit
116. time 148 Pitch section 147 150 Pitch Shift control 150 pitch shifting a selection 152 Poly Faster mode 148 Polyphonic mode 148 post production workflow 153 Processed time 148 Audio Plug Ins Guide 360 processing audio 152 pull up pull down TCE percentages 153 Sensitivity control 149 TCE Trim tool 151 Tempo original and processed 148 Time section 147 148 Time Shift control 149 time shifting a selection 152 Transient section 147 149 Transpose control 150 Unit timebase selector 148 Window control 150 Z z Eleven input impedance 258 Avid 2001 Junipero Serra Boulevard Daly City CA 94014 3886 USA Technical Support USA Visit the Online Support Center at www avid com support Product Information For company and product information visit us on the web at www avid com
117. tions and build your own custom library see Eleven Settings Presets on page 263 Chapter 44 Eleven 269 Eleven Cabinet Controls All cabinets provide Bypass Speaker Breakup Mic Type and Position controls Cabinet Bypass The Bypass switch in the Cabinet section lets you bypass cabinet and microphone processing When in the Bypass position no cabinet or microphone processing is applied to the signal When in the On position cabinet and microphone settings are ap plied Speaker Breakup Full version HDX Only The Speaker Breakup slider lets you specify how much distortion is produced by the current speaker model Increasing the Speaker Breakup setting adds distortion that is a combination of cone breakup and other types of speaker distortion em ulated by the speaker cabinet model Range is from 1 to 10 Below certain frequencies the speaker cone vi brates as one piece Above those frequencies typi cally between 1 kHz and 4 kHz the cone vibrates in sections By the time a wave travels from the apex at the voice coil out to the edge of the speaker cone a new wave has started at the voice coil The result is comb filtering and other anomalies that contribute to the texture of the overall sound Mic Type The Mic Type selector lets you choose the micro phone to use with the selected cabinet Available Mic Types include the following Dynamic 7 Dynamic 57 Dynamic 409 Dynami
118. to no compression Attack Attack sets the compressor attack time To use compression most effectively the attack time should be set so that signals exceed the threshold level long enough to cause an increase in the aver age level This helps ensure that gain reduction doesn t decrease the overall volume The range of this control is from 1 0 ms to 150 0 ms Release Release controls how long it takes for the compres sor to be fully deactivated after the input signal drops below the threshold level In general this set ting should be longer than the attack time and long enough that if signal levels repeatedly rise above the threshold they cause gain reduction only once If the release time is too long a loud section of the audio material could cause gain reduction that per sists through a soft section The range of this con trol is from 25 milliseconds to 2 5 seconds Auto Release Auto Release enables the automatic release func tion In this mode the Release control has no effect on release time Instead the D3 uses a release time value that is program dependent and based on the audio being processed Ratio Threshold Attack Auto Release Auto Release Chapter 14 Focusrite D3 83 D3 Limiter Controls The Limiter icon which represents a limiter curve acts as a three state switch for enabling disabling or bypassing the limiter Its current state is indi cated by the icon s color White indica
119. to use see Eleven Cabinet Types on page 268 Gate Noise Gate Threshold The Noise Gate Threshold control sets the level at which the Noise Gate opens or closes At minimum Threshold setting the Noise Gate has no effect At higher Threshold settings only louder signals will open the Gate and pass sound Threshold range is from Off 90 dB to 20 dB Noise Gate Release The Noise Gate Release control sets the length of time the Noise Gate remains open and passing au dio Adjust the Release to find the best setting for the current task not too fast to avoid cutting off notes and not too slow to avoid unwanted noise Release range is from 10 ms to 3000 ms Master section To learn more about the Input control see Eleven Signal Flow Notes on page 280 Input Gate Output Amp Type Cab Type When you want to adjust Eleven s output level use the Output knob For tone distor tion use the amp Master volume For suggested gate applications see Using the Noise Gate on page 265 For details on where it derives its key trigger and applies its gate see Eleven Signal Flow Notes on page 280 Chapter 44 Eleven 265 Using the Noise Gate You can use the Noise Gate to silence unwanted signal noise or hum or just for an effect To use the Noise Gate to clean up unwanted low level noise 1 Connect and calibrate your guitar as explained in Connect your Guitar and Configure Source Inp
120. value of this control is Off which effectively means bypass Low Pass Filter Low Pass Filter controls the overall high frequency content of the reverb by setting the frequency above which a 6 dB per octave filter attenuates the processed signal The maximum value of this con trol is Off which effectively means bypass Selections for D Verb AudioSuite Processing Because AudioSuite D Verb adds additional mate rial the delayed audio to the end of selected audio make a selection that is longer than the original source material to allow the additional delayed au dio to be written to the end of the audio file If you select only the original material without leaving additional space at the end delayed audio that occurs after the end of the selection to be cut off Chapter 28 Reverb One 158 Chapter 28 Reverb One Reverb One is a world class reverb processing plug in that provides the highest level of professional sonic quality and reverb shaping control A set of unique easy to use audio shaping tools lets you customize re verb character and ambience to create natural sounding halls vintage plates or virtually any type of rever berant space you can imagine Reverb One is available in DSP Native and AudioSuite formats Reverb One Chapter 28 Reverb One 159 Reverb One supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Reverb One operates as a mono multi mono mono to stereo o
121. wet Turn the Dry Mix knob clockwise to 100 to pass only the input signal 100 dry Lookahead The Lookahead control lets you add a certain amount of delay in milliseconds for analyzing in coming audio All attack transients take a certain amount of time from the onset of the signal to the actual transient peak especially those with lower frequencies like a kick drum Adjust the Looka head time to ensure that processing with an Attack setting of 0 or at least very short can be accurate to the true attack time of transients in the signal Note that as soon as Lookahead is engaged the full amount of delay time is added to the Pro Expander plug in processing latency When Lookahead is set to Off no additional latency is incurred You can compensate for plug in processing delay using Automatic Delay Compensation in Pro Tools For more information see the Pro Tools Reference Guide Chapter 19 Pro Expander 114 Pro Expander Side Chain Processing Dynamics processors typically use the detected amplitude of their input signal to trigger gain re duction This is known as a side chain signal Pro Expander provides filters for side chain processing and supports external key side chain capabilities With external key side chain processing you can trigger dynamics processing using an external sig nal such as a separate reference track or audio source instead of the input signal This external source is known as the key in
122. 0 dB The 0 0 dB value represents a standard overbias calibration of 3 dB for analog tape ma chines so the control acts as a bias offset rather than as an absolute bias control Cal Adjust Cal Adjust simulates the effect of three common calibration levels on the modeled tape machine and magnetic tape formulations With the evolution of tape formulations it was pos sible to increase the fluxivity level or magnetic strength of the signals on tape Over the years this resulted in an elevation of recorded levels relative to a standard reference fluxivity 185 nW m at 700 Hz The Cal Adjust value expresses the ele vated level in dB over this standard reference level The Cal Adjust control does not affect the overall gain but does affect the amount of saturation effect for a given input signal Available Cal Adjust values are 3 dB equivalent to 250 nW m 6 dB equivalent to 370 nW m 9 dB equivalent to 520 nW m The default value is 6 dB Chapter 47 Reel Tape Saturation 288 Reel Tape Saturation Tips Use Reel Tape Saturation on individual tracks to round out sharp transients or add color to sustained tones Use Reel Tape Saturation on a group of tracks for example drums to add cohesiveness to the sound of the group Use Reel Tape Saturation on a Master Fader to apply analog tape style compression to a mix Reel Tape Saturation Presets The sonic effect of Reel Tape Saturati
123. 00 Reset Analysis Click the Reset button to reset the analysis Pro Limiter Histogram and Loudness meters Time Reset Run Analysis Momentary Loudness elapsed Auto Analysis K Meter M S I meters Histogram Chapter 20 Pro Limiter 123 Auto Analysis When Auto is enabled Pro Limiter automatically pauses the analysis pass when the Pro Tools transport is stopped This means that the drop in levels will not be reported with the Inte grated and Range values Note that it may be useful to disable Auto if you want to include a live audio signal being monitored through Pro Tools while the transport is stopped Run Analysis Click the Run Analysis Play button to enable lit or disable unlit analysis reporting in the histogram while the Pro Tools transport is stopped Pro Limiter runs the analysis when the Pro Tools transport is running regardless of whether or not the Run Analysis option is enabled Note that when the Auto analysis option is enabled the Run Analysis option is overridden Loudness Meters The Loudness meters to the right of the histogram show the level of the summed output of Pro Lim iter The meters range from 0 LUFS down to 50 dB LUFS 23 LUFS is a common standard loudness reference level Momentary Loudness Provides a display of the loudness range as in the histogram The current peak level is shown as a yellow line using K weighted metering Momentary Loudness Meter M Graphically d
124. 10 Reel Tape Common Controls All Reel Tape plug ins share the following controls Drive Drive controls the amount of saturation effect by increasing the input signal to the modeled tape ma chine while automatically compensating by reduc ing the overall output Drive is adjustable from 12 dB to 12 dB with a default value of 0 dB Output Output controls the output signal level of the plug in after processing Output is adjustable from 12 dB to 12 dB with a default value of 0 dB Tape Machine The Tape Machine control lets you select one of three tape machine types emulated by the plug in each with its own sonic characteristics US Emulates the audio characteristics of a 3M M79 multitrack tape recorder Swiss Emulates the audio characteristics of a Studer A800 multitrack tape recorder Lo Fi Simulates the effect of a limited bandwidth analog tape device such as an outboard tape based echo effect Tape Formula The Tape Formula control lets you select either of two magnetic tape formulations emulated by the plug in each with its own saturation characteris tics Classic Emulates the characteristics of Ampex 456 exhibiting a more pronounced satura tion effect Hi Output Emulates the characteristics of Quantegy GP9 exhibiting a more subtle saturation effect Reel Tape Delay Controls In addition to the Drive Output Tape Machine and Tape Formula controls Reel Tape Delay has the following con
125. 4 Audio Plug Ins Guide iii Part I Introduction to Audio Plug Ins Chapter 1 Audio Plug Ins Overview 2 Plug In Formats 2 Avid Audio Plug Ins 2 Using Plug Ins in Pro Tools Software 4 Conventions Used in Pro Tools Documentation 5 System Requirements and Compatibility for Plug Ins 5 About www avid com 6 Chapter 2 Installing and Authorizing Avid Paid Plug Ins 7 Authorizing Avid Audio Plug Ins 7 Installing Plug Ins for Pro Tools 8 Removing Plug Ins 8 Chapter 3 Adjusting Plug In Co
126. 7 Band EQ III High Mid Peak The High Mid Peak band boosts or cuts frequencies centered around the Frequency setting The width of the band is determined by the Q setting The High Mid Gain control and its corresponding graph elements are displayed on screen in green The following control values are available Control Value Frequency Range 40 Hz to 1 kHz Frequency Default 200 Hz Low Mid Peak Q Range 0 1 to 10 0 Low Mid Peak Q Default 1 0 Low Mid Peak Gain Range 18 dB to 18 dB Mid Peak EQ Control Value Frequency Range 125 Hz to 8 kHz Frequency Default 1 kHz Mid Peak Q Range 0 1 to 10 0 Mid Peak Q Default 1 0 Mid Peak Gain Range 18 dB to 18 dB Frequency control Band Enable button Gain control Q control High Mid Peak EQ Control Value Frequency Range 200 Hz to 18 kHz Frequency Default 2 kHz Mid Peak Q Range 0 1 to 10 0 Mid Peak Q Default 1 0 Mid Peak Gain Range 18 dB to 18 dB Frequency control Band Enable button Gain control Q control Chapter 4 EQ III 24 7 Band EQ III High Shelf High Peak The High Shelf Peak band is switchable between high shelf EQ and high peak EQ functions By de fault this band is set to High Shelf High Shelf EQ Boosts or cuts frequencies at and above the Frequency setting The amount of boost or cut is determined by the Gain setting The Q set ting determines the shape of the shelving curve High Peak
127. Adjust the Harmonics fader and listen for the change in harmonics being added to the original audio signal 9 When finished experimenting set the Mix con trol to taste Keep in mind that a little Aural Ex citer goes a long way Using the Tune Fader After a while you ll get a sense of where you like your Tune setting when using Aural Exciter on in dividual tracks It s best not to process the same range of frequencies with the Tune fader during the final mix If you already processed individual tracks with Aural Exciter try starting the final mix with the Tune fader in the maximum position which is approximately 7 kHz You should get a spacious three dimensional mix with an open airy quality Using the SPR Switch The SPR function can produce a useful effect with solo voices human and instrumental or mixed programs such as drama and music There is no specific time when SPR should or shouldn t be used Experiment with it on various types of mate rial until you get used to the effect Listen carefully as you operate the SPR switch The effect may be noticed only at certain times such as specific mod ulations of a voice or during a particular instrumen tal playing style or passage Don t expect to hear the sound change radically The SPR is usually subtle adding a certain beauty and good feeling to the sound In time you will find that the SPR does indeed produce demonstrable results For instance the SPR ef
128. Audio Plug Ins Guide Version 11 2 Legal Notices 2014 Avid Technology Inc Avid all rights reserved This guide may not be duplicated in whole or in part without the written consent of Avid 003 192 Digital I O 192 I O 96 I O 96i I O Adrenaline AirSpeed ALEX Alienbrain AME AniMatte Archive Archive II Assistant Station AudioPages AudioStation AutoLoop AutoSync Avid Avid Active Avid Advanced Response Avid DNA Avid DNxcel Avid DNxHD Avid DS Assist Station Avid Ignite Avid Liquid Avid Media Engine Avid Media Processor Avid MEDIArray Avid Mojo Avid Remote Response Avid Unity Avid Unity ISIS Avid VideoRAID AvidRAID AvidShare AVIDstripe AVX Beat Detective Beauty Without The Bandwidth Beyond Reality BF Essentials Bomb Factory Bruno C 24 CaptureManager ChromaCurve ChromaWheel Cineractive Engine Cineractive Player Cineractive Viewer Color Conductor Command 8 Control 24 Cosmonaut Voice CountDown d2 d3 DAE D Command D Control Deko DekoCast D Fi D fx Digi 002 Digi 003 DigiBase Digidesign Digidesign Audio Engine Digidesign Development Partners Digidesign Intelligent Noise Reduction Digidesign TDM Bus DigiLink DigiMeter DigiPanner DigiProNet DigiRack DigiSerial DigiSnake DigiSystem Digital Choreography Digital Nonlinear Accelerator DigiTest DigiTranslator DigiWear DINR DNxchange Do More DPP 1 D Show DSP Manager DS StorageCalc DV Toolkit DVD Complete
129. C in the middle of the keyboard is selected there is no pitch transposition Click any other key to transpose the pitch of the incoming signal by the interval difference between middle C and the se lected key For example if the E flat key above middle C is selected Pitch II transposes the incom ing signal up a minor third or 3 semitones The Coarse control and the Keyboard control are linked Coarse This control adjusts the pitch of a signal in semitones over a two octave range Pitch changes are indicated in number of semitones Fine This control controls the pitch of a signal in cents hundredths of a semitone over a 100 cent range The range of this control is 50 to 50 cents Ratio The Ratio control lets you set the transposi tion between the pitch of the incoming signal and the selected transposition value as a percentage The Ratio setting is linked with the Coarse and Fine settings Link Stereo Only Enable the Link option to link the controls for the left and right channels Pitch Shift controls Chapter 23 Pitch II 136 Show Hide Panel Click the Show Hide triangle in the upper left cor ner of the Pitch Shift panel to show or hide the panel Hiding the Pitch Shift panel can be useful for conserving screen space Effects Controls Delay The Delay control lets you set the delay time between the original signal and the pitch shifted signal It has a maximum setting of 1000 millisec onds You can use the
130. Chapter 30 Space 181 Chapter 30 Space Space is an AAX format convolution reverb plug in that is available in DSP Native and AudioSuite for mats Space was designed to be the ultimate reverb for music and post production applications By combin ing the sampled acoustics of real reverb spaces with advanced DSP algorithms Space offers stunning real ism with full control of reverb parameters in mono stereo and surround formats Space supports 44 1 kHz 48 kHz 88 2 kHz and 96 kHz sample rates Space works with mono stereo and mono to stereo formats With Pro Tools HD Space also supports Quad 5 0 mono to Quad stereo to Quad mono to 5 0 and stereo to 5 0 multichannel formats Space plug in Chapter 30 Space 182 Space Feature Highlights Space features let you create the best reverb effect in the shortest possible time Reverb Features Mono Stereo Quad and 5 0 channel output support Multiband EQ Independent wet dry and decay levels Separate reverb early and late levels and length Control of early size low cut and balance Pre delay and late delay controls Precise control of low mid and high decay crossover Adjustable waveform reverse displayed in beats per minute Waveform processing bypass Interface Features Full waveform view zoom and channel high light functions On screen input and output metering with clip indicators Impulse respo
131. Control The Range control defines the maximum amount of gain reduction possible when a signal is detected at the frequency set by the Frequency control This control ranges from 40 dB maximum de essing to 0 dB no de essing De Esser III HF Only Control When the HF Only button is enabled gain reduc tion is applied only to the active frequency band set by the Frequency control When the HF Only but ton is disabled the De Esser applies gain reduction to the entire signal De Esser III Listen Control When enabled the Listen button lets you monitor the sibilant peaks used by the De Esser as a side chain to trigger compression This is useful for lis tening only to the sibilance for fine tuning De Es ser controls To monitor the whole output signal without this filtering deselect the Listen button De Esser III Frequency Graph Display The De Esser Frequency Graph display shows a curve that represents the level of gain reduction on the y axis for the range of the output signal s fre quency on the x axis The white line represents the current Frequency setting and the animated or ange line represents the level of gain reduction be ing applied to the signal Use this graph as a visual guideline to see how much dynamics processing you are applying at dif ferent points in the frequency spectrum De Esser graph display Frequency x axis Gain y axis Frequency Range Current gain reduction Ch
132. Delay Early Length Early Processing Late Processing Low Decay Mid Decay High Decay Crossover Late Delay Early Size Early Low Cut Early Balance Convolution Processor Pre Delay Chapter 30 Space 187 Impulse Response IR and Space This section covers aspects of impulse response IR and Space IR Processing Control Lag Adjusting some controls in Space requires the im pulse computer to recalculate the waveform and re load it into the convolution processor This opera tion uses DSP and host processing capacity When this occurs some control lag may be experienced This should be kept in mind if controls are being automated in real time during a session How Impulse Responses Are Captured An IR of an actual physical space is captured using a combination of an impulse sound source and cap ture microphones The sound source is used to ex cite the physical space to create a reverb and can be a starter pistol or a frequency tone played through a speaker The microphones can be placed in various configurations The resulting IR is then processed to create a digital representation of both the physi cal space potentially colored by the sound source and the type of microphone used Similarly an IR of a hardware effects unit can be captured by sending a test pulse through the unit and capturing the result digitally In addition to re flecting reverb or delay characteristics an IR also reflects t
133. Delay control in conjunction with the Feedback control to generate a single pitch shifted echo or a series of echoes that rise in pitch Mix The Mix control lets you adjust the ratio of dry signal to effected signal in the output In general this control should be set to 100 wet unless you are using the plug in in line on an Insert for an in dividual track or element in a mix This control can be adjusted over its entire range with little or no change in output level Feedback The Feedback control lets you set the amount and type of feedback positive or negative applied from the output of the Delay effect back into its input It also controls the number of repeti tions of the delayed signal You can use it to pro duce effects that spiral up or down in pitch with each successive echo shifted in pitch LPF The LPF Low Pass Filter control lets you set the frequency under which audio signal is passed The control can be set between 10 Hz and 22 05 kHz Show Hide panel controls Effects controls Show Hide Chapter 24 Time Shift 137 Chapter 24 Time Shift Time Shift is an AudioSuite plug in that provides high quality time compression and expansion TCE algorithms and formant correct pitch shift ing Time Shift is ideal for music production sound de sign and post production applications Use it to manipulate audio loops for tempo matching or to transpose vocal tracks using formant correct pitch shifting You ca
134. Drive settings emphasize power amp distortion see Plexi preset Part IX Dither Plug Ins Chapter 49 Dither 292 Chapter 49 Dither Dither is a dither generation plug in The Dither plug in minimizes quantization artifacts when re ducing the bit depth of an audio signal to 16 18 or 20 bit resolution Dither is available in DSP and Native formats Dither supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Dither operates as a mono multi mono or stereo plug in Whenever you are mixing down or bouncing to disk and your destination bit depth is lower than 24 bit insert a dither plug in on a Master Fader track that controls the output mix Using a dither plug in on a Master Fader is prefer able to an Auxiliary Input because Master Fader in serts are post fader As a post fader insert the dither plug in can process changes in Master Fader level The Dither plug in has user selectable bit resolu tion and a noise shaping on off option Dither Controls The Dither plug in has a Bit Resolution button and a Noise Shaping button Bit Resolution Button Use this pop up menu to choose one of three possi ble resolutions for the Dither processing Set this control to the maximum bit resolution of your des tination 16 bit Recommended for output to digital devices with a maximum resolution of 16 bits such as DAT and CD recorders 18 bit Recommended for output to digital devic
135. Gain 1 range is from 0 to 10 Gain 2 Gain 2 is a second Gain knob used with some amp models that determines the amount of overdrive in the pre amp stage Gain 2 also known as Pres ence on some amps allows for more harmonic subtleties in character of the amp tone The default is 5 0 Gain 2 range is from 0 to 10 Amp controls in the default Amp Type Bypass Gain 1 Bright Treble Mid Bass Presence Master Volume Amp Tone Speed Depth Tremolo All Eleven controls provide identical ranges as the original amps but some numbers have been adjusted for consistency Chapter 44 Eleven 267 Parallel or Series The Gain 2 control on the Tweed Lux AC Hi Boost and Plexiglass is in parallel jumped with the Gain 1 control The M 2 Lead is in series meaning the signal goes in and out of Gain 1 then into Gain 2 Tone Tone controls let you shape the highs mids and lows of the amp sound Electric guitar pickups tend to have a strong low mid emphasis and little high frequency response often producing a mid range heavy sound that requires some treble boost The response and interaction of the tone controls are unique to each amp Bass The Bass control determines the amount of low end in the amp tone The response of this control in some models is linked to the setting of the Treble control The default setting is 5 0 Bass range is from 0 to 10 Middle The Middle control determines the mid range stre
136. Graph You can use the Reverb Color graph to shape the tonal spectrum of the reverb by controlling the de cay times of the different frequency bands Low and high crossover points define the cut and boost points of three frequency ranges For best results set crossover points at least two oc taves higher than the frequency you want to boost or cut For example to boost a signal at 100 Hz set the crossover to 400 Hz Set the crossover to 500 Hz to boost low frequen cies most effectively Set it to 1 5 kHz to cut low frequencies most effectively Crossover Sliders Sets the frequency boundaries between the low mid and high frequency ranges of the reverberation filter The low frequency slider sets the crossover fre quency between low and mid frequencies in the re verberation filter The adjustable range is from 60 0 Hz to 22 5 kHz The high frequency slider sets the crossover fre quency between mid and high frequencies in the re verberation filter The adjustable range is from 64 0 Hz to 24 0 kHz Band Breakpoints Controls cut and boost ratios for the decay times of the low mid and high fre quency bands of the reverberation filter To cut a frequency band drag a breakpoint downward To boost drag it upward The adjustable range is from 1 8 to 8 1 HF Damp Breakpoint Sets the frequency above which sounds decay at a progressively faster rate This determines the decay characteristic of the high frequency components o
137. Hz to 18 kHz The lower rotary control adjusts the filter s amplitude gain or attenuation Amplitude range is 15 dB from unity Low Pass Filter The 18 dB octave Low Pass Filter provides a ro tary control for adjusting the filter s cutoff fre quency variable from 100 Hz to 18 kHz Enabling Disabling and Bypassing EQ Filters You can enable disable or bypass specific EQ fil ters by clicking them To disable a filter Control click Mac or Start click Windows the EQ Filter icon When disabled the icon is black To re enable a filter Click the EQ filter icon When enabled the icon is white To bypass a filter Click the EQ filter icon a second time When by passed the icon is gray High Shelf Filter Low Pass Filter If you are using all available bands of the 1 2 Band or 4 Band EQ and want to change fil ter types you must disable one filter before you can enable a different one Chapter 5 Focusrite D2 31 Using D2 in Stereo Because Focusrite D2 has a single set of Filter con trol knobs when it is used in stereo you must select which channel left or right you want to edit Left Channel and Right Channel Buttons The Left Channel and Right Channel buttons are used to select which controls are active Link Button The Link button lets you adjust controls for both channels simultaneously By default Link mode is enabled so that you can maintain parity between cha
138. In the process of achieving such a hot mix unwanted distortion can be intro duced Intersample peaks that exceed 0 dB may play without distortion in a studio environment but when the same mix is played through a consumer CD player the digital to analog conversion and oversampling process can reproduce a distorted mix Digital Audio Theory A key observation in digital audio theory is that the entire waveform is represented by the sampling points but a reconstruction process still needs to occur in order to recreate the waveform repre sented One cannot simply connect the dots be tween sample points and yield the original wave form A waveform can be represented in multiple ways during the process of sampling display and recon struction The following four figures show how the same complex waveform shown in the previous figure can be represented in the digital domain The process of recreating the original waveform from the sampled waveform involves a filter called a reconstruction filter This filter removes all con tent above the Nyquist frequency half the sample Sampling A complex waveform Waveform sampled Waveform as represented in DAW Waveform as reconstructed at the D A Chapter 56 MasterMeter 327 rate The range below the Nyquist frequency de fines the legal range of allowed frequencies as frequencies in this range can be accurately repro duced All frequencies above the Nyquist
139. Increasing the setting produces a harder heavier distortion At maximum it produces a sound similar to a wah pedal at mid boost position placed in front of a Marshall amp Crunch Brings out upper harmonic content and on guitars pick attack For cleaner sounds or smoother high end decrease as needed Drive Increases the amount of power amp distortion Power amp distortion is associated with the Vin tage Marshall sound using SansAmp you can produce the effect even at low levels Low Provides a tone control specially tuned for maxi mum musicality when used to EQ low frequencies on instruments Boost or cut by 12 dB by turning from the center point indicated by the arrows High Boosts or cuts high frequencies by 12 dB Level Boosts or cuts the overall gain to re establish unity after adding distortion or equalizing the signal PSA 1 Tips and Tricks Peace and Unity The arrows in the SansAmp controls indicate the unity gain position Louder and Cleaner For best results don t set the Pre Amp level lower than unity gain when the Drive knob is at 9 o clock or higher However if you want a crystal clear sound and the Drive control is already near mini mum decrease Pre Amp to further remove distor tion Pre Amp Versus Drive To create varying types of overdrive vary Pre Amp in relation to Drive A high Pre Amp set ting emphasizes pre amp distortion see Mark 1 preset while high
140. Levels Section Controls The Levels section has controls for adjusting source input and ReVibe II output levels ReVibe II provides individual output level controls for Front Center Rear reverb and Rear early reflections In stereo and greater than stereo formats where there is no center channel or where there are no rear channels the center and rear level controls can be used to augment the reverb sound Reverb and early reflections that would be heard either from the center channel or from the rear channels can be mixed into the front left and right channels Input Control Input adjusts the level of the source input to prevent internal clipping The range of this control is from 24 0 dB to 0 0 dB Lowering the Input control does not change the levels shown on the input side of the Input Output meter which shows the level of the signal before the Input control Front Control Front controls the output level of the front left and right outputs Front is also the main level control for stereo The range of this control is from INF minus infinity to 0 0 dB Center Control Center controls the output level of the center chan nel outputs of multichannel formats that have a center channel such as LCR or 5 1 When ReVibe II is used in a multichannel format that has no center channel such as stereo or quad the Center level control adjusts a phantom center channel signal that is center panned to the front left and rig
141. Medium Bright Room 1 Medium Bright Room 2 Medium Bright Room 3 Medium Neutral Room 1 Medium Neutral Room 2 Medium Neutral Room 3 Medium Dark Room 1 Medium Dark Room 2 Medium Dark Room 3 Chapter 29 ReVibe II 178 Small Bright Room 1 Small Bright Room 2 Small Bright Room 3 Small Neutral Room 1 Small Neutral Room 2 Small Neutral Room 3 Small Dark Room 1 Small Dark Room 2 Small Boomy Room Halls Large Natural Hall 1 Large Natural Hall 2 Large Natural Hall 3 Large Natural Hall 4 Large Natural Hall 5 Large Natural Hall 6 Large Dense Hall Large Sparse Hall Medium Natural Hall 1 Medium Natural Hall 2 Medium Natural Hall 3 Medium Natural Hall 4 Medium Dense Hall Small Natural Hall 1 Small Natural Hall 2 Theaters Large Theater 1 Large Theater 2 Medium Theater 1 Medium Theater 2 Small Theater 1 Small Theater 2 Churches Large Natural Church 1 Large Natural Church 2 Large Dense Church Large Slap Church Medium Natural Church 1 Medium Natural Church 2 Medium Dense Church Small Natural Church 1 Small Natural Church 2 Cathedrals Natural Cathedral 1 Natural Cathedral 2 Natural Cathedral 3 Dense Cathedral 1 Dense Cathedral 2
142. Pass option to apply a band pass filter to the side chain processing at the selected frequency Side Chain Processing Graph The Side Chain Processing Graph display shows the frequency curve for the selected Filter Type at the selected Filter Frequency Side Chain Filter enabled Chapter 20 Pro Limiter 117 Chapter 20 Pro Limiter Avid Pro Limiter is available in DSP Native and AudioSuite formats Pro Limiter provides true peak limiting Pro Limiter limits incoming audio to the True Peak of the signal to prevent inter sample peaks that could introduce distortion during encod ing or analog conversion Pro Limiter complies with the ITU R BS 1770 3 loudness metering standard including True Peak Integrated Loudness and Loudness Range mea surements and is suitable for both EBU R128 and ATSC A 85 CALM Act broadcast workflows Pro Limiter also provides a unique Character knob that lets you add soft saturation for more loudness and greater gain reduction without the unwanted digital artifacts of standard brick wall limiters Use Pro Limiter to ensure that your mix output never exceeds digital 0 dB when hitting the digital to analog converters on your audio interface Pro Limiter supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Pro Limiter supports mono stereo and greater than stereo multichannel formats up to 7 1 Greater than stereo formats are only available with Pro Tools HD P
143. Pro Tools 1 In Pro Tools create a new MIDI track 2 From the track s MIDI Input selector select the ReWire device that you want to record 3 Record enable the MIDI track 4 Start recording in Pro Tools 5 Switch to the ReWire client application 6 Adjust the control for which you want to record MIDI CC data Control changes are recorded to the Pro Tools MIDI track as CC data 7 When you are done adjusting the control return to Pro Tools and stop recording 8 Record disable the MIDI track 9 From the MIDI Track View selector in the Edit window select the view for the CC data you just recorded Selecting the ReWire client device to record MIDI CC data in Pro Tools You must select the ReWire device from which you want to record MIDI controller data Leaving the track s MIDI Input set to All does not record any MIDI data over ReWire Adjusting a control in a ReWire client application Reason s SubTractor shown If your external MIDI controller is correctly mapped to the corresponding ReWire client application s controls and it is correctly routed through Pro Tools use your MIDI controller to adjust the parameter you want to record MIDI CC data recorded from a ReWire client application Chapter 54 ReWire 316 Playing Back MIDI Continuous Controller Data Over ReWire Once you have recorded MIDI CC data from the ReWire client application to a MIDI track config ure the MIDI track to play the ReWi
144. Pro effect Drive control adjustments will be reduced when the AutoTrace switch is activated Link Switch The Link switch is for stereo operation only It links the left and right controls so they work as one Grab a control on one page with the cursor and move it to the desired position The control on the other page automatically updates In this way both controls can be set to the exact same position Ste reo controls may be linked temporarily by holding down the Shift key while adjusting the control The switch illuminates when Link is activated LR Left Right Switches The LR switch is for stereo operation only It al lows you to view or change parameters on one channel at a time The switch for the currently displayed channel illu minates Clicking the unlit switch changes the dis play to the other channel To edit both channels simultaneously click on the Link switch Chapter 43 Aphex Big Bottom Pro 254 Using Big Bottom Pro By putting Big Bottom Pro to use in a Pro Tools session you will find many creative uses for its powerful processing capabilities The remaining sections provide instructions on how to get the most out of Big Bottom Pro Setting the Gain Structure If the amount of Big Bottom Pro effect is limited by a lack of headroom in the input material use the Level control to adjust the signal level to avoid clipping When using Big Bottom Pro with the Phase switch in the Off position it is p
145. Record enable the track or enable TrackInput monitoring Pro Tools HD only and check your levels 6 When you re ready arm the Pro Tools Transport and press Record to record your part The audio that is recorded is the dry unprocessed signal only while playback of the recording is pro cessed through Eleven and any other plug ins in serted on the track Recording Wet Record Eleven Processed Track to Disk In this workflow the audio output of Eleven is re corded to disk while tracking Usually no addi tional dry track is recorded This method commits your track to the original Eleven tone used while tracking It requires two tracks an Auxiliary Input and an audio track but after tracking the plug in can be deactivated or re moved to up processing resources To record guitar with Eleven while playing 1 Choose Track gt New 2 Configure a new track by doing the following Create one mono Auxiliary Input track Click the Add Row button Create one mono audio track Click Create 3 In the Mix or Edit window configure the Aux Input by doing the following Click the Input selector and choose your guitar input the audio interface input your guitar is plugged in to Click the Output selector and choose Bus 1 Click the Insert selector and select Eleven 4 Configure the audio track by doing the follow ing Click the Input selector and choose Bus 1
146. Room and Reverb type also incorporates a complex room coloration EQ which models the general frequency response of various rooms and effects devices Choosing a new Reverb Type changes the early re flections and room coloration EQ only All of the other ReVibe II settings remain unchanged To cre ate a preset that includes all parameters use the Plug In Settings menu The Reverb Type display shows the Room Type Category Room Type Name the Next and Previ ous buttons and the Reverb Type Input and Output meters 5 1 For more information on saving and import ing plug in presets see the Pro Tools Refer ence Guide Reverb Type display and controls Room Type Category menu Room Type Name menu Preset Next and Previous buttons Reverb Type menu Chapter 29 ReVibe II 170 Room Type Category Menu Clicking on the Room Type Category menu lets you select one of the 14 Room Type categories and selects the first Room Type preset in that category Room Type Name Menu Click the Room Type Name menu to select from a list of all available Room Type presets Next and Previous Buttons Click the Next or Previous buttons to choose the next or previous Room Type Reverb Type Menu Click the Reverb Type menu to select the type of reverb tail There are nine basic reverb types plus Automatic Select Automatic to use the reverb tail type that is stored with the currently selected room type The reverb types are Au
147. SoundReplacer 335 Time Compression Expansion 343 Aural Exciter plug in 241 Ax switch 246 Bypass switch 246 Density switch 245 Drive meter 243 Drive switch 245 gain structure 247 Harmonics control 244 Level control 243 Link switch 246 LR Left Right switches 247 Mix control 245 Null Fill control 244 optimizing 247 Out meter 243 Peaking control 243 Solo switch 246 SPR switch 246 248 Timbre control 245 Tune control 243 Tune fader 248 AutoPan plug in Angle slider 298 Attack slider 302 Beat Clock trigger 301 Duration selector 301 ENV 299 Envelope section 302 Envelope trigger 301 LFO 299 300 LFO triggers 300 Link To Tempo option 301 Manual slider 298 Manual trigger 301 Output meters 297 Output slider 298 Panner section 298 Panning display 299 panning examples 302 Panning Field indicator 299 Path selectors 299 Place slider 298 Rate slider 300 Release slider 302 Side Chain Input option 303 Side Chain Input selector 302 Single trigger 300 Sound Location indicator 299 Spread slider 299 surround panning 303 synchronizing tempo 302 Tempo controls 301 Tempo display 302 Threshold slider 302 Waveform selector 300 Width slider 298 Index Audio Plug Ins Guide 351 B BF 2A plug in 37 side chain processing 38 BF 3A plug in 40 side chain processing 41 BF76 plug in 42 side chain processing 43 Big Bottom Pro plug in 250 AutoTrace switch 253 Compression meter 251 Drive control 252 Drive meter 251 gain structure 254 I
148. T If you are using Maxim on a Master Fader during mixdown Maxim s built in dither function saves you the trouble and DSP resources of having to use a separate Dither plug in If Dither is disabled the Noise Shaping and Bit Resolution controls will have no effect Maxim Noise Shaping Control When selected this applies noise shaped dither Noise shaping biases the dither noise to less audible high frequencies so that it is not as readily per ceived by the ear Dither must be enabled in order to use Noise Shaping Maxim Bit Resolution Button These buttons select dither bit resolution In gen eral set this control to the maximum bit resolution of your destination media 16 bit is recommended for output to digital de vices such as DAT recorders and CD recorders since they have a maximum resolution of 16 bits 18 bit is recommended for output to digital de vices that have a maximum resolution of 18 bits 20 bit is recommended for output to digital de vices that support a full 20 bit recording data path It is recommended for use with digital effects de vices that support 20 bit input and output since it provides for a lower noise floor and greater dy namic range when mixing 20 bit signals directly into Pro Tools Chapter 17 Maxim 96 Using Maxim Following are suggestions for using Maxim most effectively To use Maxim 1 Insert Maxim on a track 2 Select the portion of the track containing the
149. Take your time to explore the Presets let you hear all of Eleven s different amps and combos 3 Pick any amp and cabinet from the available types see Pairing Amps and Cabinets on page 268 4 Refer to Using Eleven on page 261 for details on Eleven s main controls and for suggested track setups for recording and mixing Using Eleven The following sections introduce you to the main sections and controls in Eleven and show you how to use them You ll also find suggested track setups and signal routing tips to help you get the most out of Eleven Inserting Eleven on Tracks Eleven can be inserted on Pro Tools audio Auxil iary Input Master Fader or Instrument tracks To insert Eleven on a track Click an Insert selector on the track and choose Eleven or Eleven LE Channel Formats Eleven is available as a mono or multi mono plug in only For use in stereo or greater formats choose the multi mono version Sample Rates Eleven supports 44 1 kHz 48 kHz 88 2 kHz and 96 kHz sample rates Category and Manufacturer When Pro Tools plug ins are organized by Category or Manufacturer Eleven is listed as follows Category Harmonic Manufacturer Avid Librarian menu left and the Settings menu right Use the Settings menu to save copy paste and manage plug in settings files To save a setting see Eleven Settings Presets on page 263 Chapter 44 Eleven 262 Adjusting Elev
150. ach channel De lay Modulation and Mix controls for stereo and mono to stereo instances of Mod Delay III can be linked or can be operated independently Input Input Meters The Input meters show peak signal levels before processing Dark Blue Indicates nominal levels from INF to 12 dB Light Blue Indicates pre clipping levels from 12 dB to 0 dB Red Indicates clipping Phase Invert The Phase Invert button at the top of the Input sec tion inverts the phase polarity of the input signal to help compensate for phase anomalies that can occur either in multi microphone environments or because of mis wired balanced connections To enable or disable phase inversion on input Click the Phase Invert button so that it is high lighted Click it again so that it is not highlighted to disable it Mod Delay III Chapter 31 Mod Delay III 205 Delay Link For stereo and mono to stereo tracks enable the Link button to link the Delay Modulation and Mix controls between the Left and Right channels This option is highlighted when it is enabled For mono tracks this option reads Mono and is dis play only Delay Time The Delay Time control sets the delay time be tween the original signal and the delayed signal from 0 0 ms to 5 000 0 ms Feedback FBK The Feedback setting controls the amount of feed back applied from the output of the delay back into its input from 100 to 100 It
151. ailable DSP re sources Black indicates disabled In this state the com pressor is not using DSP resources Gray indicates bypassed In this state the com pressor is not active but is still using available DSP resources To disable the compressor Control click Mac or Start key click Win dows the icon When the compressor is dis abled the icon is black To re enable the compressor Click the icon When the compressor is enabled the icon is white To bypass the compressor Click the icon a second time When the com pressor is bypassed the icon is gray If you are using the Compressor Limiter plug in which allows you to use either the compressor or the limiter but not both simultaneously you must disable one module by Control clicking Mac or Start clicking Windows the icon before you can enable the other Meters Compressor icon Compressor controls Chapter 14 Focusrite D3 82 Ratio Ratio sets the compression ratio If the ratio is set to 2 1 for example it will compress changes in signals above the threshold by one half The range of this control is from 1 5 1 very little compression to 10 1 heavy compression bordering on limiting Threshold Threshold sets the threshold level Signals that ex ceed this level will be compressed Signals that are below it will be unaffected The range of this con trol is from 0 dB to 48 dB A setting of 0 dB is equivalent
152. al that you choose the option most appropriate to the material that you are replacing SoundReplacer Mix Mix adjusts the mix of the replacement audio file with the original source file Higher percentage val ues weight the mix toward the replacement audio Lower percentage values weight the mix toward the original source audio The Mix button toggles the Mix control on and off When Mix is toggled off the balance is instantly set to 100 replacement audio Zoomer If you zoom the waveform display below a specific Threshold slider s amplitude zone the slider will be temporarily unavailable To access the slider again zoom back out to an appropriate magnification level For more information on using Peak Align see Getting Optimum Results with SoundReplacer on page 340 Setting Mix to 50 and clicking Preview lets you audition source audio and replacement audio together to check the accuracy of re placement triggering timing Chapter 58 SoundReplacer 339 SoundReplacer Dynamics Dynamics controls how closely the audio events in the replacement file track the dynamics of the source file Setting the ratio to 1 00 matches the dynamics of the source file Increasing the ratio above 1 00 expands the dy namic range so that softer hits are softer and louder hits are louder This is useful if the source material lacks variation in its dynamic range Decreasing the ratio below 1 00 compresses the d
153. also controls the number of repetitions of the delayed signal Nega tive feedback settings give a more intense tunnel like sound to flanging effects Low Pass Filter LPF The Low Pass Filter setting controls the cutoff fre quency of the Low Pass Filter from 10 Hz to 22 kHz Use the LPF setting to attenuate the high frequency content of the feedback signal The lower the setting the more high frequencies are at tenuated The maximum value for LPF is Off This lets the signal pass through without limiting the bandwidth of the plug in Sync When Sync is enabled and a Duration a rhythmic note value is selected the Delay Time setting is af fected by the session tempo When Sync is dis abled and a Duration is selected the Delay Time setting is affected by changes to the Tempo setting When Tempo Sync is enabled the Tempo and Me ter controls are uneditable and follow the session tempo and meter changes in the Pro Tools timeline The Duration and Groove controls apply regardless of whether Sync is enabled Meter The Meter setting lets you enter either simple or compound time signatures The Meter control de faults to a 4 4 time signature When Sync is enabled the Meter control is un available Tempo The Tempo control sets the tempo in beats per min ute from 5 00 to 500 00 bpm This setting is inde pendent of the Pro Tools session tempo When a specific Duration is selected moving this control affec
154. an nels and maintains phase coherency within those selected channels AudioSuite Preview X Form supports Pro Tools AudioSuite Preview and Bypass For more information on using Audio Suite Preview and Bypass see the Pro Tools Refer ence Guide AudioSuite TCE Plug In Preference The high quality time compression and expansion algorithms of X Form can be used with the Pro Tools TCE Trim tool To select X Form for use with the TCE Trim tool 1 Choose Setup gt Preferences 2 Click the Processing tab 3 From the TC E Plug In pop up menu select Avid X Form 4 Select a preset from the Default Settings pop up menu 5 Click OK TCE Plug In option in Processing Preferences page When using X Form for the TCE Trim tool the default 2x Range is used for an edit range of twice to half the duration of the original audio If you select a Default Setting that uses either the 4x or 8x Range the Time Shift and Pitch Shift setting are constrained to the 2x Range limit of 50 to 200 Refer to the Pro Tools Reference Guide for more information about the TCE Trim tool Chapter 26 X Form 152 Processing Audio Using X Form X Form lets you change the time and pitch of se lected audio independently or concurrently To change the time of a selected audio clip 1 Select AudioSuite gt Pitch Shift gt X Form 2 Select the Audio Type appropriate to the type of material you are processing Monophonic or Polyphonic
155. ange the detection mode for the expander Click a detection mode from the options avail able above the Dynamics graph Detection options include the following Smart Select the Smart option for tracks with di verse input signals or if you are simply not sure what detector works best with the given audio ma terial The Smart option analyzes the incoming sig nal and switches between the different detection modes as needed RMS Select the RMS option to apply processing ac cording to the detected RMS Root Mean Square amplitude of the input signal The RMS option is similar to the Average option but with a faster re lease time Average Select the Average option to apply pro cessing according to the detected average ampli tude of the input signal Peak Select the Peak option to apply processing according to the detected peak amplitude of the input signal Fast Select the Fast option for tracks with high and short transient signals such as a snare drum track The Fast option is similar to the Peak option but with faster attack and release times However be careful when using the Fast option as it distorts ear lier than the other options Be sure to configure the other compressor settings with this in mind Duck Select the Duck option to use Pro Expander as a ducker In order to use Pro Limiter as a ducker set up an external key input side chain bussed from another track When the Duck option is enabled the incomin
156. ant to create a new file that connects and consolidates all of these clips together choose Create Continuous File from the File mode pop up menu 16 From the Destination Track pop up choose the destination for the replacement audio 17 Click Render Getting Optimum Results with SoundReplacer Getting optimum results with SoundReplacer gen erally means making sure that the audio events in the replacement audio file have accurate timing in relation to the source audio The techniques given here help ensure this Using Peak Align in SoundReplacer Proper use of the Peak Align feature can signifi cantly improve the results of sound replacement Since turning Peak Align on or off controls how SoundReplacer aligns the replacement audio with the source audio it will significantly affect the tim ing of audio events in the replacement file In general Turn on Peak Align if you are replacing drum or percussion sounds whose peak level occurs at the initial attack Turn off Peak Align if you are replacing sounds whose peak level occurs somewhere after the ini tial attack Peak Align should also be turned off if the sounds you are replacing are not drum or per cussion sounds Because SoundReplacer does not allow destructive rendering the AudioSuite Overwrite Files option is not available Chapter 58 SoundReplacer 341 To illustrate why Peak Align makes a difference look at the following two illustrations Th
157. apply a low pass filter to the side chain processing at the selected frequency High Pass Select the High Pass option to apply a high pass filter to the side chain processing at the selected frequency Notch Select the Notch option to apply a notch filter to the side chain processing at the selected frequency Band Pass Select the Band Pass option to apply a band pass filter to the side chain processing at the selected frequency Side Chain Processing Graph The Side Chain Processing Graph display shows the frequency curve for the selected Filter Type at the selected Filter Frequency Side Chain Filter enabled Note that the side chain filter does not apply filtering to the compressed signal Compres sion is applied to all frequencies of the input signal when compression is triggered by the side chain Chapter 19 Pro Expander 107 Chapter 19 Pro Expander Avid Pro Expander is available in DSP Native and AudioSuite formats Pro Expander provides dy namic expansion and gating processing The Avid Pro Expander processing algorithms are based on the award winning Euphonix System 5 console channel strip effects Pro Expander supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Pro Expander supports mono stereo and greater than stereo multichannel formats up to 7 1 In addition to standard knob and fader controls Pro Expander also provides a dynamics graph to track the gain transfer cu
158. apter 12 Dynamics III 70 Dynamics III Side Chain Input Compressor Limiter and Expander Gate Only Dynamics processors typically use the detected amplitude of their input signal to trigger gain re duction This split off signal is known as the side chain The Compressor Limiter and Ex pander Gate plug ins feature external key capabili ties and filters for the side chain With external key side chain processing you trig ger dynamics processing using an external signal such as a separate reference track or audio source instead of the input signal This external source is known as the key input With side chain filters you can make dynamics processing more or less sensitive to certain fre quencies For example you might configure the side chain so that certain lower frequencies on a drum track trigger dynamics processing Dynamics III Side Chain Controls The controls in the Side Chain section let you tog gle the side chain between the internal input signal or an external key input listen to the side chain and tailor the equalization of the side chain signal so that the triggering of dynamics processing be comes frequency sensitive Dynamics III Side Chain External Key The External Key toggles external side chain pro cessing on or off When this button is highlighted the plug in uses the amplitude of a separate refer ence track or external audio source to trigger dy namics processing When this button is dark g
159. ase shift not the tone shift Our ears are very sensitive to phase and using the two knobs together you can adjust phase at the low end while also making tonal adjustments On the high end you can set Boost to 10k and Atten to 10k then adjust Boost and Atten simultaneously However because Boost is a peak equalizer and Atten is a shelving equalizer the results are much different and you don t get independent control of phase Q and Boost In the high frequency boost section the Bandwidth and Boost controls affect one another This is dif ferent from modern equalizers where adjusting Q typically doesn t affect the amount of equalization applied Guitars Have multiple guitars that sound like mush in the mix The Pultec MEQ 5 is a classic tool for achiev ing amazing guitar blends Try boosting one guitar and cutting another to achieve an octave of separa tion For example cut one guitar using 1 5 1500 Hz Dip then boost the other using 3 3000 Hz Peak View the matched pairs of presets such as Guitar 1A and 1B or 2A and 2B for fur ther examples of this technique Part III Dynamics Plug Ins Chapter 8 BF 2A 37 Chapter 8 BF 2A BF 2A is a vintage style compressor plug in that is available in DSP Native and AudioSuite formats BF 2A supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates BF 2A operates as a mono multi mono or stereo plug in Designed and manufact
160. ased echo effect Tape Formula The Tape Formula control lets you select either of two magnetic tape formulations emulated by the plug in each with its own saturation characteris tics Classic Emulates the characteristics of Ampex 456 exhibiting a more pronounced satura tion effect Hi Output Emulates the characteristics of Quantegy GP9 exhibiting a more subtle saturation effect Reel Tape Saturation Controls In addition to the Drive Output Tape Machine and Tape Formula controls Reel Tape Saturation has the following controls Speed The Speed control adjusts the tape speed in ips inches per second Tape speed affects the fre quency response of the modeled tape machine Available tape speeds include 7 5 ips 15 ips and 30 ips with a default setting of 15 ips Noise Reel Tape Saturation produces noise only during playback and recording and not when the transport is stopped The Noise control adjusts the level of simulated tape noise that is added to the processed signal The characteristics of the noise depend on the Speed Bias and Tape Machine settings and the relative level of the noise depends on the Drive Cal Adjust and Tape Formula settings Noise is adjustable from Off INF to 24 dB with the default value being Off Bias The Bias control simulates the effect of under or over biasing the modeled tape machine Bias is ad justable from 6 dB to 6 dB with a default value of 0
161. at drives the dynamics processing uses a composite of the two channels Because of this when stereo processing occurs there is no im age shift when signal levels differ between the two channels since the composite control signal drives processing for both channels Compressor and Limiter icons Compressor Limiter Chapter 14 Focusrite D3 80 D3 Common Controls Input Level Input Level attenuates signal input level to the compressor or limiter The range of this control is from 30 dB to 0 dB When you use the stereo version of the D3 plug in each channel has its own separate Input Level knob To adjust input levels for both channels si multaneously Shift drag Option Shift clicking Mac or Alt Shift clicking Windows either Input Level knob resets both channels to 0 dB Output Level Output Level adjusts the overall output gain Be cause large amounts of compression can restrict dynamic range the Output Level knob is useful for compensating for heavily compressed signals and making up the resulting difference in level When you use the stereo version of the D3 plug in this single knob controls the master output for both channels The range of this control is from 12 dB of attenuation to 18 dB of gain External Key and Key Listen The side chain is the split off signal used by a plug in s detector to trigger dynamics processing Exter nal Key lets you designate an external source known as the key i
162. ate the Amp Type selec tor and any other controls you cannot automate the selection of Pro Tools plug in Settings files De pending on the amount of overlap or crossfading you want between tones you might be better off us ing the next multi Eleven workflow See the Pro Tools Reference Guide to learn about Snapshot automation Glide and other automation features Chapter 44 Eleven 279 For maximum flexibility control and variety use a dry track bussed to multiple Aux Inputs each with a different Eleven tone see Blending Eleven Amps on page 276 for instructions Configure one for tone A configure the next Eleven on the next Aux Input for tone B which could be a com pletely different amp and sound and so on Then use Pro Tools track Volume fader automation to fade the different Eleven tracks in and out at the right times This gives the greatest amount of con trol over the transition between amps and tones while also letting you stack and layer amps Managing Eleven Plug In Resources If system resources need to be conserved or mini mized you can bus record with effects to commit Eleven tones to disk See Recording Wet Record Eleven Processed Track to Disk on page 271 Or use the AudioSuite version to print Eleven tracks to disk AudioSuite is especially useful when you re processing loops or other shorter form gui tar material Beyond Eleven Some Suggested Effects If you re
163. ates pre clipping levels from 12 dB to 0 dB White Indicates full scale levels from 0 dB to 6 dB Gain Reduction Meters The Input meter can be switched to show Gain Re duction metering for the processed signal from 0 dB to 36 dB The Gain Reduction meters are usually displayed in yellow When the Knee setting for either or both the Expander and the Compressor is greater than 0 dB the Gain Reduction meter displays the amount of the Knee level in amber over the meter s usual yellow display To toggle between the Gain Reduction and Input meters Click the Input Gain Reduction toggle in the top right hand corner of the Input section Input section 5 1 channel format shown Toggling between Input and Gain Reduction meters Chapter 11 Channel Strip 48 Channel Strip Output Section The Output section provides output metering and controls for adjusting the level of the output signal Output Volume Control The Output Volume control sets the output volume after processing letting you make up gain or pre vent clipping on the channel where the Channel Strip plug in is being used The Output Volume control can be set to apply at the end of the FX Chain POST or before the FX Chain PRE see Channel Strip FX Chain on page 49 To adjust the Output Volume do one of the following Click in the Output Volume field and type a value INF dB to 12 dB Click VOL and drag up or down to adjus
164. audio spectrum around 4 kHz where the human ear is most sensitive POW r Dither The POW r Dither plug in is not appropriate for truncation stages that are likely to be fur ther processed It is recommended that POW r Dither be used only as the last insert in the sig nal chain especially when using Type 1 Noise Shaping Chapter 50 POW r Dither 295 The POW r Dither plug in provides three types of noise shaping each with its own characteristics Try each noise shaping type and choose the one that adds the least amount of coloration to the audio be ing processed Type 1 Has the flattest frequency spectrum in the audible range of frequencies modulating and accu mulating the dither noise just below the Nyquist frequency Recommended for less stereophonically complex material such as solo instrument record ings Type 2 Has a psychoacoustically optimized low or der noise shaping curve Recommended for mate rial of greater stereophonic complexity Type 3 Has a psychoacoustically optimized high order noise shaping curve Recommended for full spectrum wide stereo field material For more information on using dither plug ins in Pro Tools see the Pro Tools Reference Guide Part X Sound Field Plug Ins Chapter 51 AutoPan 297 Chapter 51 AutoPan AutoPan is an automatic panning plug in that is available in DSP and Native formats AutoPan pans a mono input to a multichannel stereo LCR quad or 5 0 output base
165. ay The Tempo display shows the tempo in BPM The value in the Tempo display can also be edited di rectly by clicking it and typing a new value AutoPan Envelope Controls When Envelope Env is selected as the Panning source Panning as shown in the Panning display is controlled by the audio signal and the Envelope section controls Side Chain Input When the Side Chain Input selector the key icon is enabled the audio for the Envelope Detector is taken from the side chain input rather than the cur rent track Select the Side Chain Input using the Pro Tools Key Input selector at the top of the plug in window Threshold The Threshold slider sets the amplitude level re quired for the Envelope Detector The LFO Enve lope Detector light blinks brighter when audio is detected above the threshold Attack The Attack slider sets the attack rate of the Enve lope Detector Release The Release slider sets the release rate of the En velope Detector Using AutoPan AutoPan can be used for dynamic panning effects based on a Low Frequency Oscillator LFO an amplitude envelope ENV or manual control AutoPan makes it easy to pan to the beat of a music track as well as panning fly around effects The following section describes two possible scenarios for using AutoPan panning to the beat for rhythmic panning effects and surround panning effects for post production Panning to the Beat AutoPan lets you sync
166. ay and the other has a continuously variable delay The two machines are fed an input signal in paral lel and the output of the machines is then mixed When the variable delay on the second machine is changed at a constant rate using an LFO the re sulting frequency cancellations cause a periodic phasing of the original signal The use of a fixed delay on the first machine makes it possible to adjust the variable delay on the second machine to pass the zero point to a delay value less than the fixed delay resulting in phase cancel lation or the crossover flanging effect Reel Tape Flanger automatically applies tape satu ration effects that correspond to the following con trol settings in Reel Tape Saturation Speed 15 ips Bias 0 0 dB Cal Adjust 9 dB Use the BPM Sync feature to synchronize the Reel Tape Flanger effect to the current tempo of the Pro Tools session Reel Tape Flanger Chapter 39 Reel Tape Flanger 228 Reel Tape Common Controls All Reel Tape plug ins share the following controls Drive Drive controls the amount of saturation effect by increasing the input signal to the modeled tape ma chine while automatically compensating by reduc ing the overall output Drive is adjustable from 12 dB to 12 dB with a default value of 0 dB Output Output controls the output signal level of the plug in after processing Output is adjustable from 12 dB to 12 dB wi
167. ay you can compress the audio of one track using the dynamics of a different audio track The reference track or external audio source used for triggering side chain processing is referred to as the Key Input Using the Impact Side Chain Input Impact provides side chain processing capabili ties Compressors typically use the detected ampli tude of their input signal to cause gain reduction This split off signal is called the side chain How ever an external signal referred to as the Key In put can be used to trigger compression A typical use for side chain processing is to control the dynamics of one audio signal using the dynam ics of another signal referred to as the Key Input For example you could use a lead vocal track to trigger compression of a background vocal track so that their dynamics match Chapter 15 Impact 88 To use a Key Input signal for side chain processing 1 Click the Send button and select a bus path for the audio track or Auxiliary Input you want to use as the side chain signal 2 From Impact s Key Input menu select the input or bus path carrying the audio you want to use as the side chain signal to trigger Impact compres sion The Key Input source must be mono phonic 3 To activate external side chain processing click Ext 4 Begin playback Impact uses the input or bus that you selected as a Key Input to trigger its ef fect 5 If you want to hear the audio source you have se
168. based on the Radius algorithm from iZotope X Form pro vides the high quality time compression and expan sion for music production sound design and audio loop applications Use X Form to manipulate audio loops for tempo matching or to change vocal tracks for formant correct pitch shifting The X Form plug in is useful in audio post production for ad justing audio to specific time or SMPTE durations for synchronization purposes X Form is also ideal for post production pull up and pull down conver sions X Form Displays and Controls Overview The interface for X Form is organized in four sec tions Audio Time Transient and Pitch Audio Use the controls in the Audio section to se lect the most appropriate time compression and ex pansion algorithm for the type of material you want to process and to attenuate the gain of the processed audio to avoid clipping Time Use the controls in the Time section to spec ify the amount of time compression or expansion you want to apply Transient Use the controls in the Transient section to adjust the transient detection for high quality time compression or expansion Pitch Use the controls in the Pitch section to apply pitch shifting Pitch shifting can be formant correct with either the Polyphonic or Monophonic algo rithm X Form Audio Section Controls The Audio section of X Form provides controls for specifying the type of audio you want to process and gain attenuation of the p
169. better results Chapter 24 Time Shift 143 Changing the Time Using Time Shift To change the time of a selected audio clip 1 Select AudioSuite gt Pitch Shift gt Time Shift 2 Select the Audio Mode appropriate to the type of material you are processing Monophonic Poly phonic or Rhythmic 3 In Monophonic or Polyphonic mode select the appropriate Range for the selected material Low Mid High or Wide 4 If compressing the duration of the selection at tenuate the Gain control as necessary 5 If using Monophonic mode adjust the Formant Shift control 6 If using Polyphonic or Rhythmic mode adjust the Transient controls 7 Make sure Pitch Shift is set to 100 unless you also want to change the pitch of the selection 8 Adjust the Time Shift control to set the amount of time change Time change is measured in terms of the target duration using the selected timebase or as a percentage of the original 9 Click Render Changing the Pitch Using Time Shift To change the pitch of a selected audio clip 1 Select AudioSuite gt Pitch Shift gt Time Shift 2 Select the Audio Mode appropriate to the type of material you are processing Monophonic Poly phonic or Rhythmic 3 In Monophonic or Polyphonic mode select the appropriate Range for the selected material Low Mid High or Wide 4 If transposing the pitch of the selection up atten uate the Gain control as necessary 5 If using Monopho
170. c 421 Condenser 67 Condenser 87 Condenser 414 Ribbon 121 Cabinet controls Cabinet Bypass Speaker Breakup Mic Type Off On Axis When enabled Speaker Breakup draws additional CPU resources Mic Type selector in the Cabinet section Eleven is not affiliated with or sponsored or endorsed by the makers of the microphones that are emulated in the product Chapter 44 Eleven 270 Mic Axis When capturing the sound of a speaker cabinet in a studio sound engineers select different micro phones and arrange them in different placements to get a particular sound For example a mic can be pointed straight at a speaker or angled slightly off center in order to emphasize or de emphasize certain frequencies that the mic picks up On axis for most microphones is a line in the same direction as the long dimension of the microphone and will produce a noticeable difference in color ation when compared to the same microphone in the off axis position In Eleven the Axis switch lets you toggle between on and off axis setting of the currently selected microphone model The default position for Mic position is On Axis About Mic Placement All Eleven cabinets and mics were close mic d whether on or off axis This provides the purest tones possible of any room tone or ambience spe cific to the Eleven recording environment Tracks and Signal Routing for Guitar The way you set up Pro Tools track
171. cer Zoomer The Zoomer increases or decreases magnification of the waveform data currently visible in the center of the waveform display so that you can more accu rately set sample trigger thresholds To zoom in on amplitude click the Up Arrow To zoom out on amplitude click the Down Arrow To zoom in on time click the Right Arrow To zoom out on time click the Left Arrow SoundReplacer Crossfade When Crossfade is selected SoundReplacer cross fades between replacement audio files in different amplitude zones This helps smooth the transition between them When Crossfade is deselected SoundReplacer hard switches between replacement audio files in differ ent amplitude zones Crossfading is particularly useful for adding a sense of realism to drum replacement Crossfading between a straight snare hit and a rim shot for ex ample results in a much more live feel than sim ply hard switching between the two samples SoundReplacer Peak Align When Peak Align is on SoundReplacer aligns the peak of the replacement file with the peak of the source file in a way that best maintains phase co herency When Peak Align is off SoundReplacer aligns the beginning of the replace ment file with the trigger threshold point Depending on the characteristics of your source and replacement audio files using Peak Align can significantly affect the timing of audio events in the replacement file It is essenti
172. character making it sound more natural or con versely more unnatural depending on the desired effect Typically dynamics are used to give a reverb a shorter decay time when the input signal is above the threshold and a longer decay time when the in put level drops below the threshold This produces a longer more lush reverb tail and greater ambience between pauses in the source au dio and a shorter clearer reverb tail in sections without pauses For example on a vocal track use Dynamics to make the reverb effect tight clear and intelligible during busy sections of the vocal where the signal is above the Threshold setting and then bloom or lengthen at the end of a phrase where the signal falls below the threshold Similarly Dynamics can be used on drum tracks to mimic classic gated reverb effects by causing the decay time to cut off quickly when the input level is below the threshold To hear examples of decay dynamics load one of the Dynamics presets using the Plug In Li brarian menu Chapter 28 Reverb One 161 Decay Ratio Controls the ratio by which reverb time is increased when a signal is above or below the Threshold level Dynamics behavior differs when the Decay Ratio is set above or below 1 A ratio setting of greater than 1 increases reverb time when the signal is above the threshold A ratio setting of less than 1 increases a reverb s time when the signal is below the threshold For e
173. ck 3 Set the input of the MIDI track to accept input from your external MIDI device 4 Set the output of the MIDI track to Eleven 5 Right click on any control in Eleven and do one of the following Click Learn then move a control on your MIDI controller Pro Tools maps whichever control you touch to that plug in parameter If you know the MIDI CC value of your foot con troller or other device select it from the Assign menu 6 Right click on any control in Eleven To clear a MIDI assignment Right click the control and choose Forget Eleven Settings Presets You can pick a preset from the plug in Librarian menu To load a preset Click the Librarian menu and select an available Settings file You can save import copy paste and manage set tings using the Settings menu To save your settings as an Eleven preset Click the Settings menu and choose Save Set tings Name the preset choose a location and click Save You can scroll through and select preconfigured Eleven Settings files presets using the plug in Li brarian menu and the buttons Right clicking for MIDI Learn On Mac you can Control click an Eleven pa rameter to show the MIDI Learn menu Note that you won t be able to use the Control key modifier to clutch a Grouped control Plug In controls for Eleven Settings files For more information on Settings files and folders see the Pro Tools Refere
174. ckwise is slowest Variation Variation adjusts how much variation oc curs in the delay The more variation you use the more warbled and wobbly the sound becomes Sustain Sustain determines how long the delay takes to die out It is actually a feedback control similar to the one found on the Moogerfooger An alog Delay Echo Doubler Echo Doubler determines whether or not a second record head is engaged resulting in a double echo Tel Ray Tips and Tricks Each and every Tel Ray varies drastically in motor and flywheel stability resulting in different pitch and variation effects Even the same unit may sound different day to day depending on tempera ture warm up time and other factors Since the original units are basically thirty year old tuna cans bolted to plywood with springs and motors flop ping around inside the virtual Tel Ray Delay pro vides a Variation knob so that you can dial in a Tel Ray in whatever state of disrepair you desire Chapter 35 TimeAdjuster 215 Chapter 35 TimeAdjuster TimeAdjuster is a time processing plug in that is available in DSP and Native formats TimeAdjuster supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates TimeAdjuster operates as a mono multi mono or stereo plug in Use the TimeAdjuster plug in for any of the fol lowing Delay compensation Gain compensation 24 dB Phase inversion for correcting out of phase signals
175. click a control in the list 3 Click OK to close the plug in automation win dow Chapter 8 BF 2A 39 4 In the Edit window do one of the following Click the Track View selector and select Side Chain Filter from the BF 2A sub menu Reveal an Automation lane for the track click the Automation Type selector and select Side Chain Filter from the BF 2A sub menu 5 Edit the breakpoint automation for the BF 2A side chain filter Control range is from 0 the de fault setting where no filtering is applied to the side chain to 100 maximum side chain fil tering To access the side chain filter from a control surface 1 Focus the BF 2A plug in on your control sur face 2 Adjust the encoder or fader current targeting the Side Chain Filter control BF 2A Tips and Tricks AudioSuite Processing When using the AudioSuite version of the BF 2A be sure to select an auxillary side chain input nor mally the track you re processing The default is None and if you leave it set like this there is nothing feeding the detector and you will not hear any compression action Line Amp Turn the Peak Reduction knob full counterclock wise off and use the Gain control to increase the signal level Although the BF 2A does not com press the sound with these settings it still adds its unique character to the tone Feed the BF 2A into the BF76 Or vice versa Glynn Johns who has worked with the Stones the W
176. control 343 destination fields 343 Minimum Pitch control 344 Ratio control 343 Source and Destination controls 343 Time Shift plug in 137 Audio Gain control 138 Audio Mode pop up menu 138 Audio Range pop up menu 138 Audio section 138 changing pitch 143 changing time 143 changing time and pitch 143 Clip indicator 138 Decay Rate 141 displays and controls 137 Follow button 140 Formant section 140 Input modes 142 Level indicator 139 Monophonic mode 138 Original time 139 Pitch section 141 Pitch Shift 141 Polyphonic mode 138 post production workflow 144 Processed time 139 pull up pull down TCE percentages 144 Rhythmic mode 138 Speed 139 Tempo displays 139 Threshold control 141 Time section 139 timebase 139 Transient section 140 Transpose 141 Unit pop up menu 139 Varispeed mode 138 Window 141 TimeAdjuster plug in 215 DSP delay compensation 216 217 phase cancellation 216 Phase Invert button 215 Trim plug in 345 V variable mu 75 Vari Fi plug in 145 Fades controls 146 Selection controls 145 Slow Down control 145 Speed Up control 145 vinyl mastering 77 Voce plug ins Voce Chorus Vibrato 235 236 Voce Spin 236 Rotor Balance 237 X X Form plug in 147 2x 4x and 8x Range buttons 149 Audio section 147 Audio Type pop up menu 148 AudioSuite Input 151 AudioSuite Preview 151 AudioSuite processing preferences 151 Clip indicator 148 Formant Shift control 150 Gain control 148 Level indicator 148 Monophonic mode 148 Original
177. control determines the amount of Aural Exciter enhancement mixed into the original sig nal The control ranges from Min no enhance ment up to Max representing approximately a 6 dB boost when the Drive switch is set to Normal and approximately an 18 dB boost when it is set to High The amount of enhancement mixed into the origi nal signal is displayed as a percentage Switches Drive Switch The Drive switch offers two settings Normal 6 dB and High 18 dB This sets the input sen sitivity to the harmonics generator In general this switch will be left in the Normal position How ever weak signals may require more gain in which case you should place this switch in the High posi tion Use the Drive meter to determine if the signal gain needs to be increased When the meter level stays in the green area never rising into the yellow area then the input signal is too low Raise the input sen sitivity by toggling the Drive switch to High The switch illuminates when Drive is set to High Density Switch The Density switch determines the amount of har monics generated by choosing one of two different harmonics generator algorithms When set on High the output from the harmonic generator ex pands low level signals and compresses the highest peaks This setting provides a higher density of har monics with better control of peak levels Since the amount of harmonics is dependent on the input level start with th
178. covers smoothly If the release time is too short the gain can rapidly fluctuate as the compressor repeatedly tries to recover from the gain reduction If the release time is too long a loud section of the audio material could cause gain re duction that continues through soft sections of pro gram material without recovering This control ranges from 5 ms fastest release time to 4 seconds slowest release time Compressor Limiter III Knee Control The Knee control sets the rate at which the com pressor reaches full compression once the threshold has been exceeded As you increase this control it goes from applying hard knee compression to soft knee compres sion With hard knee compression compression be gins when the input signal exceeds the threshold This can sound abrupt and is ideal for limiting With soft knee compression gentle compression begins and increases gradually as the input signal approaches the threshold and reaches full com pression after exceeding the threshold This cre ates smoother compression For example a Knee setting of 10 dB would be the gain range over which the ratio gradually increased to the set ratio amount The Gain Reduction meter displays light orange while gain reduction has not exceeded the knee set ting and switches to dark orange when gain reduc tion reaches the full ratio This control ranges from 0 db hardest response to 30 db softest response
179. ction Band Enable Button 7 Band EQ Only The Band Enable button on each EQ band toggles the corresponding band in and out of circuit When a Band Enable button is highlighted the band is in circuit When a Band Enable button is dark gray the band is bypassed and available for activation Band Gain Control Each Peak and Shelf EQ band has a Gain control for boosting or cutting the corresponding frequen cies Gain controls are not used on High Pass Low Pass or Notch filters EQ Type Selectors 7 Band EQ Band Enable button Band Gain control Chapter 4 EQ III 17 Frequency Control Each EQ band has a Frequency control that sets the center frequency Peak Shelf and Notch EQs or the cutoff frequency High Pass and Low Pass fil ters for that band Q Control Peak and Notch On Peak and Notch bands the Q control changes the width of the EQ band Higher Q values represent narrower bandwidths Lower Q values represent wider bandwidths Shelf On Shelf bands the Q control changes the Q of the shelving filter Higher Q values represent steeper shelving curves Lower Q values represent broader shelving curves Band Pass On High Pass and Low Pass bands the Q control lets you select from any of the following Slope values 6 dB 12 dB 18 dB or 24 dB per octave 1 Band EQ III The Frequency Graph display in the 1 Band EQ shows a control dot that indicates the center fre quency Peak Shelf and Notch Filters or
180. ctors determine whether the audio signal pans left to right right to left or in a circular motion clockwise or counterclockwise The circu lar path selectors clockwise and counterclock wise are only available with mono to quad and mono to 5 0 formats Panner section mono to 5 0 clockwise path selected Panning Source buttons Panning display mono to 5 0 left to right path selected Path selectors left to right path selected Chapter 51 AutoPan 300 AutoPan LFO Controls The LFO section provides controls for the Low Fre quency Oscillator that can be used to modulate pan ning The controls in the LFO section only affect the panning if LFO is selected as the panning source in the panning section see Panning Source on page 299 Rate The Rate slider adjusts the rate of the LFO in beats per minute When Link to Tempo is activated the slider is ignored and the Tempo display always shows the current session tempo see Tempo Dis play on page 302 Waveform The Waveform selector determines the wave shape used by the LFO The waveform shape in use is graphically depicted by the movement of the Sound Location indicator in the Panning display LFO Triggers By default the LFO cycles continuously through the selected waveform The LFO can be set to cycle through the selected waveform just once or it can be triggered by MIDI Beat Clock the Envelope or manually Single When the Single trigger is selec
181. d Attenuation Listen and Side Chain Listen can be enabled simultaneously in which case At tenuation Listen is audible but Side Chain Listen is not Chapter 19 Pro Expander 116 Side Chain Filter On Off You can use the Side Chain input with or without filtering by enabling or disabling the Side Chain Filter On Off button To enable or disable filtering on the side chain Click the Side Chain Filter On Off button on the right side of the Side Chain section so that it is highlighted To disable it click the button again so that it is not highlighted Side Chain Filter Filter Frequency The Freq control lets you set the center frequency for the selected Filter Type from 20 Hz to 21 0 kHz Filter Q When the Filter Type is set to Band Pass or Notch the Q control is available The Q control changes the width of the filter around the center frequency band Higher Q values represent narrower band widths Lower Q values represent wider band widths Filter Type Four Filter Type options are available for side chain processing Low Pass Select the Low Pass option to apply a low pass filter to the side chain processing at the selected frequency High Pass Select the High Pass option to apply a high pass filter to the side chain processing at the selected frequency Notch Select the Notch option to apply a notch fil ter to the side chain processing at the selected fre quency Band Pass Select the Band
182. d clips increasing or decreasing their length without changing pitch It is especially useful in audio post production for adjusting audio to specific time or SMPTE dura tions for synchronization purposes Time Com pression Expansion is nondestructive Time Compression Expansion Controls The Time Compression Expansion plug in provides the following controls Source and Destination The Source fields display the length of the current selection before process ing in each of the listed formats All fields are al ways active a change made to one value is imme diately reflected in the others The Destination fields both display and control the final length of the selection after processing Enter the length of the Destination file by double click ing the appropriate field in the Destination column Use the Ratio Crossfade Min Pitch and Accuracy controls to fine tune the Time Compression Ex pansion process Ratio Sets the destination length in relation to the source length Moving the slider to the right in creases the length of the destination file while moving the slider to the left decreases its length Crossfade Adjusts the crossfade length in milli seconds optimizing performance of the Time Compression Expansion according to the type of audio material being processed This plug in achieves length modification by replicating or sub tracting very small portions of audio material and very quickly crossfading betwee
183. d enable the audio track 5 Make sure you are not overloading your input signal by checking levels in all tracks and Eleven s Input LED 6 When you re ready arm Pro Tools and begin re cording The dry guitar is recorded to the first audio track processed through Eleven then bussed to the sec ond audio track and recorded to disk Recording Eleven printing its output Eleven Aux Input Audio Track Guitar input Bus input Bus output Chapter 44 Eleven 273 Processing Pre Recorded Tracks Through Eleven You can process pre recorded guitar tracks or al most any material through Eleven To listen to pre recorded tracks through Eleven without re recording 1 Import and place your audio in a Pro Tools audio track 2 Assign the audio track Output to Bus 1 or Bus 1 2 if working with stereo material 3 Create an Aux Input track and configure it by doing the following Click its track Input selector and choose Bus 1 or Bus 1 2 Click the Insert selector and select Eleven 4 Begin playback and watch Eleven s Input LED to check your level Make sure you re not clip ping Eleven s input 5 While listening adjust Eleven s Input knob to increase or decrease input level 6 After setting your gain structure do any of the following Try different Settings files presets to get your basic amp cab mic combination Adjust amp controls Try different cabinets and varyin
184. d level If the release time is too short dis tortion can occur on low frequency signals Set this control to 0 for the fastest release time or to 10 for the slowest release time Depending on the program material and the parameters used this rep resents an approximate range of 15 ms to 1 second for Norm mode or the primary release of Warm mode Smack Output Control In all Smack compression modes Output adjusts the overall output gain which lets you compensate for heavily compressed signals by making up the resulting difference in gain When you apply Smack to stereo or multichannel tracks the Output control determines master output levels for all channels Set this control to 0 for no output gain silence or to 10 for the loudest output gain This represents an approximate range of 40 dB This control is not available in Opto mode This control is not available in Opto mode Smack has no control to directly adjust the threshold level the level that an input signal must exceed to trigger compression The amount of compression will vary with the in put signal which is adjustable by the Input control This control is not available in Opto mode Setting the Input and Output controls to 5 is equal to unity gain at a compression ratio of 1 1 Chapter 22 Smack 129 Smack Side Chain EQ Filter The side chain is the signal path that a compressor uses to determine the amount of gain reduction it applie
185. d of the note display Selecting Meter display InTune Strobe display InTune Reference Frequency Octave buttons Down Octave button Up Octave button Chapter 55 InTune 322 InTune Tone Volume The Tone Volume slider controls the volume of the test tone audio signal InTune Information Display The LCD style information display in InTune dis plays the following The reference frequency The current note to which InTune is tuning The number of cents sharp or flat from the cur rent note The status of any test tones playing InTune Presets InTune provides a selection of factory presets for stringed instruments These presets can be selected from the Plug In Librarian menu To make any preset the default when InTune is instantiated 1 From the Plug In Librarian menu select a preset 2 From the Plug In Settings menu select Set As User Default 3 From the Plug In Settings menu select Settings Preferences gt Set Plug In Default To gt User Set ting Creating InTune Tuning Presets InTune lets you create customized tuning presets that display note selections for specific instruments and tunings Once created these tuning presets can be saved as part of a standard Pro Tools plug in preset From the main InTune screen click the Edit button to display the Tuner Programming screen Chromatic Mode When selected Chromatic Mode overrides any cus tom note selections and displays
186. d on a LFO envelope follower MIDI Beat Clock or manual automation AutoPan is ideal for rhythmic panning effects based on your Pro Tools session tempo It also provides an easy and elegant way to automate panning to multichan nel surround formats for post production AutoPan Controls AutoPan provides output meters panner controls LFO controls tempo controls and envelope con trols AutoPan Output Meters The Output meters display the amplitude of the out going audio In mono to stereo mode a two meter bar is shown In mono to LCR quad or 5 0 mode three four or five channels are shown respectively The Clip indicator lights red when the channel has clipped The clip indicator for each channel can be cleared by clicking it AutoPan Output meters L C R Ls Rs Chapter 51 AutoPan 298 AutoPan Panner Controls The Panner section provides different controls for different output channel configurations AutoPan in mono to stereo and mono to LCR formats pro vide controls common to all output configurations Output Width and Manual AutoPan mono to quad and mono to 5 0 formats provide additional con trols depending on the Path selection Angle and Place or Spread Additionally the Panning Source selector Panning display and Path selectors are common to all output channel configurations Output The Output slider lets you cut or boost the output signal level from 24 dB to 12 dB Width The Width sl
187. de For more complex material that covers a broad frequency spectrum select Wide In Poly phonic mode Wide is the default setting and is usu ally best for all material when using the Polyphonic audio type Gain The Audio Gain control attenuates the input level to avoid clipping Adjust the Gain control from 0 0 dB to 6 0 dB to avoid clipping in the pro cessed signal Clip Indicator The Clip indicator indicates clipping in the pro cessed signal When using time compression or pitch shifting above the original pitch it is possible for clipping to occur The Clip indicator lights when the processed signal is clipping If the pro cessed signal clips undo the AudioSuite process and attenuate the input gain using the Gain control Then re process the selection Time Shift Audio section The range pop up menu is unavailable in Rhythmic mode and Varispeed mode Chapter 24 Time Shift 139 Level Indicator The Level indicator displays the level of the output signal using a plasma LED which uses the full range of plasma level metering colors Time Shift Time Controls The Time section of Time Shift provides controls for specifying the amount of time compression or expansion as well as the timebase used for calculat ing TCE Adjust the Time control to change the tar get duration for the processed audio Original Displays the Start and End times and Length of the edit selection Times are displayed in units of the t
188. de zone The replace ment sample will be triggered at these points The color of the Trigger markers correspond to the matching Threshold slider This lets you see at a glance which replacement samples will be trig gered and where they will be triggered SoundReplacer Load Unload Sound Buttons Clicking the Load Unload Sound icons loads or un loads replacement samples for each of the three trigger threshold amplitude zones Clicking the Floppy Disk icon loads a new sample or replaces the current sample Clicking the Trash Can icon unloads the current sample To audition a replacement sample before loading it into SoundReplacer use the Import Audio com mand in Pro Tools Once you have located and pre viewed an audio file you can then load it into SoundReplacer using the Load Unload Sound icons Threshold controls If you zoom the waveform display below a specific Trigger Threshold slider s amplitude zone the slider will be temporarily unavail able To access the slider again zoom back out to an appropriate magnification level Load Unload Sound SoundReplacer lets you choose whether or not to use sample rate conversion before loading replacement samples if they are at a different sample rate from the session SoundReplacer does not load clips that are part of larger audio files To use a clip as a replacement sample you must first save it as an individual audio file Chapter 58 SoundReplacer 338 SoundRepla
189. decimal places in the following table in the X Form Time Shift field for the corresponding post production task and the process the selected audio Use the X Form Plug In Settings for the corresponding post production task Pull up or Pull Down TCE to 10 Decimal Places Frames Pal to Film 4 tfx 96 0 25 to 24 30 PAL to NTSC 4 1 tfx 95 9040959041 25 to 23 976 29 97 Film to PAL 4 1667 tfx 104 1666666667 24 30 to 25 Film to NTSC 0 1 tfx 99 9000999001 24 30 to 23 976 29 97 NTSC to Pal 4 2667 tfx 104 2708333333 23 976 29 97 to 25 NTSC to Film 0 1 tfx 100 10 23 976 29 97 to 24 30 Part V Reverb Plug Ins Chapter 27 D Verb 155 Chapter 27 D Verb D Verb is a studio quality reverb plug in that is available in DSP Native and AudioSuite formats D Verb supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates D Verb operates as a mono multi mono mono to stereo or stereo plug in D Verb Controls D Verb provides a variety of controls for adjusting plug in parameters Input Level Meters Input meters indicate the input levels of the dry au dio source signal An internal clipping LED will light if the reverb is overloaded This can occur even when the input levels are relatively low if there is excessive feed back in the delay portion of the reverb To clear the Clip LED click it Output Level Meter Output meters indicate the output levels of th
190. ded from the IR browser If no such IR is loaded for exam ple the IR in use has been loaded from a preset or session but does not exist in the IR browser the Quick browser controls are inoperative Space Waveform Mode Waveform mode is selected using the Waveform icon at the top of the Space window Waveform mode displays the IR waveform along a horizontal axis marked in seconds and the vertical axis marked in amplitude The early section of the waveform is highlighted in a lighter color In addi tion the channel selector highlights the current channel in the waveform IR information such as sample rate and number of input and output channels is displayed at the bot tom right of the waveform The controls in Waveform mode function as fol lows Original Bypasses all waveform processing allow ing the original IR to be auditioned This control ef fectively bypasses the processing in the IR com puter as shown in the system diagram Display Mode selectors Info bar Display area Waveform mode Chapter 30 Space 193 Channel Selectors Displays from one to five chan nels in the order Left Center Right Left Sur round Right Surround Click the desired channel to display the IR waveform for that channel In Mono mode no channel selector is displayed Zoom Zooms in and out on the time axis for the waveform display Space Picture Preview Mode Picture Preview mode is selected using the Picture Preview ico
191. dio loops listen carefully and adjust the Accuracy slider until you find a setting that keeps timing solid within the clip If you don t start and end times may be pre cise but the beats in rhythmic material may appear to be shuffled if too little priority is given to Rhythm Chapter 60 Trim 345 Chapter 60 Trim Trim is a mono or multi mono plug in that is available in DSP and Native formats Trim supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Use Trim to attenuate an audio signal from Infinity dB to 6 dB or Infinity dB to 12 dB For example using a multi mono Trim plug in on a multi channel track provides simple DSP efficient muting control over the individual channels of the track This capability is useful since Track Mute buttons mute all channels of a multi channel track and do not allow muting of individual channels within the track Trim Controls The Trim plug in provides the following controls Phase Invert Inverts the phase polarity of the in put signal to change the frequency response charac teristics between multi miked sources or to correct for miswired microphone cables Gain Provides dB to 6 dB or 12 dB of gain adjustment depending whether the Gain toggle is set to 6 or 12 6 12 Gain Toggle Switches the maximum level of attenuation between dB to 6 dB and dB to 12 dB Output Meter Indicates the output lev
192. dio signal The Effect Fre quency slider then becomes the primary control for modifying the sound LFO Produces a low frequency triangle wave as a modulation source The rate and amplitude of the triangle wave are determined by the Mod Rate and Mod Amount controls respectively Envelope Follower Causes the selected effect to dynamically track the input signal by varying with the amplitude envelope of the audio signal As the signal gets louder more modulation occurs This can be used to produce a very good automatic wah wah type effect When you select the Envelope Follower the Mod Amount slider changes to a Mod Slewing control Slewing provides you with the ability to smooth out extreme dynamic changes in your modulation source This provides a smoother more continuous modulation effect The more slewing you add the more gradual the changes in modulation will be Sample Hold Periodically samples a random pseudo noise signal and applies it to the effect fre quency Sample and hold modulation produces a characteristic random stair step modulation The sampling rate and the amplitude are determined by the Mod Rate and Mod Amount controls respec tively Chapter 40 Sci Fi 234 Trigger Hold Trigger and hold modulation is simi lar to sample and hold modulation with one signif icant difference If the input signal falls below the threshold set with the Mod Threshold control mod ulation will not occur This provides interestin
193. e amount of resources as the audio track on which it is inserted Consequently you can only use a total combination of audio track channels and ReWire audio streams that does not exceed the maximum number of pos sible voices for your system For example if you are playing 96 stereo audio tracks in a 48 kHz 24 bit session on a system that supports 256 voices at that sample rate another 64 channels of audio will be available for use with ReWire However note that ReWire only supports a maximum of 64 audio streams per host application Using ReWire at higher sample rates will increase the load on the CPU For example CPU load at 96 kHz is double the load at 48 kHz You can mon itor Pro Tools CPU usage in the System Usage window making sure to not overtax your system Track Count with Pro Tools Host based Systems With Pro Tools host based systems performance is determined by several factors including host CPU speed available memory and buffer settings Avid cannot guarantee 64 simultaneous audio channel outputs with ReWire on all computer configura tions For the latest information on recommended CPUs and system configurations visit the Avid website www avid com Client software must support the same sample rate as the session using ReWire For example third party client software that does not sup port sample rates above 48 kHz cannot be used in a 96 kHz Pro Tools session With Pro Tools HD the standard Hardware
194. e Gain Structure If the input material has a very high peak to peak level and no additional headroom for Exciter ef fects use the Level fader to adjust the signal level to avoid over coloring the signal When using digital audio as a sound source such as a CD Player with S PDIF outputs there is a very high peak to peak level because the material on the CD is optimized for the best signal to noise perfor mance In this situation the Level fader can be used to adjust the signal level to gain additional head room In an analog based system you will have the same headroom problem when using a very high peak to peak level signal Using the Level fader to adjust for more headroom is also useful when restoring older recordings For optimal performance keep the peak hold meter of the Drive meter inside the yellow area The harder you drive the Exciter the more Exciter en hancement you generate If you cannot get the Drive meter to register in the yellow area try set ting the Drive switch to High Optimizing Aural Exciter Effects When using Aural Exciter the output signal level has to be equal to the input signal level plus the en hanced Exciter effect The dynamic characteristics of the harmonic generator used in the Aural Exciter plug in are based on a complex algorithm that in cludes the signal peak level the averaged steady state level and the dynamic characteristics Unlike an EQ which adds a constant boost in the h
195. e audio replacement triggering 10 Adjust the Dynamics slider to fine tune how SoundReplacer tracks and matches changes in the source audio s dynamics 11 Adjust the Mix slider to set the balance between replacement audio and source audio 12 Adjust the AudioSuite File controls These set tings will determine how the file is rendered and what effect the rendering will have on the origi nal clips 13 Render the selected clip by doing one of the fol lowing To render the selected clip only in the track in which it appears choose Playlist from the Selec tion Reference pop up To render the selected clip in the Audio Clip List only choose Clip List from the Selection Refer ence pop up If you use only a single replacement sample you should still set all three amplitude zones for optimum results This will ensure accu rate triggering For details See Mapping The Same Sample Into Multiple Amplitude Zones with SoundReplacer on page 341 Chapter 58 SoundReplacer 340 14 Determine which occurrences of the selected clip you want to render by doing one of the fol lowing To render and update every occurrence of the se lected clip throughout your session enable Use In Playlist and also choose Clip List from the Se lection Reference pop up If you do not want to update every occurrence of the selected clip disable Use In Playlist 15 If you have selected multiple clips for rendering and w
196. e delay If you don t hear any audio when you invert a signal s phase you have precisely adjusted and compen sated for the delay This is because when you mon itor duplicate signals and invert the polarity phase of one of them the signals will be of opposite po larity and cancel each other out This technique is convenient for finding the exact delay setting for any plug in To determine the delay of a plug in by inverting its signal phase 1 Place duplicate audio clips on two different au dio tracks and pan them to the center mono 2 Apply the plug in whose delay you want to cal culate to the first track and a Time Adjuster plug in to the second track 3 With TimeAdjuster invert the phase 4 Control drag Windows or Command drag Mac to fine tune delay in one sample incre ments or use the up down arrow keys to change the delay one sample at a time until the audio signal disappears 5 Change the polarity back to normal 6 Save the TimeAdjuster setting for later use Comb Filter Effect Cancellation Adjust the delay with the signal in phase until any comb filter effects cancel out Some plug ins such as Avid s Maxim have different delays at different sample rates Chapter 35 TimeAdjuster 217 Viewing Channel Delay and TimeAdjuster Because plug ins display their delay values in the channel delay indicators this can be used as an other method for determining delay compensation To view time
197. e effect This is useful for rendering the ef fect in place especially if the selection is con strained by the grid or by adjacent clips When this option is enabled processing is applied to only two thirds of the selection so that the resul tant rendering maintains the original duration of the selection Vari Fi Chapter 25 Vari Fi 146 Extend When the Extend option is selected all audio in the current Edit selection is processed and rendered The resulting rendering is 150 the duration of the Edit selection The selection start point does not change but the rendered clip extends beyond the end of the Edit selection This can be useful if the last third for speeding up or the first third for slowing down of the Edit se lection needs to be heard in the rendered effect Fades Controls On When the On option is selected a fade out is ap plied if the Slow Down option is selected or a fade in is applied if the Speed Up option is selected Off When the Off option is selected no fade in or fade out is applied in the rendered Edit selection This can result in a more pronounced tape stop or tape start effect and can also be useful for pre serving the dynamic level at the end of the Edit se lection when the Slow Down option is selected or the beginning of the selection when the Speed Up option is selected Chapter 26 X Form 147 Chapter 26 X Form X Form is an AudioSuite plug in that is
198. e first figure shows a fast peaking kick drum whose peak level occurs at its initial attack The second figure shows a slower peaking kick drum whose peak level occurs after its initial at tack If you turn on Peak Align and attempt to replace the fast peaking kick with the slow peaking kick or vice versa SoundReplacer will align their peaks which occur at different points in the sound The audible result would be that the replace ment audio file slow peaking kick would trigger too early Mapping The Same Sample Into Multiple Amplitude Zones with SoundReplacer If you are performing drum replacement and intend to use just a single replacement sample mapping it into multiple amplitude zones will ensure more ac curate triggering Here is why Imagine that you are replacing a kick drum part If you look at the waveform of a kick drum you will often see a pre hit portion of the sound that oc curs as soon as the ball of the kick pedal hits the drum This is rapidly followed by the denser attack portion of the sound where most of sound s weight is With a sound like this using a single amplitude threshold presents a problem because typically in pop music kick drum parts consist of loud accent hits and softer off beat hits that are often 6 dB or more lower in level If you use a single amplitude threshold to trigger the replacement sample you have to set the thresh old low enough to trigger at the soft hits Th
199. e left side of the unit All low frequency equalization is a gentle shelving type 6 dB per octave High Frequency Boost Section Boost mid and high frequencies using the Bandwidth and Boost knobs and the High Frequency switch High Frequency Attenuate Section Cut high fre quencies using the Atten knob and the Atten Sel switch located at the right side of the plug in Use caution because the Sharp bandwidth setting results in up to 10 dB higher output than Broad bandwidth at maximum Boost just like on the orig inal But don t feel like you re getting cheated Consider anything that encourages very careful and infrequent use of peaky boosts to be a Very Good Thing Pultec EQP 1A Chapter 7 Pultec Plug Ins 34 Pultec EQH 2 The Pultec EQH 2 is a program equalizer similar to the Pultec EQP 1A It is designed to provide smooth equalization across final mixes or individ ual tracks The Pultec EQH 2 offers three equalization sec tions low frequency boost and attenuation mid range boost only and 10k attenuation Like its EQP 1A sibling it features high quality transform ers and a tube gain stage But unlike the EQP 1A the tube stage in the EQH 2 is a push pull design As a result the EQH 2 offers a beefier tone Pultec EQH 2 Controls Low Frequency Section Adjust low frequencies using the top row of Boost and Atten knobs and the CPS cycles per second switch All low frequency equalization is a gentle shelving type
200. e original source signal unaffected A setting of 100 pro duces a hard panned stereo image Settings above 100 use phase inversion to create an even wider stereo effect The Stereo Width slider displays red above the 100 mark to remind you that a phase effect is being used to widen the stereo field The range of this control is from 0 to 150 The default setting is 100 100 Wet and Dry Mix Buttons These buttons set the Wet control to 100 Wet or 100 Dry and the current setting A 100 wet mix contains only the reverb effect with none of the di rect signal This setting can be useful when using pre fader sends to achieve send return bussing The wet dry balance in the mix can be controlled using the track faders for the dry signal and the Auxiliary Input fader for the effect return You can also achieve a 100 wet mix by clicking the 100 Wet Mix button The Stereo Width control does not affect the reverberation effect coming through the rear channels If you want to produce a strictly mono reverb be sure to set the Rear Reverb setting Levels section to INF dB Chapter 29 ReVibe II 175 ReVibe II Decay EQ Graph The EQ display lets you adjust the Decay EQ set tings for ReVibe II Click the EQ button to toggle the display to show the Decay EQ settings To ad just a setting on the graph drag the corresponding control point Each control point on the graph has corresponding text fields above and below the d
201. e pro cessed signal Gain and Input Level Controls D Verb provides a Gain control above the Input Level meter to let you adjust the input gain Mix Control The Mix slider adjusts the balance between the dry signal and the effected signal giving you control over the depth of the effect This control is adjust able from 100 to 0 D Verb Chapter 27 D Verb 156 Algorithm Control This control selects one of seven reverb algorithms Hall Church Plate Room 1 Room 2 Ambience or Non linear Selecting an algorithm changes the preset provided for it Switching the Size setting changes characteristics of the algorithm that are not altered by adjusting the decay time and other user adjustable controls Each of the seven algorithms has a distinctly different character Hall A good general purpose concert hall with a natural character It is useful over a large range of size and decay times and with a wide range of pro gram material Setting Decay to its maximum value will produce infinite reverberation Church A dense diffuse space simulating a church or cathedral with a long decay time high diffusion and some pre delay Plate Simulates the acoustic character of a metal plate based reverb This type of reverb typically has high initial diffusion and a relatively bright sound making it particularly good for certain per cussive signals and vocal processing Plate reverb has the general effect of thickening the initial
202. e prob lem occurs at the loud hits The threshold is now set so low that the pre hit portion of the loud hits can exceed the threshold triggering the replacement sample too early This results in a replacement track with faulty timing A fast peaking kick drum A slower peaking kick drum A kick drum with a pre hit preceding a denser attack A single low threshold causes the second louder kick to trigger too early as evidenced by the trigger marker at the very start of the waveform Chapter 58 SoundReplacer 342 The best way to avoid this problem is to set multi ple threshold zones for the same sample using a higher threshold for the louder hit Soft hits will trigger threshold 1 and louder hits will trigger threshold 2 To set the precise threshold for louder hits you may need to zoom in carefully to examine the waveform for trigger points indicated by color coded trigger markers and then Command drag the Threshold slider for more precise adjustment If there is a great deal of variation in the dynamics of the source audio you may need to use all three Trigger Thresholds Amplitude Zones for optimum results Using the Audio Files Folder for Frequently Used SoundReplacer Files If you often use the same settings and replacement sounds in different sessions Sound Replacer provides a convenient way to keep the re placement audio files and settings linked together When you choose a preset from the Plug In Librar
203. e switch set to Normal Switch to High if you still want a greater density of harmonics after the input level is set The switch illuminates when Density is set to High Chapter 42 Aphex Aural Exciter Type III 246 Ax Switch The Ax switch gives you the choice of turning the Aural Excitement process On or Off The Ax switch illuminates when Ax is engaged confirming that the effect is On Unlike the Bypass switch the audio signal from the input does travel through the processing algorithm on the way to the output whenever Ax is Off This means that the SPR effect can still be active by switching Ax Off and SPR On Solo Switch When engaged the Solo switch gives you a choice of auditioning the Aural Excitement signal without the main audio The switch illuminates when Solo is active As an application for the Solo switch select Solo to return the pure effect back to the mixing console for precise memory control of the Aural Excitement signal only SPR Switch The SPR switch controls the Spectral Phase Re fractor effect This effect is independent of all other controls or switches except Bypass The switch il luminates when SPR is engaged confirming that SPR is On SPR processes the main audio signal in such a way that bass frequencies up to 150 Hz lead phase in relation to the rest of the spectrum Through the many steps of recording duplicating distributing and reproducing sound the phase of the low frequency
204. e the HF and LF controls to select a frequency range 5 Begin playback The plug in uses the filtered in put signal to trigger dynamics processing 6 To fine tune side chain triggering adjust the plug in controls Side Chain section Chapter 13 Fairchild Plug Ins 75 Chapter 13 Fairchild Plug Ins The Fairchild plug ins are a pair of vintage com pressor plug ins that are available in DSP Native and AudioSuite formats The Fairchild plug ins support 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates The Fairchild plug ins operate as mono multi mono or stereo plug ins Fairchild 660 Re introducing the undisputed champion in price weight and performance the 35 000 one hun dred pound Fairchild 660 Avid s no compromise replica captures every detail of this studio classic Designed in the early 1950s the Fairchild 660 is a variable mu tube limiter Variable mu designs use an unusual form of vacuum tube that is capable of changing its gain dynamically The result In addition to featuring a tube audio stage like the LA 2A the Fairchild actually achieves gain reduction through the use of tubes The heart of the Fairchild limiter a 6386 triode is one such variable mu tube In fact four of these tubes are used in parallel A key part of the Fair child design it ensures that the output doesn t get darker as the unit goes further into gain reduction and also reduces distortion a
205. e the Peak Reduction control on an LA 2A Time Constant Selects the attack and release times One is fastest and six is slowest Seven and eight are custom presets See Fairchild 670 on page 77 for details on these custom settings AGC Lets you select Left Right processing or Lat Vert processing of the two channels Left Right works like a dual mono compressor with separate controls for the left and right channels In Lat Vert mode the top row of controls affects the in phase Lat information and the bottom row of controls affects the out of phase Vert information Al though originally designed for vinyl mastering where excess Vert vertical information could cause the needle to jump out of the groove you can use the Lat Vert mode to achieve some amazing ef fects especially on drums Fairchild 670 Tips and Tricks To exactly match the settings between channels hold down the Shift key while adjusting a control This is useful when trying to preserve the existing Left Right balance on stereo material Fairchild 670 Chapter 14 Focusrite D3 78 Chapter 14 Focusrite D3 Focusrite D3 is a high quality dynamics processor plug in Developed in cooperation with Focusrite the D3 is based on the highly acclaimed Red Range 3 dual mono stereo compressor amp limiter designed by Rupert Neve D3 is available in DSP Native and AudioSuite formats D3 supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kH
206. e them a try you ve already heard them on hit songs on the radio Pump It Up With a carefully adjusted Input Gain and Thresh old you can use Time Constant 1 to achieve a cool pumping effect on drums The sound gets darker and fuller and sits beautifully in a mix Chapter 13 Fairchild Plug Ins 77 Fairchild 670 Avid s no compromise replica captures every de tail of the Fairchild 670 The Fairchild 670 is a dual channel unit and as such is only available on stereo tracks Note that the companion Fairchild 660 also sup ports stereo operation Both a Fairchild 660 and a Fairchild 670 were modeled from scratch using two different hardware units This gives you a choice of two different sounding Fairchild units to try on your stereo tracks The internal design of the Fairchild 670 is similar to the Fairchild 660 However the Fairchild 670 of fers two channels of compression instead of one Combined with the AGC control this gives you even more compression options on stereo tracks Fairchild 670 Controls Adjust the Input Gain and Threshold controls to gether on both channels until you get the sound you want Like many classic compressors after a little bit of tweaking you ll know immediately when you get it right Input Gain Sets the input level to the unit and the compression threshold just like the Input control on an 1176 Full clockwise is loudest Threshold Adjusts the gain to the sidechain just lik
207. e times BF76 Controls BF76 provides the following controls Input The Input control sets the input signal level to the compressor which in the 1176 design deter mines both the threshold and amount of peak re duction Output The Output control sets output level Use it to bring the signal back to unity after applying gain reduction Attack and Release The Attack and Release con trols set the attack and release times of the com pressor Full counterclockwise is slowest and full clockwise is fastest Attack times vary between 0 4 milliseconds to 5 7 milliseconds Release times vary between 60 and 1 100 milliseconds Ratio The Ratio Push switches select the compres sion ratio from 4 1 to 20 1 Meter The Meter Push switches affect the meter ing GR shows the amount of gain reduction 18 and 24 show the output level calibrated so that 0VU indicates 18dB FS and 24dB FS re spectively The Off switch turns off the meter BF76 Chapter 10 BF76 43 BF76 Tips and Tricks AudioSuite Processing When using the AudioSuite version of BF76 be sure to select a side chain input normally the track you are processing The default is None and if you leave it set like this there s nothing feeding the detector and you won t hear any compression ac tion Unexpected Visit from A amp R Weevil Yields Instant Hit Mix A favorite feature on one megabuck mixing con sole is its st
208. e will be the absolute ceiling level for limited peaks Maxim Attenuation Meter This meter displays the amount of gain reduction being applied over the course of playback with the maximum peak displayed in the numeric readout at the bottom of the meter For example if the numer ical display at the bottom of the Attenuation meter displays a value of 4 dB it means that 4 dB of lim iting has occurred Since this is a peak hold read out you can temporarily walk away from a session during playback and still know the maximum gain reduction value when you come back To clear the numeric readout click it with the mouse Maxim Release Knob This knob sets how long it takes for Maxim to ease off of its attenuation after the input signal drops be low the threshold level Because Maxim has an at tack time of zero milliseconds the release control has a very noticeable effect on the character of lim iting In general if you are using heavy limiting you should use proportionally longer release times in order to avoid pumping that may occur when Maxim is forced to jump back and forth between limited and unlimited signal levels Lengthening the release time has the effect of smoothing out these changes in level by introducing a lag in the ramp up or ramp down time of attenuation Use short release times on material with peaks that are relatively few in number and that do not occur in close proximity to each other The Release control has a d
209. ea is in Snapshot mode a snapshot can be selected by clicking the selection area next to the snapshot name Select Lets you select which snapshot is currently loaded Name Displays the name of each snapshot By de fault snapshots are named Snapshot 1 through Snapshot 10 Snapshots can be renamed by click ing on the snapshot name and entering a new name followed by the Enter key Windows or the Return key Macintosh Sample Path Displays the name of the IR selected for each snapshot Copy Copies the currently selected snapshot set tings into a clipboard Display area Picture Preview mode Display area Snapshot mode Chapter 30 Space 194 Paste Pastes the clipboard into the currently se lected snapshot Note that the name of the existing snapshot is not changed by pasting a new snapshot in order to avoid duplicate snapshot names Clear Clears the IR from the currently selected snapshot Space Preferences Mode Preferences mode is selected using the Preferences icon at the top of the Space window This displays a number of preferences settings for Space Embed IRs in Preset amp Session Files Enables or disables the embedding of IR waveforms in presets and session file By default this is enabled Installed IR Packages Displays a list of installed Space IR packages and their versions Space Meters The Meters display the amplitude of the incoming and outgoing audio signals by channel The numb
210. each other Peaking Control The Peaking control provides a damping effect on the leading frequency edge of the high pass filter controlled by Tune As you vary this control from Min to Max the Tune frequency becomes more ac centuated as shown in the following figure How ever at the same time a dip is created just before the accentuated Tune frequency This dip or null becomes larger as Peaking is increased Tune control Peaking control TYPE III TUNING RANGE RESPONSE dB FREQUENCY Hz 7 0 5 0 3 0 1 0 1 0 20 0 100 0 1000 0 10000 0 TUNE at 700 Hz TUNE at 7 kHz NULL FILL at Minimum PEAKING at midpoint for all curves TUNE at 3 kHz midpoint TYPE III PEAKING RESPONSE RESPONSE dB FREQUENCY Hz 10 0 6 0 2 0 2 0 6 0 20 0 100 0 1000 0 10000 0 PEAKING at MAX NULL FILL at MIN TUNE at midpoint for all curves PEAKING at MIN PEAKING at MIDPOINT Chapter 42 Aphex Aural Exciter Type III 244 The amount of Peaking is displayed as a percent age Null Fill Control The Null Fill control adjusts the curve of the high pass filter to fill in the null caused by the sum ming of the side chain return signal and the input signal The amount of Null Fill is displayed as a percent age This control compensates for phase pulling which occurs as a side effect of the time delay pres ent in the side chain signal an important part of the Aural
211. eaks that occur at a specific dB level the longer the X axis line If there appears to be a pronounced spike at a certain dB level 4 dB for example it means that there are a relatively large number of waveform peaks occurring at that level You can then use this information to decide how much limiting to apply to the signal Maxim introduces a delay that is proportional to the session sample rate To preserve phase synchronicity between multiple audio sources when Maxim is only applied to one of these sources use Delay Compensation or the Time Adjuster plug in Chapter 17 Maxim 94 By dragging the Threshold slider downwards you can visually adjust the level at which limiting will occur Maxim displays the affected range in or ange Maxim Threshold Slider This slider sets the threshold level for limiting Sig nals that exceed this level will be limited Signals below it will be unaffected Limited signal peaks are attenuated to match the threshold level so the value that you set here will determine the amount of reduction applied Maxim Output Meter This meter displays the amplitude of the output sig nal The value that appears here represents the pro cessed signal after the threshold ceiling and mix ing settings have been applied Maxim Ceiling Slider This slider determines the maximum output level After limiting is performed you can use this slider to adjust the final output gain The value that you set her
212. early reflections can be viewed at the same time Buttons below the dis play allow you to select the type of data being dis played Early Reflections Button The Early Reflections button toggles display of early reflections on or off within the Contour dis play When the Early Reflections button is illumi nated early reflections data is displayed When the Early Reflections button is not illuminated early re flections data is not displayed Both early reflec tions and reverb contour data can be displayed si multaneously Reverb Contour Button The Reverb Contour button toggles display of the reverb contours for both the front and rear channels on or off within the Contour display When the Re verb Contour button is illuminated the reverbera tion envelopes are displayed When the Reverb Contour button is not illuminated the reverberation envelopes are not displayed Both early reflections and reverb contour data can be displayed simulta neously Contour display Chapter 29 ReVibe II 177 Front Button The Front button toggles display of the front chan nel reverb contour and the front channel early re flections on or off within the Contour display When the Front button is illuminated the initial re verberation envelope and early reflections for the front channels are displayed When the Front but ton is not illuminated they are not displayed Rear Button The Rear button toggles display of the rear channel
213. ection algorithm It you encounter undesirable frequency artifacts in the middle of long held notes try raising the Threshold setting If audio transients are obscured in the transposed signal try lowering the Threshold setting Window Pitch II changes the pitch by splitting the incoming audio up into small grains 6 0 ms to 42 ms re sampling those grains and adding those grains back together The Window control deter mines the size of the grains If you are working with long pad like material such as legato strings then increasing the Window size may improve audio quality having too short of a Window in this situa tion may result in robotic or buzzy sounding audio If you are pitch shifting material with sharp tran sients such as a drum part reducing the Window size improves transient response having too long of a Window in this situation may make the audio sound smeared and choppy Follow Enable Follow to match the overall dynamic envelope of the source audio If pitch shifted audio does not match the same decay sound as the origi nal turning Follow on may improve the sound In most cases however leaving Follow off should sound just fine Pitch Shift Controls Keyboard Relative Pitch Transposition Click any note on the keyboard to set a relative pitch transposition value that will be applied to the audio signal The C key in the middle of the keyboard represents the original pitch of the incoming signal if the
214. ections section has controls for the various early reflection elements including ER set ting level spread and delay Calculating Early Reflections A particular reflection within a reverberant field is usually categorized as an early reflection Early re flections are usually calculated by measuring the reflection paths from source to listener Early re flections typically reach the listener within 80 mil liseconds of the initial audio event depending on the proximity of reflecting surfaces Simulating Early Reflections Different physical environments have different early reflection signatures that our ears and brain use to pinpoint location information These reflec tions influence our perception of the size of a space and where an audio source sits within it Changing early reflection characteristics changes the per ceived location of the reflecting surfaces surround ing the audio source This is commonly accomplished in digital rever beration simulations by using multiple delay taps at different levels that occur in different positions in the stereo spectrum through panning Long rever beration generally occurs after early reflections dis sipate Reverb One provides a variety of early reflections models These let you quickly choose a basic acoustic environment then tailor other reverb char acteristics to meet your precise needs For an interesting musical effect set the Pre Delay time to a beat interval such
215. ed and designed the values and ranges of these timings were chosen by experimentation using wide ranges of program material Because of these intentional effects produced by the compressor the JOEMEEK makes a perfect tool for general enhancement of tracks to brighten tighten clarify and catch the at tention of the listener functions that are difficult or impossible to achieve with conventional compres sor designs Chapter 17 Maxim 91 Chapter 17 Maxim Maxim is a unique and powerful peak limiting and sound maximizing plug in that is available in DSP Native and AudioSuite formats Maxim is ideal for critical mastering applications as well as standard peak limiting tasks Maxim offers several critical advantages over tra ditional hardware based limiters Maxim takes full advantage of the random access nature of disk based recording to anticipate peaks in audio mate rial and preserve their attack transients when per forming reduction This makes Maxim more trans parent than conventional limiters since it preserves the character of the original audio signal without clipping peaks or introducing distortion Maxim supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Maxim operates as a mono multi mono or stereo plug in Maxim features include Perfect attack limiting and look ahead analy sis accurately preserve transient attacks and the character of origi
216. efault value of 1 millisecond Histogram density of waveform peaks at each level dB level of waveform peaks Chapter 17 Maxim 95 Maxim Mix Slider This slider sets the ratio of dry signal to limited sig nal In general if you are applying Maxim to a main output mix you will probably want to set this control to 100 wet If you are applying heavy lim iting to an individual track or element in a mix to modify its character this control is particularly use ful since it lets you control precisely the amount of the processed effect mixed with the original signal Maxim Link Button When depressed this button located between the Threshold and Ceiling numeric readouts links the Threshold and Ceiling controls These two sliders will then move proportionally together As you lower the Threshold control the Ceiling control is lowered as well When these controls are linked you can conveniently compare the effect of limit ing at unity gain by clicking the Bypass button Maxim Dither Button When selected this applies dither Dither is a form of randomized noise used to minimize quantization artifacts in digital audio systems Quantization arti facts are most audible when the audio signal is near the low end of its dynamic range such as during a quiet passage or fade out Applying dither helps reduce quantization noise that can occur when you are mixing from a 24 bit source to a 16 bit destination such as CD R or DA
217. el including any gain compensation added using the Gain con trol Mute Mutes the signal output Trim Alt click Windows or Option click Mac the Trim selector to open a plug in window for each channel of a multi channel track Automation data adjusts to reflect the cur rent Gain setting When working with au tomation data from an older version of the Trim plug in ensure the Gain setting is set at 6 dB Chapter 61 Other AudioSuite Plug In Utilities 346 Chapter 61 Other AudioSuite Plug In Utilities The following AudioSuite only utility plug ins are installed when you install Pro Tools DC Offset Removal Duplicate Gain Invert Normalize Reverse DC Offset Removal The DC Offset Removal plug in removes DC off set from audio files DC offset is a type of audio ar tifact typically caused by miscalibrated analog to digital converters that can cause pops and clicks in edited material To check for DC offset find a silent section in the audio material If DC offset is present a near verti cal fade in with a constant or steady state offset from zero will appear in the waveform Use the DC Offset Removal plug in to remove it To remove DC offset from an audio clip 1 Select the clip with DC offset 2 Choose AudioSuite gt Other gt DC Offset Removal 3 Ensure that Use In Playlist is enabled 4 Click Render DC Offset Removal Chapter 61 Other AudioSuite Plug I
218. el that an input signal must exceed to trigger compression The amount of compression will vary with the in put signal which is adjustable by the Input control Chapter 22 Smack 127 Smack Controls and Meters Smack includes controls for multiple compression modes and a VU meter Smack Compression Mode Buttons Smack has three modes of compression Norm Normal Opto and Warm Use the corresponding button to select a mode Norm Mode Button Enable the Norm button to emulate FET compres sors which can have significantly faster attack and release times than opto electrical based compres sors It can be used for a wide range of program ma terial and with extreme settings can be used for sound effects such as pumping In Norm mode you can precisely adjust the Ratio Attack and Release controls to fine tune the com pression characteristics Warm Mode Button Enable the Warm button for compression that is based on Norm mode but which has program de pendent release characteristics These characteris tics often described as transparent or smooth can be less noticeable to the listener and can reduce waveform distortion caused by some sustained low frequency tones As with Norm mode Warm mode can be used for a wide range of program material including vocals or low frequency instruments such as tom toms or bass guitar Extreme settings can be used to pro duce pumping effects Li
219. en Controls This section tells you how to adjust controls using your mouse a Pro Tools controller or with a MIDI device Navigating the Amp Cab and Mic Type Selectors You can click on the name of the current Amp Type Cab Type or Mic Type to display their pop up menus and select an item You can also click the Previous Next arrows to step through Amp Cabinet and Mic choices one at a time Groups and Linked Plug In Controls Eleven s parameters can follow Pro Tools Groups Mix Edit or Mix Edit for linked control of mul tiple inserts For more information see the Pro Tools Reference Guide Using Automation All of Eleven s parameters can be automated When a parameter has been enabled for automa tion an LED appears lit near that control Using a Controller with Eleven Eleven can be controlled directly from any compat ible Pro Tools controller Eleven appears along with other plug ins and can be assigned edited by passed and automated using the Insert section as available on the particular controller being used About Unused Controls and Controllers Some amps that have relatively few controls such as the Tweed Lux will display controls on a con troller that are not actually available with that par ticular amp model Even though you can adjust these unused encoders or switches only those con trols seen on screen for any amp can be adjusted from a controller Changing an unused control does nothi
220. en simultaneously 272 processing pre recorded tracks 273 Release 264 resonance 275 resources 279 ribbon 269 sample rates 261 Settings Files 263 signal flow 280 signal routing and track setups 270 single coil 260 Speaker Breakup 269 Speed 267 stacking 274 standby 266 stereo 261 tap 279 Tempo Sync 267 Audio Plug Ins Guide 353 Threshold 264 tips 278 tracks processed through Eleven 271 Tremolo 267 EQ Channel Strip 44 EQ III 12 Focusrite D2 26 JOEMEEK VC5 Meequalizer 32 Pultec EQH 2 34 Pultec EQP 1A 33 Pultec MEQ 5 34 EQ III plug in 12 1 Band EQ 12 7 Band EQ 13 Frequency Graph display 13 56 gain inverting 13 Expander Gate Dynamics III 65 external side chain processing 84 F Fairchild plug ins 660 75 670 77 Focusrite D2 plug in 26 bypassing 28 configurations 26 curves 28 disabling 28 enabling 28 filter controls 28 High Mid Peak Filter 29 High Pass Filter 29 High Shelf Filter 30 Input Level parameter 27 Left Channel button 31 Link button 31 Low Mid Peak Filter 29 Low Pass Filter 30 Low Shelf Filter 29 meters 27 Output Level parameter 27 Right Channel button 31 Focusrite D3 plug in 78 Attack 82 Auto Release button 82 brick wall limiter 79 Compressor 78 Compressor Limiter 78 Compressor Limiter 78 controls 80 83 External Key control 80 84 Gain Reduction meter 81 in out icon 81 83 Limit LED 83 Limiter 79 meters 81 Ratio control 82 RMS detector 79 side chain processing 84 Threshold parameter 82
221. ency variable from 20 Hz to 6 4 kHz Low Shelf Filter The Low Shelf Filter provides two rotary controls The upper rotary control adjusts the corner fre quency variable from 33 Hz to 460 Hz The lower rotary control adjusts the filter s amplitude gain or attenuation Amplitude range is 15 dB from unity Low Mid Peak Filter The Low Mid Peak Filter provides three rotary controls The upper rotary control adjusts the center frequency variable from 33 Hz to 6 4 kHz The lower left rotary control adjusts the filter s ampli tude gain or attenuation Amplitude range is 15 dB from unity utilizing a reciprocal curve for both gain and attenuation The lower right rotary control adjusts filter Q which is variable from 0 7 to 4 0 High Mid Peak Filter The High Mid Peak Filter provides three rotary controls The upper rotary control adjusts the center frequency variable from 120 Hz to 18 kHz The lower left rotary control adjusts the filter s ampli tude gain or attenuation Amplitude range is 15 dB from unity utilizing a reciprocal curve for both gain and attenuation The lower right rotary control adjusts filter Q which is variable from 0 7 to 4 0 High Pass Filter Low Shelf Filter Low Mid Peak Filter High Mid Peak Filter Chapter 5 Focusrite D2 30 High Shelf Filter The High Shelf Filter provides two rotary controls The upper rotary control adjusts the corner fre quency variable from 3 3 k
222. ential aspects of each amplifier including characteristics of the input stage Providing an appropriate level of signal delivers the most accurate response from the plug in If you are working with pre recorded guitar tracks see Using Eleven with Pre Recorded Tracks on page 260 If you are working with a live guitar signal follow the steps on the next few pages for optimal input level calibration Input calibration takes only a couple of minutes and helps ensure the best results with Eleven its amps and its factory presets Basic gain stages to calibrate live guitar input for Eleven Pro Tools Pro Tools level Source Hardware Eleven Vol at max Hardware input gain Should be yellow or orange Input LED Chapter 44 Eleven 258 Connect your Guitar and Configure Source Input If your setup includes pedals or other gear it helps to know whether the final output device is provid ing an instrument or line level signal Choose and configure your input and source settings accord ingly Check the Setup Guide that came with your system for more information To connect your guitar to a Pro Tools host based system 1 Do one of the following depending on your hardware configuration If you are using an interface that has a DI input such as a Pro Tools Mbox Pro plug your gui tar into an available DI input If you are using your computer s built in inputs plug your guitar into an ava
223. equency Graph display or its Gain knob in the plug in win dow This changes a gain boost to a cut 9 to 9 or a gain cut to a boost 9 to 9 Gain values can not be inverted on Notch High Pass Low Pass or shelving bands Dragging in the Frequency Graph Display You can adjust the following by dragging the con trol points directly in the Frequency Graph display Frequency Dragging a control point to the right in creases the Frequency setting Dragging a control point to the left decreases the Frequency setting Gain Dragging a control point up increases the Gain setting Dragging a control point down de creases the Gain setting Q Control dragging Mac or Start dragging Win dows a control point up decreases the Q setting Control dragging Mac or Start dragging Win dows a control point down increases the Q setting 7 Band EQ Dragging a control point in the Frequency Graph display Chapter 4 EQ III 14 Using EQ III in Band Pass Mode You can temporarily set any EQ III control to Band Pass monitoring mode Band Pass mode cuts monitoring frequencies above and below the Fre quency setting leaving a narrow band of mid range frequencies It is especially useful for adjusting limited bandwidth in order to solo and fine tune each individual filter before reverting the control to notch filter or peaking filter type operations To switch an EQ III control to Band Pass mode Hold Control Shift Mac or
224. er of meters shown will depend on the number of in put and output channels Input meters may be mono or stereo and output meters may be mono stereo quad or 5 0 channels Each meter is marked as ei ther mono left right center left surround or right surround A logarithmic scale marked in decibels and momentary peaks are also displayed on the me ter The red Clip indicators at the top of the meters in dicate clipping on the corresponding channel When a channel has clipped once the clip indicator remains lit and additional clips will be shown by a variation in the color of the indicator The clip indi cator for all channels can be cleared by clicking on any clip indicator or selecting Track gt Clear All Clip Indicators in Pro Tools or pressing Option C Mac or Alt C Windows Display area Preferences mode Meters stereo input to 5 0 output shown Chapter 30 Space 195 Space IR Browser The IR browser lets you quickly and easily install locate and organize IRs on local hard drives The Load and Edit buttons in the IR browser let you in stall and import IRs create Favorites and change the IR groups displayed Space automatically highlights each IR that matches the current channel configuration For ex ample when using a Space Stereo to Quad inset each IR with that configuration is highlighted Im pulses that are not highlighted can still be loaded and Space tries to adapt the IR to the current chan nel
225. erator The Noise slider mixes a percentage of pseudo white noise into the audio signal Noise is useful for adding grit into a signal especially when you are processing percussive sounds This noise is shaped by the envelope of the input signal The range of this control is from 0 to 100 When noise is set to 100 the original signal and the noise are equal in level Distortion Saturation The Distortion and Saturation sliders provide sig nal clipping control The Distortion slider determines the amount of gain applied and lets clipping occur in a smooth rounded manner The Saturation slider determines the amount of sat uration added to the signal This simulates the ef fect of tube saturation with a roll off of high fre quencies Output Meter The Output Meter indicates the output level of the processed signal Note that this meter indicates the output level of the signal not the input level If this meter clips the signal may have clipped on in put before it reached Lo Fi Monitor your send or insert signal levels closely to prevent this from hap pening Chapter 46 Recti Fi 283 Chapter 46 Recti Fi Recti Fi provides additive harmonic processing ef fects through waveform rectification Recti Fi fea tures the following effects Subharmonic synthesizer Full wave rectifier Pre filter for adjusting effect frequency Post filter for smoothing generated waveforms Recti Fi is available
226. ered string usu ally the lowest tone to the highest from left to right For example a six string guitar in standard tuning is shown as E2 A2 D3 G3 B3 E4 which are the notes and octaves for the sixth string through to the first string respectively Chapter 55 InTune 324 For best tuning results with guitars do the following Use headphones as loud monitors can modulate the guitar string Switch your guitar to its rhythm neck pickup if it has one Roll your guitar s tone knobs all the way off to remove all the highs Pluck the open string right over the twelfth fret not over the pickup To produce convenient test tones select the appro priate preset from the Librarian menu and select an appropriate test tone from the Test Tone menu Click a Note button to produce the appropriate test tone Test tones can be routed to headphones as re quired for musicians during session Chapter 56 MasterMeter 325 Chapter 56 MasterMeter MasterMeter is an oversampling meter plug in that is designed for critical mixing and mastering applica tions MasterMeter is available in DSP and Native formats MasterMeter Chapter 56 MasterMeter 326 MasterMeter Overview This section provides an overview of metering and mastering and how MasterMeter can help you pro duce great sounding mixes Understanding Digital Distortion Clients in the music industry regularly demand the loudest possible mixes
227. ereo bus compressor With the flick of a switch a punchy 8 1 compressor grabs the current mix producing instant radio hit It s also a handy way to make quick headphone submixes when tracking overdubs Give the Kids What They Want Shift click one of the Ratio Push switches to enable the All Buttons In mode The compression ratio is still only 20 1 but the knee changes drastically and the compressor starts mis behaving a little bit like an expander watch the meter for details Hey try it sometimes it even sounds good Selecting Proper Attack and Release Times As on the original unit setting either the attack or release time too fast generates signal distortion Again this may or may not be a effect A good starting point for attack and release is 6 and 3 the defaults and you can adjust as follows When compressing use the slowest attack you can that preserves a dynamic range Faster attacks re move the punch from the performance slower attacks inhibit the compression you need to smooth things out When limiting use the fastest attack time you can before you start to hear signal distortion in the low end With BF76 the attack time ranges from in credibly fast to really damn fast by modern stan dards It can be hard to hear the difference Release times are more critical with BF76 To set release times listen for loud attacks and what hap pens immediately after the peak
228. erial delivered without dither or level optimization Mastering engineers typically want to receive au dio material as undisturbed as possible in order to have leeway to adjust the level of the material rela tive to other material on a CD In such cases it is advisable to apply only the limiting that you find creatively appropriate adding a little punch to certain instruments in the mix for example However if you intend to produce 16 bit output from a source with a higher bit depth use appropri ate limiting and add dither Doing so will optimize the dynamic range and preserve the activity of the lower or least significant bits in the audio signal smoothly dithering them into the 16 bit output If you are using Maxim on an output mix that will be faded out enable the dithering options you want to improve the signal performance of the material as it fades to lower ampli tudes Chapter 18 Pro Compressor 97 Chapter 18 Pro Compressor Avid Pro Compressor is available in DSP Native and AudioSuite formats Pro Compressor provides dynamic compression processing The Avid Pro Compressor processing algorithms are based on the award winning Euphonix System 5 console chan nel strip effects Pro Compressor supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Pro Compressor supports mono stereo and greater than stereo multichannel formats up to 7 1 In addition to standard knob and fader con
229. es with a maximum resolution of 18 bits Dither For more information on using dither plug ins in Pro Tools see the Pro Tools Reference Guide If you are mixing down to an analog destination with any 24 bit capable interface you do not need to use Dither This allows maximum output fidelity from the 24 bit digi tal to analog converters of the interface Chapter 49 Dither 293 20 bit Recommended for output to digital devices that support a full 20 bit recording data path such the Sony PCM 9000 optical mastering recorder or the Alesis ADAT XT 20 The 20 bit setting can also be used for output to digital effects devices that support 20 bit input and output since it provides for a lower noise floor and greater dynamic range when mixing 20 bit signals directly in Pro Tools Noise Shaping Button The Noise Shaping button engages or disengages Noise shaping Noise shaping is on when the button is highlighted in blue Noise shaping can further improve audio perfor mance and reduce perceived noise inherent in dith ered audio Noise shaping uses filtering to shift noise away from frequencies in the middle of the audio spectrum around 4 kHz where the human ear is most sensitive The Dither plug in only provides eight chan nels of uncorrelated dithering noise If Dither is used on more than eight tracks the dither ing noise begins to repeat and dither perfor mance is impaired For example if two Quad Dithers are used bo
230. eshold will be reduced in gain Signals that are above it will be unaffected The Dynamics Graph display shows the threshold as a vertical line Attack The Attack control sets the attack time or the rate at which gain is reduced after the input signal crosses the threshold Use this along with the Ratio setting to control how soft the Expander s gain reduction curve is Ratio The Ratio control sets the amount of expansion For example if this is set to 2 1 it will lower signals be low the threshold by one half At higher ratio levels the Expander Gate functions like a gate by cutting off signals that fall below the threshold As you ad just the ratio control refer to the Dynamics Graph display to see how the shape of the expansion curve changes Depth The Depth control sets the depth of the Ex pander Gate when closed Setting the gate to higher range levels allows more and more of the gated au dio that falls below the threshold to peek through the gate at all times Hold The Hold control specifies the duration in seconds or milliseconds during which the Expander Gate will stay in effect after the initial attack occurs This can be used as a function to keep the Ex pander Gate in effect for longer periods of time with a single crossing of the threshold It can also be used to prevent gate chatter that may occur if varying input levels near the threshold cause the gate to close and open very rapidly Release
231. esponds with gentle dis tortion rather than abrupt clipping as in the digital domain Magnetic tape also has a frequency depen dent saturation characteristic that can lend punch to the low end and sweetness to the highs Reel Tape Saturation models the sonic characteris tics of analog tape including the effects of tape speed bias setting and calibration level of the modeled tape machine Reel Tape Saturation can be placed on mono ste reo or multichannel tracks Reel Tape Common Controls All Reel Tape plug ins share the following controls Drive Drive controls the amount of saturation effect by increasing the input signal to the modeled tape ma chine while automatically compensating by reduc ing the overall output Drive is adjustable from 12 dB to 12 dB with a default value of 0 dB Output Output controls the output signal level of the plug in after processing Output is adjustable from 12 dB to 12 dB with a default value of 0 dB Tape Machine The Tape Machine control lets you select one of three tape machine types emulated by the plug in each with its own sonic characteristics US Emulates the audio characteristics of a 3M M79 multitrack tape recorder Reel Tape Saturation Chapter 47 Reel Tape Saturation 287 Swiss Emulates the audio characteristics of a Studer A800 multitrack tape recorder Lo Fi Simulates the effect of a limited bandwidth analog tape device such as an outboard tape b
232. ess of an acoustical space Early reflections are followed by reverberation and repetitive reflec tions and attenuation of the original sound reflected from walls ceilings floors and other objects This sound provides a sense of depth or size Reverb One provides control over these reverbera tion elements so that you can create extremely nat ural sounding reverb effects Reverb One Controls Reverb One has a variety of controls for producing a wide range of reverb effects Controls can be ad justed by dragging their sliders or typing values di rectly in their text boxes The harmonic spectrum of the reverb can also be adjusted on the graph displays See Reverb One Graphs on page 164 Reverb One Master Mix Controls The Master Mix section has controls for adjusting the relative levels of the source signal and the re verb effect and also the width of the reverb effect in the stereo field Wet Dry Adjusts the mix between the dry unprocessed sig nal and the reverb effect Stereo Width Controls the width of the reverb in the stereo field A setting of 0 produces a mono reverb A setting of 100 produces maximum spread in the stereo field 100 Wet Toggles the Wet Dry control between 100 wet and the current setting Reverb One Dynamics Controls The Dynamics section has controls for adjusting Reverb One s response to changes in input signal level Dynamics can be used to modify a reverb s decay
233. est possible actual attack time for compression can only be about 20 ms The selected Detec tion mode similarly affects the compressor release time Chapter 18 Pro Compressor 104 Depth The Depth control sets the maximum amount of gain reduction applied regardless of the input sig nal For example if Ratio is set to LMTR between 20 0 1 and 20 0 1 and Depth is set to Off up to 20 dB of gain reduction is applied to the incoming signal at 0 dB If you set Depth to 10 dB no more than 10 dB of gain reduction is applied to the incoming signal Dry Mix The Dry Mix control sets the balance between the compressed signal wet and the original signal dry The Dry Mix setting determines how much of the original signal is sent to the output rather than the processed signal For example at 30 the out put will be 30 dry and 70 wet Turn the Dry Mix knob counterclockwise to 0 to pass only the pro cessed signal 100 wet Turn the Dry Mix knob clockwise to 100 to pass only the input signal 100 dry Makeup The Makeup control lets you boost overall output gain to compensate for heavily compressed or lim ited signals Pro Compressor Side Chain Processing Dynamics processors typically use the detected amplitude of their input signal to trigger gain re duction This is known as a side chain signal Pro Compressor provides filters for side chain process ing and supports external key side chain capabili ties
234. f the reverb HF Damp works in conjunction with HF Cut to shape the overall high frequency contour of the re verb HF Damp filters the entire reverb with the ex ception of the early reflections At low settings high frequencies decay more quickly than low fre quencies simulating the effect of air absorption in a hall The adjustable range is from 120 0 Hz to 24 0 kHz Reverb Contour Graph The Reverb Contour graph displays the envelope of the reverb as determined by the early reflections and reverb tail ER and RC Buttons Toggles the display mode Se lecting ER early reflections displays early reflec tions data in the graph Selecting RC reverb con tour displays the initial reverberation envelope in the graph Early Reflections and Reverb Contour can be displayed simultaneously Reverb Contour graph AAX version Chapter 28 Reverb One 166 Other Reverb One Controls In addition to its reverb shaping controls Reverb One also features online help and level metering Tool Tips To use tool tips move the cursor over the name of any control and an explanation appears as a tool tip Input Level Meters Input meters indicate the input levels of the dry au dio source signal Output meters indicate the output levels of the processed signal An internal clipping LED will light if the reverb is overloaded This can occur even when the input levels are relatively low if there is excessive feed back in the delay
235. fect can Drop pitch of ultra low bass Increase apparent bass power Unmask instruments or sonic details hidden in the mix Improve definition of high frequency sounds such as on cymbals and bells Improve speech articulation and presence Increase depth and clarity of male voices Chapter 42 Aphex Aural Exciter Type III 249 Successful use of the SPR depends on the character of the original sound It is hard to predict in ad vance what will be the effect of the SPR Typically you may find it useful about 50 of the time At other times there will be little or no discernible ef fect Seldom does the effect damage good audio so it could be left on as a matter of course A few examples of audio material likely to be helped by the SPR are Highly overdubbed tape tracks Live acoustic recordings Electronic keyboards Productions composed from tape cartridge and cassette sources Material recorded with transformer coupled mic preamps Vocals recorded with dynamic microphones Reverberant live sound or recordings Highly equalized material Delayed flanged or digitally processed material Material from broadcast audio reception such as store casting and muzak Chapter 43 Aphex Big Bottom Pro 250 Chapter 43 Aphex Big Bottom Pro Aphex Big Bottom Pro is an AAX plug in that retains the look and sound of Aphex Systems renowned hardware units
236. fect will be applied in the same way Reel Tape Flanger Presets 12 String Moderate depth ADT setting that works well with acoustic guitar sounds Flutter Flange Moderate depth flange setting with Wow Flutter Flutter Extreme Wow Flutter setting with flanging turned off and a Mix setting that passes only the variable delay Manual Flange Settings with LFO Depth set to zero ready for manual flanging by adjusting or au tomating the Range control Slow Flange High Depth setting combined with slow LFO Rate suitable for flanging vocals Vocal ADT Settings for creating doubling effect without flanging crossover effect suitable for vocals Vocal Walrus Drive boosted settings for extreme vocal doubling effect Wobble A high LFO Rate setting combined with a Mix setting that passes only the variable delay Works well on sustained parts Chapter 40 Sci Fi 232 Chapter 40 Sci Fi Sci Fi features analog synthesizer type effects that include Ring modulation Frequency modulation Variable frequency positive and negative reso nator Modulation control by LFO envelope follower sample and hold or trigger and hold Sci Fi is available in DSP Native and AudioSuite formats Sci Fi supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Sci Fi operates as a mono multi mono or stereo plug in Sci Fi is designed to mock synthesize audio by adding effects such as ring mod
237. ffect as the variable delay crosses the zero point ADT Artificial Double Tracking Range settings outside the narrow center band simulate artificial double tracking in which the variable delay does not cross the zero point This varying delay cre Operation with Flange Range setting no offset zero point LFO Depth LFO Rate Chapter 39 Reel Tape Flanger 229 ates a unique doubling effect essentially an analog precursor to chorusing You can hear ADT type effects on many classic analog recordings such as those of the Beatles or Led Zeppelin Feedback The Feedback control adds a short delay to the flanged signal Feedback amount is adjustable from 0 to 100 percent with a default value of 0 percent This is not the same feedback effect as on an elec tronic flanger or delay Wow Flutter The Wow Flutter control adjusts the amplitude of the variable delay tape machine s wow and flutter or the amount of fluctuation in tape speed A higher setting results in wider fluctuations in speed A lower setting results in narrower fluctuations in speed Wow Flutter is adjustable from 0 to 1 per cent with a default value of 0 03 percent Rate The LFO Rate control adjusts the rate of change in the variable delay A higher setting results in faster fluctuations in speed A lower setting results in slower fluctuations in speed LFO Rate is adjust able from 0 05 Hz to 5 Hz with a default setting of 0 14
238. fferent tape speeds 3 75 ips Sets the delay time to correspond to a Speed Control setting of 3 75 inches per second 3 75 ips Flutter Includes the 3 75 ips setting plus Wow Flutter 7 5 ips Sets the delay time to correspond to a Speed Control setting of 7 5 inches per second 7 5 ips Flutter Includes the 7 5 ips setting plus Wow Flutter 30 ips Flutter Adds Wow Flutter to the highest Speed Control setting Rockabilly A common tape slap effect useful on vocals or electric guitar Sets the delay time to 130 ms which corresponds to the delay time result ing from the distance between the record and play heads on an Ampex 300 series or Ampex 350 se ries tape transport Rockabilly Plus Includes the Rockabilly setting plus Feedback Wow Flutter Bass and Treble ad justments on feedback BPM Sync controls You can override the settings derived from BPM Sync at any time by manually adjusting the plug in Speed control To set the delay time to a specific time value turn off BPM Sync and enter the delay time in msec in the Tempo Rate display Tempo Rate Triplet button On button Dot button Note Value display display Chapter 34 Tel Ray Variable Delay 213 Chapter 34 Tel Ray Variable Delay Tel Ray Variable Delay is a delay echo plug in that is available in DSP Native and AudioSuite for mats Tel Ray Variable Delay supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates
239. fields allow entry of individ ual notes from A0 to G7 Flat semitones are entered with a b for example Ab2 and sharp semitones are entered with a hash or pound character for ex ample A 2 To clear an entry enter Note fields are committed by pressing Return Macintosh or Enter Windows If you do not press Return or Enter the note field will return to the previous value entered InTune will automati cally justify the note buttons as needed so they fit in the correct area on the main screen The Note Entry fields are not available in Chromatic mode Exit In the Tuner Programming screen click the Exit button to return to the main InTune screen Using InTune When InTune detects a signal the meter lights up and displays the relative pitch of the incoming sig nal With stringed instruments this will vary during the attack and decay of the note In Automatic mode InTune estimates the note to which you are trying to tune If the correct note is not lit in automatic mode click on the note to which you are trying to tune for greater accuracy This will lock InTune to the specified note The meter will display the frequency of the note de tected and the accuracy is displayed on a scale of plus minus 50 cents In addition the information display will display the note and the number of cents from perfect tuning When loading factory presets stringed instruments are laid out from the highest numb
240. for calculating TCE Adjust the Time control to change the target duration for the processed audio Original The Original column displays the Start and End times and Length of the edit selection Times are displayed in units of the timebase selected in the Units pop up menu Processed The Processed column displays the target End time and Length of the processed signal Times are dis played in units of the timebase selected in the Units pop up menu You can click the Processed End and Length fields and type values These values update automatically when adjusting the Time control Tempo The Tempo row displays the Original Tempo and Processed Tempo in beats per minute bpm You can click the Original Tempo and Processed Tempo fields and type values The Processed Tempo value updates automatically when adjusting the Time control Unit Select a timebase for the Original and Processed time fields Bars Beats Min Sec Timecode Feet Frames or Samples When previewing Polyphonic Poly Faster is used for faster previewing However when you process the audio selection the high quality Polyphonic setting is used X Form Time section Chapter 26 X Form 149 Shift The Shift setting displays the target time compres sion or expansion as a percentage of the original Adjust the Time control or click the Shift field and type a value Time can be shifted by as much as 12 50 to 800 00 of the original speed or 8 time
241. for heavily compressed or limited signals Side Chain Processing Controls Dynamics processors typically use the detected amplitude of their input signal to trigger gain re duction This split off signal is known as the side chain Compressor Limiter and Expander Gate processing features external key capabilities and filters for the side chain With external key side chain processing you trig ger dynamics processing using an external signal such as a separate reference track or audio source instead of the input signal This external source is known as the key input With side chain filters you can make dynamics processing more or less sensitive to certain fre quencies For example you might configure the side chain so that certain lower frequencies on a drum track trigger dynamics processing Source The Source selector lets you set the source for side chain processing Internal Key or All Linked Internal If Internal is selected the plug in uses the amplitude of the input signal to trigger dynamics processing With greater than stereo multichannel processing the input signal for each stereo pair ef fects only those same channels and likewise mono channels are effected only by their own input sig nal For example with an LCR multichannel for mat the processing for the Center channel is only triggered when the Center channel input signal Dynamics section Side Chain tab Chapter 11 Channel Strip 54 reaches
242. fre quency do not adhere to Nyquist or Shannon s the orems regarding allowable frequencies cannot be reproduced and are therefore considered illegal frequencies Because of mathematical realities ob served by Fourier in the 1800s and subsequently by Shannon in 1948 when a waveform has all fre quencies removed above the Nyquist frequency the resulting waveform will be the original wave form that was sampled This process is significantly more involved than simply connecting the dots between sample points Today it involves extremely sophisticated means of reconstructing the waveform using filters that are highly complex mathematical systems uti lizing oversampling upsampling linear phase equiripple FIR designs and much more Oversampling creates a more accurate digital rep resentation of an analog signal by sampling some number of times per second frequency and con verting into digital form Oversampling requires at least twice the bandwidth of the frequency being sampled For example a consumer CD player us ing 2x oversampling is processing information at 88 2 kHz The result is that today s digital to analog convert ers get closer to the original than ever before mak ing music played on systems today as accurate as possible Even today s inexpensive components such as off the shelf CD players have drastically improved filters and thus better reconstruction abil ities than in years
243. from 100 to 100 At 0 the early reflections are set to their optimum value for the room preset Typical spread values range between 25 and 25 Pre Delay Control The Pre Delay control in the Early Reflect section determines the amount of time that elapses between the onset of the dry signal and the first early reflec tion delay tap Some Room Types such as those that produce slapback effects have additional built in pre delay The range of this control is from 300 0 ms to 300 0 ms Negative Pre Delay times imply that some early re flection delay taps should sound before the original dry signal Since this is not possible any of the de lay taps that would sound before the dry signal are not used and do not sound When Pre Delay Link is enabled negative early re flection Pre Delay times can be used to make the early reflections start before the reverb tail Pre Delay Link Button The Early Reflections Pre Delay Link button tog gles linking of the Early Reflection Pre Delay con trol and the Reverb Pre Delay control When linked the Early Reflection Pre Delay is offset by the Reverb Pre Delay amount so that the total de lay for the early reflections is the sum of the Early Reflection Pre Delay and the Reverb Pre Delay Early Reflections On Button This button toggles early reflections on or off When early reflections are off the reverb effect consists entirely of reverb tail ReVibe II Room Coloration
244. g rhythmic effects where modulation occurs primar ily on signal peaks Modulation will occur in a pe riodic yet random way that varies directly with peaks in the audio material Think of this type of modulation as having the best elements of both sample and hold modulation and with an envelope follower Sci Fi Mod Amount and Mod Rate Controls These two sliders control the amplitude and fre quency of the modulating signal The modulation amount ranges from 0 to 100 The modulation rate when LFO or Sample Hold are selected ranges from 0 1 Hz to 20 Hz If you select Trigger Hold as a modulation type the Mod Rate slider changes to a Mod Threshold slider which is adjustable from 95 dB to 0 dB It determines the level above which modulation oc curs with the trigger and hold function If you select Envelope Follower as a modulation type the Mod Rate slider changes to a Mod Slew ing slider which is adjustable from 0 to 100 Sci Fi Output Meter The Output Meter indicates the output level of the processed signal Note that this meter indicates the output level of the signal not the input level If this meter clips the signal may have clipped on in put before it reached Sci Fi Monitor your send or insert signal levels closely to prevent this from hap pening Chapter 41 Voce Plug Ins 235 Chapter 41 Voce Plug Ins The Voce plug ins provide a pair of vintage modu lation effect plug ins Voce Chorus Vibrato and
245. g amounts of Speaker Breakup Try different mics and positions to hear how they affect the track 7 Apply other plug ins or bus the Aux Input to an other track for additional processing To process and re record tracks through Eleven 1 Import and place your audio in a Pro Tools audio track 2 Configure the source audio track by doing the following Assign the audio track Output a bus such as Bus 1 if mono or Bus 1 2 if stereo Click the Insert selector and select Eleven 3 Choose Track gt New and create one mono audio track 4 Configure the new audio track by doing the fol lowing Click its track Input selector and choose the Bus 1 or Bus 1 2 Click the Insert selector and select Eleven 5 Record enable the new audio track or enable TrackInput monitoring if using Pro Tools HD 6 Begin playback and start listening 7 While listening adjust Eleven s Input knob to increase or decrease input level 8 When everything sounds and looks good locate to where you want to begin recording or make a time selection arm the Pro Tools Transport and press Play to start recording Chapter 44 Eleven 274 Blending Eleven Cabinets and Amps You can use Eleven for multi cabinet and multi amp setups so you can blend their signals together This classic technique lets you get tones that no sin gle combo cabinet or amp could produce Unlike working with real amps this is simple to ach
246. g signal is attenuated when the side chain input crosses the Threshold setting For the most predictable results set Ratio to GATE Use the Depth control to adjust the amount of ducking ap plied to the input signal Dialog or voice over is most commonly used as an external key input to duck attenuate the program material typically music on the track where Pro Expander is inserted Cursor indicates Depth adjustment Detection mode options Smart mode selected As a general rule when ducking program ma terial with dialog or voice over set the Attack and Release controls in a range of 300 to 500 milliseconds When Duck is enabled the Upward option is automatically disabled Chapter 19 Pro Expander 112 Attenuation Listen Mode Attenuation Listen mode lets you isolate the pro cessed part of the audio signal This can help you hear what parts of the input signal are triggering ex pansion or gating which in turn can help you bet ter understand the characteristics of the current set tings of the expander gate To enable or disable Attenuation Listen mode Click the Attenuation Listen button the speaker icon at the top right of the dynamics graph so that it is highlighted The button flashes while Attenuation Listen mode is enabled To disable it click the button again so that it is not high lighted Pro Expander Controls Threshold The Threshold control sets the level below which an input signal must fal
247. ge the Frequency Graph Gain resolution Click the Graph Resolution toggle EX Filters section High Mid Frequency tab shown Frequency x axis Gain y axis Graph Resolution toggle EQ control point Filter control point Chapter 11 Channel Strip 56 Dragging in the Frequency Graph to Adjust Controls You can adjust the following EQ controls by drag ging the control points directly in the Frequency Graph display Frequency Dragging a control point to the right in creases the Frequency setting Dragging a control point to the left decreases the Frequency setting Gain Dragging a control point up increases the Gain setting Dragging a control point down de creases the Gain setting Q Click within the curve of an EQ control point and drag up or down to increase or decrease the Q set ting Low Frequency EQ Controls The LF tab provides controls for the low frequency band of the EQ The low frequency band can be set to be a Peak or Low Shelf EQ EQ Type Select either the Peak or Low Shelf button to set the EQ type for the low frequency band Frequency The Frequency control lets you set the center fre quency for the low frequency band Peak or Shelf EQ Gain The Gain control lets you boost or attenuate the corresponding frequencies for the low frequency band Q With the low band EQ set to Peak the Q control changes the width of the EQ band Higher Q values represent narrower bandwidths L
248. gh experimentation But don t forget to try using it in subtle ways add ing just a hint to harshen up or add a metallic quality to individual tracks buried in the mix Al most all the great MOOG sounds feature subtle clever uses of Ring Modulation Chapter 39 Reel Tape Flanger 227 Chapter 39 Reel Tape Flanger Reel Tape Flanger is part of the Reel Tape suite of tape simulation effects plug ins Reel Tape Flanger simulates a tape machine flanging effect modeling the frequency sweep and crossover comb filter ing effects that can result when the flanger variable delay is adjusted It also reproduces the frequency response noise wow and flutter and distortion characteristics of analog tape recording Reel Tape Flanger is available in DSP Native and AudioSuite formats Reel Tape Flanger supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Reel Tape Flanger operates as a mono multi mono or stereo plug in For years engineers have relied on analog tape to add a smooth warm sound to their recordings When driven hard tape responds with gentle dis tortion rather than abrupt clipping as in the digital domain Magnetic tape also has a frequency depen dent saturation characteristic that can lend punch to the low end and sweetness to the highs Reel Tape Flanger models a classic tape flanging setup with two analog tape machines and a mixer where one tape machine has a fixed del
249. gram material and the parameters used this rep resents an approximate range of 100 s to 80 milli seconds Smack Ratio Control In the Norm and Warm modes Ratio controls the compression ratio or the amount of compression applied as the input signal exceeds the threshold For example a 2 1 compression ratio means that an input level that is 2 dB above the threshold will be attenuated resulting in an output level that is 1 dB over the threshold As you increase the Ratio control Smack goes from applying soft knee compression to hard knee compression as follows With soft knee compression gentle compression begins and increases gradually as the input signal approaches the threshold This creates smoother compression In hard knee compression compression begins when the input signal exceeds the threshold This can sound abrupt and is ideal for limiting or de essing Smack compression ratios range from subtle com pression to hard limiting At ratios of 10 1 and higher Smack functions as a limiter Selecting the Smack setting lowers the threshold slightly and applies hard limiting which keeps the output level constant regardless of the input level This setting can also be used for extreme compression effects Smack Release Control In Norm and Warm modes Release controls the length of time it takes for the compressor to be fully deactivated after the input signal drops below the threshol
250. h the online Knowledge Base or join the worldwide Pro Tools community on the User Conference Training and Education Study on your own using courses available online or find out how you can learn in a classroom setting at a certified Pro Tools training center Products and Developers Learn about Avid products download demo software or learn about our Development Partners and their plug ins applications and hardware News and Events Get the latest news from Avid or sign up for a Pro Tools demo Chapter 2 Installing and Authorizing Avid Paid Plug Ins 7 Chapter 2 Installing and Authorizing Avid Paid Plug Ins A core set of audio plug ins is installed automati cally with your version of Pro Tools No additional steps are required to authorize these plug ins for use on your Pro Tools system Installers for additional plug ins purchased or rented from the Avid store shop avid com can be downloaded from your online Avid account These plug ins are authorized using an iLok USB key About iLok All paid plug ins from Avid are authorized using an iLok USB key from PACE Anti Piracy An iLok can hold hundreds of authorizations for all of your iLok enabled software After a software license is placed on an iLok you can use the iLok to authorize that software on any computer An iLok USB key is not supplied with plug ins or software options You can use the iLok included with certain Pro Tools systems or purchase one separate
251. h the ratio set at 10 1 and higher Large ratios effectively limit the dy namic range of the signal to a specific value by set ting an absolute ceiling for the dynamic range Compressor Limiter III Input Output Level Meters The Input and Output meters show peak signal lev els before and after dynamics processing See Dy namics III Levels Section on page 59 for more in formation Unlike scales on analog compressors metering scales on a digital device reflect a 0 dB value that indicates full scale fs the full code signal level Compressor Limiter III Graph Display The Dynamics Graph display lets you visually see how much expansion or gating you are applying to your audio material See Dynamics III Graph Dis play on page 61 Compressor Limiter III Threshold Control The Threshold Thresh control sets the level that an input signal must exceed to trigger compression or limiting Signals that exceed this level will be compressed Signals that are below it will be unaf fected This control has an approximate range of 60 dB to 0 dB with a setting of 0 dB equivalent to no com pression or limiting The default value for the Threshold control is 24 dB An orange arrow on the Input meter indicates the current threshold and can also be dragged up or down to adjust the threshold setting The Dynamics Graph display also shows the threshold as an orange vertical line This control ranges from 60 dB
252. have long rough envelopes and often sound better with less dramatic changes in the filter Other sounds like bass or snare drum are quick and sharp and sound great when the filter closely tracks their at tack Mix The Mix control blends the original input sig nal with the filtered signal Use it to get any mixture of filtered and unfiltered sound Filter Section Control the filter using the Cutoff and Resonance knobs and the 2 Pole 4 Pole switch Cutoff Cutoff opens and closes the filter Turned counterclockwise fewer high frequencies pass through the filter Turned clockwise more high fre quencies pass Resonance Resonance changes the way the filter sounds At low resonance low frequencies come through evenly At high resonance frequencies near the cutoff frequency are boosted creating a whistling or vowel type quality When resonance is maxed out the filter oscillates and produces its own tone at the cutoff frequency This oscillation interacts with other tones as they go through the fil ter producing the signature Moog sound Envelope Time Time Envelope of the sound Audio waveform of the sound Audio waveform of a musical sound Envelope signal of the same sound The 2 4 pole switch selects the filter slope Frequency Gain 4 Pole 2 Pole Chapter 36 Moogerfooger Lowpass Filter 221 2 Pole 4 Pole The 2 Pole 4 Pole switch selects whether the signal goes through half the filter 2
253. he maximum setting used for input calibration Don t worry about the Input LED showing yel low or orange when playing normally As long as the plug in isn t indicating clipping your gain staging should be established 5 Adjust the Output knob in Eleven s Master sec tion to raise or lower the plug in output signal Using Eleven with Pre Recorded Tracks If the pre recorded tracks were not calibrated with the Eleven plug in using the method previously de scribed you can use the Input control in Eleven to adjust the signal level feeding the input stage of the amp model Use your ears as a guide and adjust to taste Since the Input LED measures the signal level entering the plug in and precedes the input control you will not see any changes to the Input LED as you make adjustments Eleven s Input LED top and Clip LED bottom Proper input calibration of live guitar does not require any adjustment of Eleven s Input con trol To learn how this control was designed to work with the amp models see Input on page 264 See Processing Pre Recorded Tracks Through Eleven on page 273 for more information Chapter 44 Eleven 261 Getting Started Playing Music with Eleven To get started playing music with Eleven 1 Make sure you already calibrated your input sig nal as explained in the previous sections of this chapter 2 Click the plug in Librarian menu and choose a factory preset then play guitar
254. he Lo Hi switch selects the range of the Rate control When the switch is Lo the Rate con trol varies from 0 01 Hz one cycle every hundred seconds to 2 5 Hz 2 5 cycles every second When the switch is Hi the Rate control varies from 2 5 Hz 2 5 cycles every second to 250 Hz two hundred fifty cycles per second With such a wide range of rates available obviously you ll need to adjust Rate after you flick the Lo Hi switch to get the sound you desire Phaser Section Control the Phaser with the Sweep and Resonance knobs and the 6 Stage 12 Stage switch Resonance Resonance adjusts the feedback of the analog filters As you add more resonance the peaks caused by the filters get sharper and more no ticeable Sweep Sweep adjusts the center frequency point of the filters Use it in conjunction with Amount to control the frequencies affected by the phaser Drive The Drive control sets the input gain LED Indicators Three LEDs provide visual feedback Level Level glows green when signal is present LFO LFO blinks to show the LFO rate Bypass Bypass glows either red bypassed or green not bypassed to show whether or not the ef fect is in the signal path Responses of a phaser with high resonance Sweep adjusts the center frequency point Frequency Mid Shift Frequency Gain 1 Frequency Mid shift frequency Gain 1 moves Chapter 37 Moogerfooger 12 Stage Phaser 224 Moogerfooger 12 Stage Pha
255. he Meter track for the session If this option is disabled un highlighted you can set the rhythmic values for Click 1 and Click 2 independently of the Meter track To enable or disable the Follow Meter option Click the Follow Meter toggle below the Beat display BPM Display The BPM display shows the current tempo in the session If Tempo is set manually with the Con ductor track disabled the tempo as set in the Trans port window is displayed If the Conductor track is enabled the Tempo at the current location of the Playback Cursor is displayed Click II Click II does not sound if the Click option is disabled Options gt Click or if the Click track has been muted Chapter 53 Click II 308 ON Button and MIDI IN LED Click the ON button to manually turn the Click on or off It is on when it is lit Just below the ON but ton is an LED that illuminates each time the Click plug in receives a click message from the Pro Tools application indicating the click tempo Click 1 The Click 1 section provides controls for the down beat click Accent Fader The Accent fader lets you set the relative strength of the accent output MIDI velocity for the down beat click Click Sound Selector You can choose from several click sound options using the Click Sound selector Click Beat Value If the Follow Meter option is disabled you can manually set the rhythmic value for the downbeat click If the Follow
256. he Offset field Chapter 56 MasterMeter 332 MasterMeter Offset Field The Offset field offsets the values displayed in both the browsers by the value entered This is useful for historical metering but the session was started from a point other than the beginning The Enter key must be used after a new offset is typed for it to be come active The information shown in the brows ers is updated immediately when the new Offset is entered For example if the session was started from the point 1 03 901 1 minute 3 901 seconds this value should be entered into the Offset to ensure the time code displayed in both of the browsers matches that of the Pro Tools session MasterMeter Clip Field The Clip field can be used to set the clip threshold at a lower point For example if a session must not exceed 10 dB the Clip field can be set to 10 dB and MasterMeter will treat that as the clip threshold for both signal and oversampled clip events When the Clip field is set to a non zero value the Min and Max values of the Signal Clip browser are used to indicate the clip range Chapter 57 Signal Generator 333 Chapter 57 Signal Generator The Signal Generator plug in produces audio test tones in a variety of frequencies waveforms and amplitudes It is particularly useful for generating reference signals with which to calibrate audio in terfaces and other elements of your studio Signal Generator is a mono or multi mono
257. he processed vocal Generator Leakage Of all the sounds to pass through a Leslie no sound has been amplified more often than the sound of B3 Organ generator leakage Even with no notes keyed a small amount of B3 sound leaks out Part VIII Harmonic Plug Ins Chapter 42 Aphex Aural Exciter Type III 241 Chapter 42 Aphex Aural Exciter Type III Aural Exciter Type III is an AAX plug in that retains the look and sound of Aphex Systems renowned hard ware units Aural Exciter makes it possible to recreate and restore missing harmonics Aphex Systems Inc first introduced Aural Exciter in 1975 Since then several refinements and im provements have been incorporated into its original design The Aural Exciter plug in is modeled after the TYPE III Aural Exciter Aural Exciter has be come a standard in the professional audio industry and has been used on many albums CDs movies broadcast productions commercials and concerts The Aural Exciter plug in for Pro Tools continues this tradition of success and is ready for use with the latest cutting edge music productions Harmonics are musically and dynamically related to the original sound and reveal the fine differ ences between voices and various instruments Re produced sound is audibly different from the origi nal live sound because of the loss in harmonic detail often sounding dull and lifeless Aural Exciter is an audio process that recreates and restores missing har
258. he rear channels The range of this control is from 0 to 100 ReVibe II Early Reflection Section Different physical environments have different early reflection signatures that our ears and brain use to pinpoint location information in physical space These reflections influence our perception of the size of a space and where an audio source sits within it Changing early reflection characteristics changes the perceived location of the reflecting surfaces surrounding the audio source In general the reverb tail continues after early reflections dissipate ReVibe II room presets use multiple delay taps at different levels different times and in different po sitions in the multichannel environment through 360 panning to create extremely realistic sound ing environments Chapter 29 ReVibe II 172 The Early Reflect section has controls for adjusting the various early reflection elements including level spread and pre delay Level Control Level controls the output level of the early reflec tions Setting the Level slider to INF minus infin ity eliminates the early reflections from the reverb effect The range of this control is from INF to 6 0 dB Spread Control Spread globally adjusts the delay characteristics of the early reflections moving the individual delay taps closer together or farther apart Use Spread to vary the size and character of an early reflection preset The range of this control is
259. he time information displayed in this browser is relative to where the transport started The Offset field can be used to adjust the timecode values if MasterMeter is being used for historical metering but the session was started from a point other than the beginning If MasterMeter is being used in real time the timecode information in this browser can be ignored At the bottom of the browser the Peak field dis plays the highest dB value of the audio signal re ceived so far The Events field shows the historical total of clip events in the audio signal Once Mas terMeter reaches 2 000 clip events it ceases to re cord additional events Although the meters remain active and the Peak field continues to be updated new events will not be added to the browsers The Events field flashes 2000 to indicate this condi tion The information in this browser is cleared using the Clear button or is cleared automatically whenever the Pro Tools transport is started Oversampled Clip Events Browser The Oversampled Clip Events browser displays historical clip events from the DSP oversampling of the session audio The amount of potential clip ping in excess of 0 dB is also displayed The columns displayed show the relevant timecode for the beginning and ending of a clip event as well as the minimum and maximum clip values created after passing through the DSP processing When used in a stereo track the first column shows L or R to i
260. her settings lengthen both the attack and buildup of the initial reverb contour The range of this control is from 0 to 100 Pre Delay Control The Pre Delay control in the Reverb section sets the amount of time that elapses between signal in put and the onset of the reverb tail Under natural conditions the amount of pre delay depends on the size and construction of the acoustic space and the relative position of the sound source and the listener Pre delay attempts to duplicate this phenomenon and is used to create a sense of dis tance and volume within an acoustic space Ex tremely long pre delay settings produce effects that are unnatural but sonically interesting The range of this control is from 0 0 ms to 300 0 ms Attack Time Control Attack Time adjusts the length of time between the start of the reverb tail and its peak level Settings are Short Medium or Long Attack Shape Control Attack Shape determines the contour of the attack portion of the reverberation envelope At 0 there is no buildup contour and the reverb tail begins at its peak level At a high Attack Shape setting the re verb tail begins at a relatively low initial level and ramps up to the peak reverb level The range of this control is from 0 to 100 Rear Shape Control Rear Shape adjusts the envelope of the reverb in the rear channels to control the length of the attack time This gives more reverb presence and a longer reverb bloom in t
261. higher average audio levels potentially better signal to noise ratio and a smoother mix Limiting Individual Instruments The primary purpose of applying limiting to indi vidual instruments is to alter their dynamic range in subtle or not so subtle ways A common applica tion of this type of limiting is to modify the charac ter of drums Many engineers do this by applying heavy limiting to flatten the snap of the attack por tion of a drum hit By adjusting the release time of the limiter it is possible to bring up room tone con tained in the decay portion of the drum sound In some cases this type of limiting can actually change a drum s character from a very dry sound to a relatively wet sound if there is enough room tone present This method is not without its drawbacks however since it can also bring noise levels up in the source audio if present Chapter 17 Maxim 93 How Maxim Differs From Conventional Limiters Maxim is superior to conventional limiters in sev eral ways Unlike traditional limiters Maxim has the ability to anticipate signal peaks and respond instantaneously with a true zero attack time Maxim does this by buffering audio with a 1024 sample delay while looking ahead and analyzing audio material on disk before applying limiting Maxim can then instantly apply limiting before a peak builds up The result is extremely transparent limiting that faithfully preserves the attack tran sients and retains
262. ho and others popularized the early 70s British trick of combining a slower com pressor with a faster one The effect can produce very interesting sounds Try applying Peak Reduc tion using the BF 2A then squash the missed at tacks using the faster BF76 For more information on plug in automation see the Pro Tools Reference Guide Chapter 9 BF 3A 40 Chapter 9 BF 3A BF 3A is a vintage style compressor plug in that is available in DSP Native and AudioSuite formats BF 3A supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates BF 3A operates as a mono multi mono or stereo plug in BF 3A is based on the classic LA 3A that adds a smoothness and sonic texture that makes sounds jump right out of the mix Designed and manufac tured in the late 1960s the original LA 3A shares many components in common with the LA 2A compressor Just like the LA 2A the heart of the LA 3A is the T4B Electro Optical Attenuator This is a device that converts audio to light and back and is largely responsible for the compression character of the unit While the LA 2A s gain comes from a tube ampli fier the LA 3A s gain comes from a solid state transistor amplifier This gives the LA 3A a solid midrange and more aggressive tone Other subtle modifications change the behavior of the T4B causing it to respond differently particularly in response to percussive material The LA 3A is famous for its unique s
263. hot Session and preset file sizes will increase as Space stores each IR waveform inside the file This provides maximum compatibility be tween different Pro Tools systems without the need for them to have identical IR libraries IR embedding can be disabled in Space s Prefer ences If IR embedding is disabled Space stores only a reference to the name of the IR file When the session is transferred to a different system Space attempts to load the matching IR file from the Space IR library For maximum compatibility ensure that all of the appropriate IR files are avail able on the new system When working with an IR that only exists in a ses sion file ensure it is saved to a separate snapshot or preset If the IR is overwritten by loading a new IR and the session is saved the original IR cannot be recovered without access to the original IR file IR files are audio files only and do not con tain information about Space control set tings If you wish to save specific control set tings for an IR you should save them using the Pro Tools Plug In Librarian or using the snapshot facility of Space By default Pro Tools presets or session files created using Space automatically include copies of all relevant IR waveforms This pro vides maximum compatibility of session files between different Pro Tools systems It is your responsibility to ensure that you ob serve the copyright on any IR transferred to a third party in this fa
264. how much dynamics processing you are applying to the incoming audio signal Using the Dynamics Graph to Adjust Controls You can drag in the Dynamics Graph display to adjust the corresponding Expander controls The cursor updates to show which control is being adjusted Threshold Ratio Knee Depth To adjust the Threshold setting using the Dynamics graph Position the cursor over the vertical Threshold line in the graph and drag left or right to make the adjustment To adjust the Ratio setting using the Dynamics graph Position the cursor over the ratio curve in the graph and drag up or down or left or right to make the adjustment To adjust the Knee setting using the Dynamics graph Position the cursor over the knee of the curve in the graph and drag up or down or left or right to make the adjustment Dynamics graph display Input signal level x axis Threshold Amount of processing y axis Cursor indicates Threshold adjustment Cursor indicates Ratio adjustment Cursor indicates Knee adjustment Chapter 19 Pro Expander 111 To adjust the Depth setting using the Dynamics graph Position the cursor over the horizontal Depth line in the graph and drag up or down or left or right to make the adjustment Detection Modes Pro Expander provides several different detection options for determining how the expander responds to the input signal To ch
265. hronize the LFO to MIDI Beat Clock for rhythmic panning effects To synchronize AutoPan to MIDI Beat Clock 1 Make sure that your session tempo matches the tempo of the music 2 Insert a mono to stereo instance of AutoPan on the mono audio track containing the audio you want to pan The track s channel width changes from mono to stereo Tempo display Envelope section When Envelope Env is not selected as the Panning Source the controls in this section have no effect on the sound Side Chain Input selector enabled Chapter 51 AutoPan 303 3 In the AutoPan Plug In window enable Link To Tempo This sets the LFO rate to follow the ses sion tempo 4 Select a duration from the Duration selector For example select 2 Beats 5 Select a waveform for the LFO from the Wave form selector For example select 4 Step Trian gle 6 Enable Beat Clock for the LFO Trigger This en sures that the LFO is synchronized to the beat 7 Play back the session to hear the panning effect Post Production Panning Pro Tools HD Only AutoPan lets you pan a mono track to a greater than stereo LCR Quad or 5 0 output in a surround path This is especially useful for post production applications The following example describes how to use TL AutoPan to pan a mosquito sound in 5 0 surround To pan a mono track to 5 0 with AutoPan 1 Insert a mono to 5 0 instance of AutoPan on the mono track containing the audio you want t
266. ht outputs The range of this control is from INF minus in finity to 0 0 dB Rear Reverb Control Rear controls the output level of the rear outputs of multichannel formats that have rear channels such as quad or 5 1 When ReVibe II is used in a multichannel format that has no rear channels such as a stereo or LCR the Rear level control instead adjusts rear channel signals hard panned to the front left and right out puts The range of this control is from INF minus in finity to 0 0 dB Rear Early Reflections Control Rear ER controls the output level of early reflec tions in the rear outputs The range of this control is from INF minus infinity to 0 0 dB Rear Level Link Button The Rear Level Link button toggles linking of the Rear Reverb and Rear Early Reflections controls on or off The Rear Reverb and the Rear Early Re flections controls are linked by default When linked the Rear Early Reflections and Rear Reverb The Rear ER control has no effect when the early reflections are turned off with the ER On Off button Chapter 29 ReVibe II 174 controls move in tandem when either is adjusted When unlinked the Rear Early Reflections and the Rear Reverb controls can be adjusted inde pendently ReVibe II Chorus Section Controls The Chorus section has controls for adjusting the depth and rate of chorusing applied to the reverb tail Chorusing thickens and animates sounds and produces a more
267. ick the track s MIDI Out put selector a and select the ReWire client appli cation Some ReWire clients such as Reason may list multiple devices If so choose the de vice that you want 6 Choose Options gt MIDI Thru and record enable the MIDI track Play some notes on your MIDI controller to trigger the client application The selected ReWire device responds to MIDI sent from Pro Tools and plays back audio through the assigned Pro Tools track Instrument Auxiliary Input or audio track If your ReWire client application is a sequencer and you want to begin synchronized playback with Pro Tools press the Spacebar or click the Play but ton on the Pro Tools Transport MIDI Automation with ReWire You can use Pro Tools MIDI tracks to record MIDI continuous controller CC data from a ReWire cli ent application and then play back MIDI from Pro Tools to send the recorded MIDI CC data back to the ReWire client application In this way you can adjust controls in the ReWire client application using the mouse or an external MIDI controller and record those changes in Pro Tools Recording MIDI Continuous Controller Data Over ReWire The first step in automating a ReWire client appli cation s controls is to record the CC data to a MIDI track in Pro Tools Selecting the ReWire client device to receive MIDI from Pro Tools Instrument track shown Chapter 54 ReWire 315 To record MIDI from a ReWire client application in
268. ider controls the width of the panning field At 100 the panning field is at its widest At 0 the panning field is centered and stationary The Width slider effectively determines the amount of LFO or Envelope control on the pan position Manual The Manual slider directly controls the pan posi tion this lets you manually control the pan position from a control surface or by using automation The amount of manual control is affected by the setting of the Width slider For full manual control set the Width slider to 0 When the Width slider is at 100 the Manual slider has no effect on the pan position When Width is set to 50 the LFO sweeps the position through 50 of its range and the Manual slider lets you move the position of that 50 range Angle The Angle slider adjusts the orientation of the pan ning field from 90 to 90 At 0 the panning field is oriented strictly left right At 90 or 90 the panning field is oriented strictly front back The Angle slider is only available with mono to quad and mono to 5 0 formats and a left to right or right to left path selected Place The Place slider adjusts the front back placement of the panning field At 0 the panning field is centered front back At 100 it is placed all the way front At 100 it is placed all the way back The Place slider is only available with mono to quad and mono to 5 0 formats and a left to right or right to left pa
269. ides several different detec tion options for determining how the compressor responds to the input signal To change the detection mode for the compressor Click a detection mode from the options avail able above the Dynamics graph Detection options include the following Smart Select the Smart option for tracks with di verse input signals or if you are simply not sure what detector works best with the given audio ma terial The Smart option analyzes the incoming sig nal and interpolates between the different detection modes as needed This lets you apply a lot of com pression without distortion or pumping RMS Select the RMS option to apply processing ac cording to the detected RMS Root Mean Square amplitude of the input signal The RMS option is similar to the Average option but with a faster re lease time Average Select the Average option to apply pro cessing according to the detected average ampli tude of the input signal Peak Select the Peak option to apply processing according to the detected peak amplitude of the input signal Fast Select the Fast option for tracks with high and short transient signals such as a snare drum track The Fast option is similar to the Peak option but with faster attack and release times However be careful when using the Fast option as it distorts ear lier than the other options Be sure to configure the other compressor settings with this in mind Cursor indicates K
270. ieve with Pro Tools track signal routing and plug in features Blending Eleven Cabinets In this example you ll see how to take the output of one Eleven amp and send it to multiple cabinets so you can blend different cabinets multi mic one cabinet or both To blend multiple cabinets 1 Choose Tracks gt New 2 Configure a new track by doing the following Create one mono Audio Track Click the Add Row button Create three mono Aux Inputs Click Create 3 In the Mix or Edit window configure the audio track by doing the following Click the audio track Input selector and choose your guitar input the audio interface input your guitar is plugged in to Click the Output selector and choose Bus 1 Click the Insert selector and select Eleven 4 Select all three Aux Input tracks by Shift click ing their Track Name displays make sure your audio track isn t still selected This lets you work with the three Aux tracks as one in the next few steps 5 Hold Option Shift Mac or Alt Shift Windows while doing each of the following Choose Bus 1 from the Input selector of any of the three selected Aux Inputs Click the Insert selector of any of the three and select Eleven Click the next available Insert selector on any of three selected Aux Inputs and select the TimeAd juster short plug in 6 Open the Eleven plug in on the audio track and click the Cabi
271. igh end Aural Exciter enhancement is added into the input signal is such a way that the average sig nal level will be virtually unchanged The Level Tune Peaking Null Fill Harmonics Timbre and Mix faders provide separate left and right faders when in stereo For stereo a separate set of switches for independent control of the left and right channels is provided for Ax Solo SPR Bypass Drive and Density Chapter 42 Aphex Aural Exciter Type III 248 The Tune fader adjusts the corner frequency of the high pass filter and the Mix fader varies the amount of Aural Exciter enhancement that is mixed with the unmodified signal Experiment with the Aural Exciter controls to hear how each one enhances the original audio signal To experiment with Aural Exciter 1 Set the Level fader on Max 2 Set the Drive switch to reflect the current nomi nal level 3 Make sure the Bypass switch is deactivated By pass light off 4 Make sure the Ax switch is activated Ax light on 5 Set Density to Normal Density light off As you make the following adjustments alternate the Density switch between Normal and High to hear the change in the Aural Exciter effect 6 Put the Aural Exciter Mix fader on Max making it easier to hear the effect as it changes 7 Vary the Tune fader and listen for the frequency range that is being enhanced The Tune fader can be used to enhance a particular instrument so it stands out in the mix 8
272. il the Envelope Detector has completely released after the audio goes below the specified threshold Increasing the release time re duces the rate at which triggers can occur and de creasing the release time increases the rate at which triggers can occur Manual When the Manual trigger is selected the LFO is triggered manually This can be especially useful if you want to trigger the LFO using Pro Tools automation With control surfaces and automation the Manual trigger acts like an on off switch and triggers the LFO every time it changes state AutoPan Tempo Controls Link To Tempo When the Link To Tempo option is enabled the LFO rate is set to the Pro Tools session tempo and any tempo changes in the session are followed au tomatically In addition the LFO rate slider is ig nored and the Tempo display always shows the cur rent session tempo Duration Selector The Duration selector works in conjunction with the session tempo LFO rate and Beat Clock trig ger By default Duration is set to 1 bar At that set ting the LFO cycles once within one bar When Duration is set to 1 beat the LFO cycles within the duration of one beat When Link to Tempo is en abled the Duration menu allows the LFO rate to be set as a function of the tempo of the Pro Tools ses sion The Duration menu also controls how often the Beat Clock trigger is activated Tempo controls Selecting Duration Chapter 51 AutoPan 302 Tempo Displ
273. ilable input 2 Make sure to use the correct input on your inter face For example on Mbox Pro plug your gui tar into front panel DI Inputs 1 or 2 To connect your guitar to a Pro Tools HD system 1 Make sure you have a pre amp such as an Pro Tools PRE or similar unit connected to a Pro Tools HD audio interface such as a Pro Tools HD I O Note that Pro Tools HD OMNI provides built in pre amps 2 Plug your guitar into an available pre amp input and set its source impedance and other settings as needed for your setup For example if you are using a PRE you can plug your guitar directly into the front panel Line Inst 1 input then set its source to Inst If you use a direct box to convert your gui tar s hi impedance output to a low imped ance signal connect the direct box to a mic or line input instead of the DI input Mbox Pro DI Inputs 1 and 2 Mbox Pro back panel 1 4 inputs are line level only and should not be used with a guitar If you use a direct box to convert your gui tar s hi impedance output to a low imped ance signal set the Line Inst 1 input to Line source or the equivalent on your particular pre amp Guitar into Avid PRE into a 192 I O PRE or other pre amp HD audio interface Chapter 44 Eleven 259 Set Hardware and Levels After plugging in do the following to set your pri mary gain and configure your Pro Tools hardware by watching its input indicators meters Thi
274. imebase selected in the Units pop up menu Processed Displays the target End time and Length of the processed signal Times are dis played in units of the timebase selected in the Units pop up menu You can click the Processed End and Length fields and type values These values update automatically when adjusting the Time control Tempo Displays the Original Tempo and Processed Tempo in beats per minute bpm You can click the Original Tempo and Processed Tempo fields and type values The Processed Tempo value up dates automatically when adjusting the Time con trol Unit Select a timebase for the Original and Processed time fields Bars Beats Min Sec Timecode Feet Frames or Samples Speed Displays the target time compression or ex pansion as a percentage of the original Adjust the Time control or click the Speed field and type a value Time can be changed from 25 00 to 400 00 of the original speed or 4 to 1 4 times the original duration The default setting is 100 00 or no change 25 00 results in 4 times the original duration and 400 00 results in 1 4 of the original duration The Speed field only displays up to 2 decimal places but lets you type in as many decimal places as you want up to the IEEE standard While the display rounds to 2 decimal places the actual time shift is applied based on the number you typed This is especially useful for typing post production pull up and pull down factors see Post Pr
275. imum output level of the audio signal from 30 to 0 dBTP dB relative to full scale measured as a true peak value To adjust the output ceiling do one of the following Click in the Ceiling field either below the Ceil ing control or below the Output meters to type a value between 30 to 0 dBTP Click the Ceiling control and drag up or down Click the Output Ceiling control on the Output meters and drag up or down Output section with Meters Output Sample Peak Hold display Output Ceiling control Output Ceiling Shift click and drag either the Ceiling control or the Threshold control to link both controls and adjust them to match the same value Chapter 20 Pro Limiter 121 Character The Character control adds soft saturation process ing with no additional gain before applying limit ing Release The Release control sets the length of time it takes to cease limiting after the input signal crosses the threshold Release times should be set long enough that if sig nal levels repeatedly rise above the threshold the gain reduction recovers smoothly If the release time is too short the gain can rapidly fluctuate as the limiter repeatedly tries to recover from the gain reduction If the release time is too long a loud sec tion of the audio material could cause gain reduction that continues through soft sections of program material without recovering When the Auto Release option is e
276. in DSP Native and Audio Suite formats Recti Fi supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Recti Fi operates as a mono multi mono or stereo plug in Recti Fi provides additive synthesis effects through waveform rectification Recti Fi multiplies the harmonic content of an audio track and adds subharmonic or superharmonic tones Recti Fi Controls Recti Fi Pre Filter Control The Pre Filter control filters out high frequencies in an audio signal prior to rectification This is de sirable because the rectification process can cause instability in waveform output particularly in the case of high frequency audio signals Filtering out these higher frequencies prior to rectification can improve waveform stability and the quality of the rectification effect If you wish to create classic subharmonic synthesis effects set the Pre Filter and Post Filter controls to a relatively low fre quency such as 250 Hz The range of the Pre Filter is from 43 Hz to 22 kHz with a maximum value of Thru which ef fectively means bypass Recti Fi Normal waveform Chapter 46 Recti Fi 284 Recti Fi Rectification Controls Positive Rectification This rectifies the waveform so that its phase is 100 positive The audible ef fect is a doubling of the audio signal s frequency Negative Rectification This rectifies the wave form so that its phase is 100 negative The audi ble effect is a doubl
277. in the Graphic Display to Adjust Controls 168 ReVibe II Input and Output Meters 169 Contents viii ReVibe II Controls 169 ReVibe II Decay EQ Graph 175 ReVibe II Decay Color Graph 175 ReVibe II Contour Display 176 ReVibe II Room Types 177 Chapter 30 Space 181 Space Feature Highlights 182 Space Overview 183 Impulse Response IR and Space 187 Space Presets
278. include plug ins that come with your Pro Tools system as well as many other plug ins that can be purchased or rented from Avid sep arately This guide documents all 64 bit AAX plug ins available from Avid for Pro Tools 11 Plug In Formats AAX Avid Audio Extension plug ins provide real time plug in processing using host based Native or DSP based HDX systems only pro cessing The AAX plug in format also supports AudioSuite non real time file based rendered pro cessing AAX plug in files use the aaxplugin file suffix There are three plug in formats used in Pro Tools AudioSuite non real time file based processing AAX Native real time host based plug ins AAX DSP real time DSP based plug ins HDX systems only Avid Audio Plug Ins Avid includes a comprehensive set of sound processing effects and utility plug ins with all Pro Tools systems Other Avid plug ins are available for purchase or rental from the Avid store visit shop avid com or in Pro Tools choose Marketplace gt Plug Ins Avid Audio Plug Ins Included with Pro Tools Pro Tools includes a suite of digital signal process ing effects including EQ dynamics delay and other essential audio processing tools The follow ing plug ins are included with Pro Tools 11 EQ Channel Strip see Dynamics EQ III 1 Band 7 Band Dynamics BF76 Compressor Channel Strip Dyna
279. ine once play back is started any changes made to loop or playback markers within the ReWire client application will deselect the Pro Tools Time line selection and remove the loop For information on drawing automation see the Pro Tools Reference Guide Part XII Other Plug Ins Chapter 55 InTune 319 Chapter 55 InTune InTune is a professional instrument tuner plug in that is available in DSP and Native formats It of fers the features and performance of a rack mounted digital tuner in the convenience of a plug in InTune provides accurate and rapid tuning for a wide range of musical instruments saving valuable studio time and adding a level of unprecedented convenience for musicians and audio engineers To use InTune with Pro Tools simply create a new mono audio or Auxiliary Input track in Pro Tools and select InTune from the plug in menu for that track When InTune detects an audio signal from the track the meter lights up and displays the relative pitch of the incoming signal With stringed instru ments this will vary during the attack and decay of the note By default InTune loads the Chromatic tuner pre set This displays all notes in the scale and automat ically displays the required octave InTune provides a number of factory presets for stringed instruments in alternate tunings Each fac tory preset is programmed with the specific notes for each string of the instrument in order to speed the tuning
280. ing of the audio signal s fre quency Alternating Rectification This alternates between rectifying the phase of the first negative waveform excursion to positive then the next positive excur sion to negative and so on throughout the wave form The audible effect is a halving of the audio signal s frequency creating a subharmonic tone Alt Max Rectification This alternates between holding the maximum value of the first positive ex cursion through the negative excursion period switching to rectify the next positive excursion and holding its peak negative value until the next zero crossing The audible effect is a halving of the au dio signal s frequency and creating a subharmonic tone with a hollow square wave like timbre Positive rectification Negative rectification Alternating rectification Alt Max rectification Chapter 46 Recti Fi 285 Recti Fi Gain Control Gain lets you adjust signal level before the audio reaches the Post Filter This is particularly useful for restoring unity gain if you have used the Pre Filter to cut off high frequencies prior to rectifica tion The range of this control is from 18dB to 18dB Recti Fi Post Filter Waveform rectification particularly alternating rectification typically produces a great number of harmonics The Post Filter control lets you remove harmonics above the cutoff frequency and smooth out the sound This is useful for filtering audio that contains
281. input level keeps rising gradually the gain reduction becomes less effective and the amplifier goes back to being a linear amplifier except with the volume turned down This is by design and is based on an understanding of how the human ear behaves The result is that the listener is fooled into thinking that the JOE MEEK compressed sound is louder than it really is but without the strange psychoacoustic effect of deadness that other compressors suffer from Overshoot At fast Attack settings it is possible to make the JOEMEEK overshoot on percussive program material This means that the compression elec tronics are driven hard before the light cells re spond to the increased level The cells catch up and overcompress momentarily giving a tiny dip imme diately following the start of the note To hear it use a drum track set Slope to 5 and At tack and Release to Fast Used sparingly this effect can contribute to musical drive in your tracks Attack and Release Times It may be difficult to understand the interactions between the Attack and Release controls because the JOEMEEK Compressor behaves very differ ently than typical compressors Experimentation is the best option but an explanation may help you understand what s going on The JOEMEEK Compressor uses a compound re lease circuit that reacts quickly to short bursts of volume and less quickly to sustained volume While the unit was being prototyp
282. input signal exceeds the threshold For example a 2 1 compres sion ratio means that a 2 dB increase of level above the threshold produces a 1 db increase in output The compression ratio ranges from 1 0 1 to 20 0 1 Once the Ratio control passes 20 0 1 the Compres sor Limiter effect functions as a limiter rather than a compressor At the limiter setting LMTR for every decibel that the incoming signal goes over the set Threshold 1 dB of gain reduction is applied Once the Ratio control passes the LMTR setting it provides negative ratio settings from 20 0 1 to 0 1 With these settings for every decibel that the in coming signal goes over the set Threshold more than 1 dB of gain reduction is applied according to the negative Ratio setting For example at the set ting of 1 0 1 for each decibel over the set thresh old 2 db of gain reduction is allied Consequently the output signal is both compressed and made softer You can use this as an creative effect or as a kind of ducking effect when used with an external key input Depth The Depth control sets the amount of gain reduc tion that is applied regardless of the input signal For example if the Limiter is set at a Threshold of 20 dB and Depth is set at 0 dB up to 20 dB of gain reduction is applied to the incoming signal at 0 dB If you set Depth to 10 dB no more than 10 dB of gain reduction is applied to the incoming signal Dynamics section Comp
283. ion and the settings in the Click Countoff Options dia log The Note Velocity Duration and Output options in this dialog are for use with MIDI instrument based clicks and do not affect the Click II plug in Click Options dialog See the Pro Tools Reference Guide for more information on configuring Click options Chapter 54 ReWire 310 Chapter 54 ReWire Pro Tools supports ReWire version 2 0 technology developed by Propellerheads Software ReWire is available in Pro Tools using the ReWire Native plug in ReWire provides real time audio and MIDI streaming between applications with sample accu rate synchronization and common transport func tionality Using ReWire Pro Tools can send and receive MIDI to and from a ReWire client application such as a software synthesizer and receive audio back from the ReWire client Pro Tools applies MIDI time stamping to all incoming MIDI Compatible ReWire client applications are auto matically detected by Pro Tools and are available in the Plug Ins Native Insert menus in Pro Tools Selecting a ReWire client application within Pro Tools automatically launches that application if the client application supports this feature Any corresponding MIDI nodes for that application are available in any Instrument track s MIDI Output selector Instrument view and any MIDI track s Output selector Once the outputs of your software synthesizers and samplers are routed to Pro Tools
284. ion provides output metering and controls for adjusting the level of the output signal Output Meters The Output meters show peak signal levels after processing Dark Blue Indicates nominal levels from INF to 12 dB Light Blue Indicates pre clipping levels from 12 dB to 0 dB Red Indicates full scale levels clipping Output Gain The Output Gain control sets the output level after processing For mono instances of Mod Delay III there is a single Gain control For stereo and mono to stereo instances of Mod Delay III there are in dependent Gain controls for each channel left and right Selections for Mod Delay III AudioSuite Processing Because AudioSuite Delay adds additional mate rial the delayed audio to the end of selected audio make a selection that is longer than the original source material to allow the additional delayed au dio to be written to the end of the audio file If you select only the original material without leaving additional space at the end delayed audio that occurs after the end of the selection will be cut off Chapter 32 Moogerfooger Analog Delay 207 Chapter 32 Moogerfooger Analog Delay Moogerfooger Analog Delay is a delay plug in that is available in DSP Native and AudioSuite formats Moogerfooger Analog Delay supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Moogerfooger Analog Delay operates as a mono multi mono or stereo plug
285. is plays the current true peak level Short Term Loudness Meter S Graphically dis plays the current short term output level Integrated Loudness I Graphically displays the current integrated level of the processed signal us ing K weighted metering Adjusting the Loudness Meters View The Loudness meters show a range of 27 LUFS The default view shows from 14 to 41 LUFS Drag up or down in the histogram to adjust the viewed range for loudness metering from 0 LUFS on the upper end down to 50 LUFS on the lower end AudioSuite Processing with Pro Limiter When used as an AudioSuite plug in Pro Limiter does not provide real time analysis data in the his togram or the loudness meter Once you have ren dered the audio selection with Pro Limiter process ing Pro Limiter loudness reporting updates to display the results of the processing K weighted metering implements a filter curve that models the human ear s perception of loudness It is an integral part of the ITU R BS 1770 standard for loudness metering Pro Limiter AudioSuite plug in Chapter 20 Pro Limiter 124 AudioSuite Processing with Pro Limiter Loudness Analyzer Use the Pro Limiter Loudness Analyzer to check loudness levels before rendering Pro Limiter pro cessing if you need to read the current loudness measurements of your program material To analyze audio using the Pro Limiter Loudness Analyzer 1 Make an audio selection in the Edit windo
286. is wired balanced connections The stereo version of Pitch II provides adjacent left and right channel Input Polarity but tons Range The Range selector lets you adjust the range of frequencies Low Mid High Wide used for pitch detection For most program material the Wide setting should work well for pitch detection and transposition If you encounter undesirable fre quency artifacts with pitch transposition experi ment with other settings Set this parameter to match the expected frequency content of source material For example when working with a bass part set Range to Low When pitch shifting audio from material similar to a soprano vocalist or a vi olin set Range to High Clip Indicator The Clip indicator shows whether clipping has occurred on output It is a clip hold in dicator If clipping occurs at any time the clip light will remain on To clear the Clip indicator click it Long delay times and high feedback times increase the likelihood of clipping Pitch II plug in mono Input and Transient controls Chapter 23 Pitch II 135 Level Indicator The Level indicator shows the pres ence of an input signal Threshold Pitch II detects and responds to tran sients in the incoming audio signal to prevent smearing of sharp attacks such as drum hits or vocal plosives The Threshold control 40 dB to 0 dB and Off determines how strong a transient needs to be in order to be recognized by the pitch det
287. isplay that show the current values Low Frequency Control The Lo Freq control sets the frequency boundary between low and mid cut or boost points in the re verb EQ The range of this control is from 50 0 Hz to 1 5 kHz Low Gain Control The Lo Gain control sets cut and boost values for the low and mid frequencies of the reverb decay EQ The range of this control is from 24 0 dB to 12 0 dB High Frequency Control The Hi Freq control sets the frequency boundary between mid and high cut or boost points in the re verb EQ The range of this control is from 1 5 kHz to 20 0 kHz High Gain Control The Hi Gain control sets cut and boost values for the mid and high frequencies of the reverb decay EQ The range of this control is from 24 0 dB to 12 0 dB High Frequency Rear Cut Control The Rear control rolls off additional high frequen cies in the rear channels of the early reflections and reverb tail The application of this filter is distinct from the application of Decay Color and Decay EQ The range of this control is from 250 0 Hz to 20 0 kHz ReVibe II Decay Color Graph The Color display lets you adjust the Decay Color settings for ReVibe II Click the Color button to toggle the display to show Decay Color settings To adjust a setting on the graph drag the correspond ing control point You can use the controls in the Decay Color graph to shape the tonal spectrum of the reverb by adjust ing the decay times
288. it is not highlighted Selecting the Source for side chain processing Side Chain Listen mode enabled Attenuation Listen and Side Chain Listen can be enabled simultaneously in which case Attenuation Listen is audible but Side Chain Listen is not Chapter 18 Pro Compressor 106 Side Chain Filter On Off You can use the side chain input with or without filtering by enabling or disabling the Side Chain Filter On Off button To enable or disable filtering on the Side Chain Click the Side Chain Filter On Off button on the right side of the Side Chain section so that it is highlighted To disable it click the button again so that it is not highlighted Side Chain Filter The side chain filter applies only to the side chain signal feeding the Pro Compressor detection algo rithm Compression is triggered only when the sig nal exceeds the Threshold setting at the frequencies passing through the side chain filter Filter Frequency The Freq control lets you set the center frequency for the selected Filter Type from 20 Hz to 21 0 kHz Filter Q When the Filter Type is set to Band Pass or Notch the Q control is available The Q control changes the width of the filter around the center frequency band Higher Q values represent narrower band widths Lower Q values represent wider band widths Filter Type Four Filter Type options are available for side chain processing Low Pass Select the Low Pass option to
289. itching 317 client applications 310 looping playback 317 meter changes 316 MIDI automation 314 MIDI continuous controller CC data 314 316 MIDI Output selector 314 quitting client applications 316 recording MIDI over 315 requirements 312 signal flow for audio and MIDI 311 software synthesizer 310 tempo sync 316 track count 312 using with Pro Tools 313 Audio Plug Ins Guide 358 S SansAmp PSA 1 plug in 289 amp simulation 289 buzz 290 cabinet simulation 289 crunch 290 distortion 290 equalization 289 harmonic generation 289 lo fi textures 289 punch 290 tube sounds 289 unity 290 wah 290 Sci Fi plug in 232 Effect Amount control 233 Effect Frequency control 233 Envelope Follower 233 Freak Mod 233 Input Level 232 LFO Low Frequency Oscillator 233 Mod Amount Mod Rate control 234 Mod Slewing control 233 Modulation Type control 233 Output Meter 234 Resonator 233 Resonator 233 Ring Mod control 232 ring modulation 232 Sample Hold control 233 Slewing 233 triangle wave 233 Trigger and Hold 234 side chain processing 41 43 53 70 84 87 104 114 Signal Generator plug in 333 Frequency control 333 Level control 333 pink noise 333 Signal control 333 white noise 333 Smack plug in 126 adjusting input 127 Attack control 128 band emphasis 129 clip indicator 130 131 compression modes 127 Distortion control 130 hard limiting ratio setting 128 high pass detector 129 high pass filter 130 Input meter 131 key input 131
290. ive DSP or AudioSuite version of this plug in any settings for this parameter will be active Note that this control does not affect the first delayed signal only the repeated de lays caused by the Feedback control Note that this control does not affect the first delayed signal only the repeated de lays caused by the Feedback control Settings for this parameter are saved with plug in presets If you use a preset for the Native DSP or AudioSuite version of this plug in any settings for this parameter will be active Chapter 33 Reel Tape Delay 212 Synchronizing Reel Tape Delay to Session Tempo You can set the delay time Speed control in the Reel Tape Delay to synchronize to the session tempo in beats per minute To synchronize the delay time to the session tempo 1 In the BPM Sync section click the On button The Tempo Rate display changes to match the current session tempo 2 To set a rhythmic delay click the Note Value to choose from the available note values whole half quarter eighth sixteenth or thirty second note 3 To adjust the rhythm further do any of the following To enable triplet rhythm delay timing click the Triplet 3 button so that it is lit To set a dotted rhythm delay value click the Dot button so that it is lit Reel Tape Delay Presets The Reel Tape Delay presets coordinate Speed Wow Flutter Feedback and the Bass and Treble controls for di
291. k Macintosh Return key board focus to Pro Tools Escape key Chapter 30 Space 196 Space IR Browser Edit Menu The IR browser s Edit menu contains the following commands Download Space IR Package Opens a Web browser to the Space online IR library Install Space IR Package Installs a new IR pack age downloaded from the Space online library see Installing Space IR Packages on page 196 Import Other IR Folder Lets you import a new IR folder in common file formats By default the new IR is given the same name as the selected folder Remove Imported IR Folder Lets you remove the currently selected IR folder Rename Imported IR Folder Lets you rename the currently selected IR folder Add to Favorites Adds the currently selected IR to the Favorites group at the top of the browser win dow New Folder in Favorites Creates a folder in the Fa vorites group Favorite IRs can be dragged and dropped into the folder Rename Favorites Folder Lets you rename the currently selected Favorites folder Remove from Favorites Removes the currently se lected IR from the Favorites group This function only removes the link in the Favorites group and does not remove the original IR file from the sys tem Reset to Default IR Library Resets Space to the de fault library This also removes any user imported IR folder but does not affect the Favorites folder or IR packages installed from the Space online IR library
292. ke Norm mode Warm mode lets you precisely adjust the Ratio Attack and Release controls to fine tune the compression characteristics Opto Mode Button Enable the Opto button to emulate opto electro compressors Opto mode produces soft knee compression with gentle attack and release charac teristics and is ideal for compressing thin vocals bass guitars kick drums and snare drums In Opto mode only the Input and Output controls are avail able for adjusting the amount of compression The Attack Release and Ratio controls are greyed out and cannot be manually adjusted Smack Input Control In all Smack compression modes Input adjusts the level of input gain to the compressor For more compression increase the amount of input gain For less compression reduce the amount of input gain Norm Warm and Opto mode buttons Some sustained low frequency tones can cause waveform distortion in Norm mode The release characteristics of Warm mode which is based on Norm mode can be used to remedy this distortion by reducing wave form modulation Setting the Input and Output controls to 5 is equal to unity gain at a compression ratio of 1 1 Chapter 22 Smack 128 Smack Attack Control In Norm and Warm modes Attack controls the rate at which gain is reduced after the input signal crosses the threshold Set this control to 0 for the fastest attack time or to 10 for the slowest attack time Depending on the pro
293. ke compression and limiting which modify the dynamics of audio material normalization pre serves dynamics by uniformly increasing or de creasing amplitude Level Specifies how close to maximum level clipping threshold the peak level of a selection is boosted Set this value by adjusting the Max Peak At slider by entering a numeric decibel value below the clip ping threshold or by entering a percentage of the clipping threshold Invert To prevent clipping during sample rate conversion Normalize in Peak mode to no greater than the range between 2 dB to 0 5 dB Optimum settings will vary with your program material and your Conversion Quality setting in the Editing tab of the Pref erences dialog Observe caution when nor malizing in RMS mode as that mode of analysis does not account for instantaneous peaks Normalize Chapter 61 Other AudioSuite Plug In Utilities 349 Channel Mode When processing a selection that spans across mul tiple channels or tracks the Normalize plug in has two modes of operation Mono Normalizes each channel independently Multi Input Normalizes program material across all selected channels together so that the channels are processed relative to each other Peak RMS Toggle Switches the calibration of normalizing between Peak or RMS modes Peak Normalizes the input signal at the maximum possible level without clipping RMS Normalizes the input signal at a level consis
294. l Surface Access Only The Wow Speed parameter adjusts the frequency of the tape machine s wow effect or the rate of fluctuation in tape speed A higher value results in faster fluctuations in speed A lower value results in slower fluctuations in speed Wow Speed is ad justable from 0 to 100 percent with a default value of 50 percent This parameter is accessible only from the plug in automation playlist or from a supported control surface Bass The Bass control boosts or cuts the amount of low frequencies fed to the echo feedback loop Bass amount is adjustable from 10 dB to 10 dB with a default value of 0 dB Treble The Treble control boosts or cuts the amount of high mid frequencies fed to the echo feedback loop Treble amount is adjustable from 10 dB to 10 dB with a default value of 0 dB Mix The Mix control adjusts the amount of processed signal mixed with the input signal in the final out put of the plug in The default Mix value is 25 per cent Noise Plug In Automation Playlist or Control Surface Access Only The Noise parameter controls the level of simu lated tape hiss that is added to the processed signal Noise is adjustable from Off INF to 24 dB with a default value of 80 dB This parameter is accessible only from the plug in automation playlist or from a supported control surface Settings for this parameter are saved with plug in presets If you use a preset for the Nat
295. l Tape Delay simulates an analog tape echo effect modeling the frequency response noise wow and flutter and distortion characteristics of analog tape It also re produces the varispeed effect you get when the tape speed control is adjusted Reel Tape Delay is available in DSP Native and AudioSuite formats Reel Tape Delay supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Reel Tape Delay operates as a mono multi mono or stereo plug in For years engineers have relied on analog tape to add a smooth warm sound to their recordings When driven hard tape responds with gentle dis tortion rather than abrupt clipping as in the digital domain Magnetic tape also has a frequency depen dent saturation characteristic that can lend punch to the low end and sweetness to the highs Reel Tape Delay models a studio tape machine in record playback mode with a fixed distance be tween the record head and the play head and a con tinuously variable tape speed Reel Tape Delay automatically applies tape satura tion effects that correspond to the following control settings in Reel Tape Saturation Speed 15 ips Bias 0 0 dB Cal Adjust 9 dB You can use the BPM Sync feature to synchronize the Reel Tape Delay effect to the current tempo of the Pro Tools session Reel Tape Delay can be placed on mono stereo or multichannel tracks Reel Tape Delay Chapter 33 Reel Tape Delay 2
296. l sound Keeping this shape constant is critical to formant correct pitch shifting and achieving a natural sounding result Time Shift Transient Controls The Transient section is only available when Poly phonic or Rhythmic is selected as the Audio Type and provides slightly different controls for each When Polyphonic is selected as the Audio Type the Transient section provides controls for setting the transient detection threshold and for adjusting the analysis window length for processing audio When Rhythmic is selected as the Audio Type the Transient section provides controls for setting the transient detection threshold and for adjusting the decay rate of the transients in the processed audio when time stretching Follow The follow button enables an envelope fol lower that simulates the original acoustics of the audio being stretched Click the Follow button to enable or disable envelope following Follow is only available when Polyphonic is selected as the Audio Type Time Shift Formant section Time Shift Transient section with Polyphonic selected as the Audio Type Time Shift Transient section with Rhythmic selected as the Audio Type Chapter 24 Time Shift 141 Threshold The Threshold controls sets the tran sient detection threshold from 0 0 dB to 40 0 dB Disable transient detection by setting the Threshold control to Off turn the knob all the way to the right Part of Time Shift s processing relies upon
297. l to trigger expansion or gat ing Signals that fall below the threshold will be re duced in gain Signals that are above it will be un affected The Dynamics Graph display shows the threshold as a vertical line Ratio The Ratio control sets the amount of expansion For example if this is set to 2 1 it will lower signals be low the threshold by one half At higher ratio lev els Pro Expander functions like a gate by cutting off signals that fall below the threshold As you ad just the ratio control refer to the Dynamics Graph display to see how the shape of the expansion curve changes Upward The Upward button enables Upward Expansion mode When Upward Expansion mode is enabled Pro Expander amplifies signals above the Thresh old When it is disabled the signal is attenuated when the signal falls below threshold To enable or disable Upward Expansion mode Click the Upward button so that it is highlighted To disable it click the button again so that it is not highlighted Knee The Knee control sets the rate at which the proces sor reaches full expansion or gating effect once the threshold has been exceeded Attenuation Listen mode enabled Chapter 19 Pro Expander 113 Attack The Attack control sets the attack time or the rate at which gain is reduced after the input signal crosses the threshold Use this along with the Ratio setting to control how soft Pro Expander s gain reduction curve is
298. ld and reaches full com pression after exceeding the threshold This cre ates smoother compression Attack The Attack control sets the attack time or the rate at which gain is reduced after the input signal level crosses the threshold The smaller the value the faster the attack The faster the attack the more rapidly the compressor applies attenuation to the signal If you use fast at tack times you should generally use a proportion ally longer release time particularly with material that contains many peaks in close proximity Release The Release control sets the length of time it takes for compression to be fully deactivated after the in put signal drops below the threshold Release times should be set long enough that if sig nal levels repeatedly rise above the threshold the gain reduction recovers smoothly If the release time is too short the gain can rapidly fluctuate as the compressor repeatedly tries to recover from the gain reduction If the release time is too long a loud section of the audio material could cause gain re duction that continues through soft sections of pro gram material without recovering Compressor Ratio set to a negative value The actual compression attack time is also dependent on the selected Detection mode Each mode has its own attack and release times that are calculated in advance of com pression processing If a slower Detection mode is selected such as AVG the fast
299. lect Eleven 4 Record enable the audio track or enable its TrackInput monitoring button Pro Tools HD software only Input 1 Gain on Mbox Pro One audio track for input calibration on Pro Tools Eleven Guitar input Track meter Chapter 44 Eleven 260 Set Up Eleven Use Eleven s Input LED to make final gain adjust ments and complete the input calibration process To calibrate your input signal to the Eleven plug in 1 Open the Eleven plug in window by clicking its insert slot Leave it at its default settings 2 Strum as hard as you can a few more times and watch Eleven s Input LED to see where your level registers The Input LED lights green yel low orange or red to indicate the following level ranges Green Off to 8 Indicates signal is present but too low Yellow 8 to 4 Indicates the best level for low output sources such as single coil pickups Orange 4 to 0 Indicates the best level for higher output sources such as humbucker pickups Red 0 and above Indicates that you have clipped the plug in input Click the Input LED to clear the clip indicator 3 Leaving the Input control on the plug in at its default setting of 0 12 00 position set the sig nal level going to the plug in by adjusting the in put gain control on your hardware until Eleven s Input LED shows yellow or orange 4 After calibrating strum as you normally would and or back down your guitar volume from t
300. level 126 limiting 128 Meter Mode button 130 Norm mode 127 Opto mode 127 output gain 128 Output meter 131 Ratio control 128 reducing waveform distortion 127 Release control 128 Side Chain EQ control 129 side chain frequency filters 129 side chain processing 131 threshold and ratio 128 unity gain 127 VU meter 130 SoundReplacer plug in 335 Crossfade control 338 Dynamics controls 339 Load Unload Sound icons 337 MIDI triggered samplers 336 Mix control 338 Online Help 339 Peak Align control 338 Trigger envelope 336 Trigger markers 336 Trigger Threshold 337 waveform display 336 Zoomer 336 338 Space plug in 181 clip indicator 194 Control group selector 198 convolution reverb 184 Decay controls 201 Delay controls 198 Early Reflection controls 199 EQ controls 200 impulse computer 185 impulse response IR 187 installing IR packages 196 IR browser 195 196 IR channel compatibility 189 IR channel formats 188 IR filename conventions 188 IR formats 187 IR library 202 IR Name 192 Level controls 198 Audio Plug Ins Guide 359 Meters display 194 Online IR Library 196 Picture Preview mode 193 Preferences mode 194 presets 190 201 primary controls 197 Quick browser 192 Snapshot menu 192 Snapshot mode 193 snapshots 190 201 system design 185 Waveform mode 192 T TCE Trim tool 142 Tel Ray Variable Delay plug in 213 Electronic Memory Unit 213 tuna can 213 Time Compression Expansion plug in 343 Accuracy control 344 Crossfade
301. leven When multi tracking guitar experienced engineers know how to identify and take advantage of the phase relationships that exist between different sig nals Adjusting phase is not just a corrective tech nique either it s also a powerful creative technique for tone as well as for special effects You can use the TimeAdjuster plug in to flip phase and to adjust timing in single sample increments as described in the next sections Flipping Phase Polarity Electric guitar is often recorded to more than one track such as one dry or DI track plus one or more tracks of a mic d amp The different signal paths of direct tracks versus mic tracks affect the timing re lationships of the audio Depending on the signal chain of each track the signals can get so out of alignment that they nearly cancel each other out Sending a single source track through multiple unique amps can pose an additional challenge in that each tube stage in an amp usually inverts the signal So depending on whether the number of tube stages in an amp is odd or even that amp will either be inverting or non inverting respectively If you send an identical signal to two amps and one is inverting while the other is non inverting signal cancellation will result All amps in Eleven accu rately model the number of amp stages found in all the original hardware If you want to keep it simple and be able to experi ment with phase flip do the following
302. lick Choose Chapter 30 Space 188 Space Multichannel IR Formats Space supports IRs in multichannel or multiple mono audio files IRs with a single input are used for mono or summed stereo processing and can be stored as a single interleaved multichannel file or as multi mono files IRs with stereo inputs used for true stereo processing must be stored as multi mono files The following table shows Space IR channel for mats For multi mono files Space understands the fol lowing filename conventions based on those used by Pro Tools The filename format is based on the impulse name plus two suffixes which indicate in put and output channels as follows Impulsename inputchannel outputchannel type Impulsename is the name of the impulse Mixing multiple IR files with the same Impulsename in the same folder is not supported Inputchannel refers to the number of sources used for the impulse starting at the number 1 An IR captured in true stereo will usually have two input channels numbered 1 and 2 If there is only one input channel then inputchannel is optional and can be omitted Also instead of using num bers 1 and 2 the inputchannel can be designated as L and R Outputchannel refers to the microphones used to capture the impulse and corresponds to your stu dio monitors outputchannel is designated using the standard L C R Ls and Rs extensions Type is optionally WAV or AIFF For best per fo
303. ller CC capable MIDI device Support for sample rates of 44 1 kHz 48 kHz 88 2 kHz and 96 kHz Support for mono or multi mono operation in up to 8 channel 7 1 format Eleven Plug In Features Classic amp models that faithfully recreate the sound and dynamic response of the original amps Highly accurate speaker cabinet models with variable speaker breakup cone distortion Selectable mics with on and off axis options Amps cabs and mics can be mixed and matched into nearly limitless combinations Amps and cabs can be bypassed separately All controls can be automated Noise Gate to control any unwanted noise Settings files presets to store and recall factory and custom tones Support of any compatible Ethernet or MIDI controller MIDI Learn provides effortless map ping to any continuous controller CC capable MIDI device Support for sample rates of 44 1 kHz 48 kHz 88 2 kHz and 96 kHz Support for mono or multi mono operation in up to 8 channel 7 1 format Eleven can share preset data with the Eleven Rack guitar processor audio interface from Avid For more information see the Eleven Rack User Guide Chapter 44 Eleven 257 Eleven Input Calibration and QuickStart This section shows you how to get connected calibrated and cranking through Eleven as quickly as possible Before You Begin with Eleven Eleven was designed to model the ess
304. log Delay 207 Moogerfooger Analog Delay Controls 208 Moogerfooger Analog Delay Tips and Tricks 208 Chapter 33 Reel Tape Delay 209 Reel Tape Common Controls 210 Reel Tape Delay Controls 210 Synchronizing Reel Tape Delay to Session Tempo 212 Reel Tape Delay Presets 212 Chapter 34 Tel Ray Variable Delay 213 Tel Ray Controls 214 Tel Ray Tips and Tricks 214 Contents ix Chapter 35 TimeAdjuster 215
305. lue is 100 no pitch shift Time Shift Pitch section Chapter 24 Time Shift 142 AudioSuite Input Modes and Time Shift Time Shift supports the Pro Tools AudioSuite In put Mode selector for use on mono or multi input processing Mono Mode Processes each audio clip as a mono file with no phase coherency maintained with any other simultaneously selected clips Multi Input Mode Processes up to 48 input chan nels and maintains phase coherency within those selected channels AudioSuite Preview and Time Shift Time Shift supports Pro Tools AudioSuite Preview and Bypass For more information on using Audio Suite Preview and Bypass see the Pro Tools Refer ence Guide Time Shift as AudioSuite TCE Plug In Preference The Time Shift plug in s high quality time com pression and expansion algorithms that can be used with the Pro Tools TCE Trim tool To select Time Shift for use with the TCE Trim tool 1 Choose Setup gt Preferences 2 Click the Processing tab 3 From the TC E Plug In pop up menu select Time Shift 4 Select a preset from the Default Settings pop up menu 5 Click OK Processing Audio Using Time Shift Time Shift lets you change the time and pitch of se lected audio independently or concurrently TCE Plug In option in Processing Preferences page See the Pro Tools Reference Guide for more information about the TCE Trim tool Normalizing a selection before using Time Shift may produce
306. ly Authorizing Avid Audio Plug Ins When you purchase or rent an Avid Audio plug in you receive an activation code either on an activa tion card or through your Avid account To authorize your plug in follow the steps below or visit www avid com activationcard and follow the online instructions To authorize Avid Audio plug ins 1 If you don t already have an iLok account visit www ilok com to sign up for an account 2 Visit www avid com activation and log into your Avid account if you don t already have an Avid account click Create Your Account 3 Enter your activation code and your iLok com User ID 4 Follow the on screen instructions to deposit your license into your iLok com account 5 Once the activation process is complete the download links for your Avid audio plug in will be available in the My Products section of your Avid account 6 Download and install the plug in that you purchased 7 Make sure your iLok is connected to an avail able USB port on your computer iLok USB key 2nd generation For more information visit the iLok website www iLok com Chapter 2 Installing and Authorizing Avid Paid Plug Ins 8 8 Launch Pro Tools and follow the on screen in structions to transfer the plug in license to your iLok and authorize the plug in Installing Plug Ins for Pro Tools Installing Paid Plug Ins on Mac To install a plug in on Mac 1 Download the installer for Mac from
307. lying compression Signals that exceed the Threshold will be com pressed by the amount of gain reduction set with the Ratio control Signals that are below the Threshold will be unaffected The range of the Threshold control is from 70 dB to 0 dB A set ting of 0 dB is equivalent to no compression Greater than stereo formats are only available with Pro Tools HD Impact Chapter 15 Impact 86 Impact Release Control Release controls the length of time it takes for the compressor to be fully deactivated after the input signal drops below the threshold level In general this setting should be longer than the attack time and long enough that if signal levels repeatedly rise above the threshold they cause gain reduction only once If the release time is too long a loud segment of audio material could cause gain reduction to per sist through a low volume segment if one fol lows Setting this control to its maximum value Auto selects a release time that is program depen dent based on the audio being processed The range of this control is from 20 milliseconds to 2 5 seconds Impact Make up Control Make Up adjusts the overall output gain Because large amounts of compression can restrict dynamic range the Make Up control is useful for compen sating for heavily compressed signals and making up the resulting difference in level When Impact is used on stereo or multichannel tracks the Make Up control determines mas
308. marks the highest nor mal compression mode before the onset of nega tive compression values from 20 0 1 to 0 1 At the LMTR setting for every decibel that the in coming signal goes over the set Threshold 1 dB of gain reduction is applied Attenuation Listen mode enabled Compressor Ratio set to LMTR Chapter 18 Pro Compressor 103 Once the Ratio control passes the LMTR setting it provides negative ratio settings from 20 0 1 to 0 1 With these settings for every decibel that the in coming signal goes over the set Threshold more than 1 dB of gain reduction is applied according to the negative Ratio setting For example at the set ting of 1 0 1 for each decibel over the set thresh old 2 dB of gain reduction is applied Conse quently the output signal is both compressed and made softer You can use this as a creative effect or as a kind of ducking effect when used with an ex ternal key input Knee The Knee control sets the rate at which the com pressor reaches full compression once the threshold has been exceeded As you increase this control it goes from applying hard knee compression to soft knee compres sion With hard knee compression compression be gins when the input signal exceeds the threshold This can sound abrupt and is ideal for limiting With soft knee compression gentle compression begins and increases gradually as the input signal approaches the thresho
309. ments recorded with multiple microphones or stereo pairs If you are working with mono signals and the accumulated delays are small just a few samples for example you prob ably needn t worry about delay compensation Determining the DSP delay of track inserts Mix window shown For more information about delays and mixing with Pro Tools see the Pro Tools Reference Guide Part VII Modulation Plug Ins Chapter 36 Moogerfooger Lowpass Filter 219 Chapter 36 Moogerfooger Lowpass Filter Moogerfooger Lowpass Filter features a 2 pole 4 pole variable resonance filter with envelope fol lower Use it to achieve classic 60s and 70s sounds on bass and electric guitar or just dial in some warm fat virtual analog resonance when you need it Moogerfooger Lowpass Filter is available in DSP Native and AudioSuite formats Moogerfooger Lowpass Filter supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Moogerfooger Lowpass Filter operates as a mono multi mono or stereo plug in With the invention of the MOOG synthesizer in the 1960s Bob Moog started the electronic music revolution A direct descendant of the original MOOG Modular synthesizers the Moogerfooger Lowpass Filter provides two classic MOOG mod ules a Lowpass Filter and an Envelope Follower A low pass Filter allows all frequencies up to a cer tain frequency to pass and cuts frequencies above the cutoff frequency It removes
310. mics III Compressor Limiter Expander Gate De Esser Maxim Additional plug ins are available from third party developers For more information visit www avid com plugins Chapter 1 Audio Plug Ins Overview 3 Pitch and Time Shift Pitch II Time Shift Vari Fi Reverb D Verb Delay Mod Delay III TimeAdjuster Modulation Sci Fi Harmonic Eleven Free Lo Fi Recti Fi SansAmp PSA 1 Dither Dither POW r Dither Sound Field AutoPan Down Mixer Instrument Click II ReWire Other DC Offset Removal AudioSuite only Duplicate AudioSuite only Gain AudioSuite only Invert AudioSuite only Normalize AudioSuite only Reverse AudioSuite only Signal Generator Time Compression Expansion InTune MasterMeter Metro Trim Additional Avid Audio Plug Ins The following plug ins are available separately for purchase and rental Aphex Aural Exciter Type III Aphex Big Bottom Pro BF 2A BF 3A Eleven guitar amplifier modeling plug in Fairchild 660 and 670 Focusrite d2 d3 Impact JOEMEEK SC2 Compressor JOEMEEK VC5 Meequalizer Moogerfooger plug ins Moogerfooger Analog Delay Moogerfooger Ring Modulator Moogerfooge
311. monics It actually adds har monics restoring the sound s natural brightness clarity and presence effectively improving detail and intelligibility Use Aural Exciter on specific in struments or in the final mix to bring life back to re cordings Unlike EQs and other brightness enhancers which only boost the high frequencies that often alter the overall tonal balance Aural Exciter extends the high frequencies The stereo image is enhanced with Aural Exciter This results in a greater per ceived loudness without an introduction of noise into the audio path commonly caused by increased gain Aphex Aural Exciter Type III Chapter 42 Aphex Aural Exciter Type III 242 Aural Exciter is a single ended process which can be inserted at any point within the audio chain The input signal is split into two paths One path goes to the output unmodified while the other path known as a side chain goes through the Aural Exciter which includes a tunable high pass filter and a har monics generator Aural Exciter applies frequency dependent phase shift and amplitude dependent harmonics The output of the Aural Exciter s har monic generator is mixed back with the unmodified signal lower in level When used at nominal settings Aural Exciter does not add significant average level to the original sig nal Even though the added information is low level the perception is a dramatic increase in mid and high frequencies The Aural Excite
312. mono multi mono or stereo plug in Like the Lowpass Filter the Moogerfooger Ring Modulator has its roots in the original MOOG Modular synthesizers It provides three classic MOOG modules a Low Frequency Oscillator a Carrier Oscillator and a Ring Modulator Low Frequency Oscillators or LFOs create slow modulations like vibrato and tremolo The LFO in the Moogerfooger Ring Modulator is a wide range dual waveform sine square oscillator The Carrier Oscillator is a wide range sinusoidal oscillator It s called the Carrier Oscillator because like the carrier of an AM radio signal it s always there ready to be modulated by the input A Ring Modulator takes two inputs and outputs the sum and difference frequencies of the two inputs For example if the first input contains a 500 Hz sine wave and the second input contains a 100 Hz sine wave then the output contains a 600 Hz sine wave 500 plus 100 and a 400 Hz 500 minus 100 sine wave Moogerfooger Ring Modulator Chapter 38 Moogerfooger Ring Modulator 226 Moogerfooger Ring Modulator Controls LFO Section Control the LFO using the Amount and Rate knobs and the Square Sine waveform selector switch Amount Amount determines the amount of LFO waveform that modulates the frequency of the car rier oscillator When the knob is full counterclock wise the carrier is unmodulated Fully clockwise the carrier oscillator is modulated over a range of three octaves
313. mplifier which imparts further char acter to the tone In fact it s common to see engi neers using the LA 2A simply as a line amp with out any compression applied to the signal One beautiful side effect of the LA 2A s elegant design is that it s easy to hear the compression ac tion When the BF 2A s two knobs are set properly you know you got it right BF 2A Chapter 8 BF 2A 38 BF 2A Controls The Peak Reduction and Gain controls combine with the Comp Limit switch to determine the amount and sound of the compression The follow ing controls and meters are provided Gain Gain provides makeup gain to bring the signal back after passing through peak reduction Peak Reduction Peak Reduction controls the amount of signal entering the side chain which in turn affects the amount of compression and the threshold The more Peak Reduction you dial in the more squashed the sound Too little peak re duction and you will not hear any compression ac tion too much and the sound becomes muffled and dead sounding Comp Lim The Comp Limit switch affects the compression ratio The common setting for audio production is Comp which provides a maximum compression ratio of approximately 3 1 In Limit mode the unit behaves more like a broadcast lim iter with a higher threshold and compression ratio of approximately 12 1 Meter Both Gain Reduction and Output metering are provided The Meter knob operates as follows
314. mps on page 274 Make sure each Aux Input has an Eleven plug in followed by a TimeAd juster short plug in 2 Open the plug in window for each of the Time Adjuster plug ins click the first one to open it then Shift click each of the other TimeAdjuster plug ins 3 Adjust the Delay slider in one sample incre ments Listen to the effect it has on the combined signal Repeat increasing the Delay by one sam ple each time 4 Try combinations of TimeAdjuster settings with flipped and non flipped Phase settings for end lessly variable tonal possibilities Eleven Tips and Suggestions This section leaves you with some tips and sugges tions for other ways you can integrate Eleven into your sessions Changing Settings Versus Switching Amps Many guitarists use different tones to maximize the contrast between sections of a song intro verse chorus or bridge Some examples include Soft or clean tone for the verse kick in the dis tortion for the chorus Using tremolo during the intro and the bridge Doubling the rhythm track halfway through the verse to build momentum Pro Tools automaton is the key to these and other techniques For simple single amp contrasts such as soft loud choose an amp and automate its gain drive volume or other parameter to achieve the de sired tone change This uses the least amount of processing resources of the examples provided here To switch amps autom
315. n hancement you generate If you cannot get the Drive meter to register in the yellow area try set ting the Drive switch to High Drive switch en abled Out Meter The Out meter lets you monitor the output level af ter Aural Exciter processing Rotary Controls Level Control The Level control sets the attenuation of the input signal For normal operation set the Level control on Max no attenuation Aural Exciter has an internal gain structure that boosts 6 dB of the output from the high pass filter into the side chain The Drive switch further boosts the signal level fed into the harmonics generator When Drive is set to Normal you obtain a boost of 6 dB in the High position you can get an addi tional 6 dB of gain for a total maximum boost of 18 dB You can also generate a boost in the high pass filter section by setting Peaking to Max If you run into headroom problems when adjusting the Mix control adjust the Level control to gener ate the necessary headroom Tune Control The Tune control sets the bandwidth corner fre quency of the second order high pass filter in the side chain prior to the harmonics generator The range of the control extends from 700 Hz to 7 kHz The following figure shows the range of the Tune control from 700 Hz to 7 kHz with Null Fill set near Min and Peaking set at the mid point position Notice the interaction that the Peaking and Null Fill controls have on Tune as well as on
316. n 286 Bias control 287 Cal Adjust control 287 Noise control 287 presets 288 removing plug ins 8 reverb D Verb 155 Reverb One 158 ReVibe II 167 Space 181 reverb about 183 184 Reverb One plug in 158 100 Wet control 160 acoustic environments 159 anechoic chamber 159 Attack control 161 Band Breakpoints 164 Chorus controls 161 clipping indicator 166 Crossover sliders 165 Decay Ratio control 161 Delay Master control 163 Depth control 161 Diffusion control 162 Dynamics controls 160 early reflections 159 Early Reflect On 163 Early Reflection control 162 ER early reflection button 165 ER Settings control 163 presets 163 simulating 162 Frequency control 164 HF Cut control 164 HF Damp control 165 input level meters 166 late reverberation 159 Level control 161 163 Master Mix controls 160 meters clipping indicator 166 input 166 output 166 Output Level meters 166 Pre delay 159 presets 162 Rate control 161 RC reverb contour button 165 reverb character 159 Audio Plug Ins Guide 357 reverb graphs editing 164 Reverb Color 165 Reverb Contour 165 Reverb EQ 164 simulating early reflections 162 Size control 162 Spread control 162 163 Stereo Width control 160 Threshold control 161 Time control 161 tool tips 166 Wet Dry control 160 Reverse plug in 349 ReVibe II plug in 167 100 Dry Mix button 174 100 Wet Mix button 174 Attack Shape control 171 Attack Time control 171 bipolar controls 9 Center control 173 Chorus On
317. n Out switch 252 Level control 252 Link switch 253 Mix control 252 optimizing 254 Out meter 251 Phase switch 253 Solo switch 253 Tune control 252 block diagram 242 251 C Channel Strip plug in 44 bypassing effects modules 49 Compressor Limiter 51 52 Dynamics Graph display 50 Dynamics section 49 enabling or disabling effects 46 Expander Gate 51 FX Chain 49 Gain Reduction meters 47 Input meters 47 Input Trim 47 Listen button 46 Phase Invert 47 Side Chain Detection options 54 Filter Frequency control 54 Filter Type options 54 Side Chain Processing Graph display 54 Source selector 53 Click plug in 307 Compressor Limiter Dynamics III 62 D D2 27 D3 78 DC Offset Removal plug in 346 Dither plug in 292 294 bit resolution 292 Noise Shaping 293 Down Mixer plug in 304 Duplicate plug in 347 flattening a track 347 D Verb plug in 155 Algorithm control 156 Church algorithm 156 clipping indicator 155 Diffusion control 157 Hall algorithm 156 Hi Frequency Cut control 157 input level meters 155 Low Pass Filter control 157 output level meters 155 Size control 156 dynamics BF 2A 37 BF 3A 40 BF76 42 Channel Strip 44 Dynamics III 59 Fairchild 660 75 Fairchild 670 77 Focusrite D3 78 Impact 85 JOEMEEK SC2 Compressor 89 Maxim 91 Pro Compressor 97 Pro Expander 107 Pro Limiter 117 Purple Audio MC77 125 Smack 126 Dynamics III plug ins 59 common controls 59 Compressor Limiter 62 De Esser III 68 Expander Gate 65 side chain
318. n Presets 288 Chapter 48 SansAmp PSA 1 289 PSA 1 Controls 290 PSA 1 Tips and Tricks 290 Contents xi Part IX Dither Plug Ins Chapter 49 Dither 292 Dither Controls 292 Chapter 50 POW r Dither 294 POW r Dither Controls 294 Part X Sound Field Plug Ins Chapter 51 AutoPan 297 AutoPan Controls 297 Using AutoPan
319. n Utilities 347 Duplicate The Duplicate plug in duplicates the selected audio in place Depending on how its controls are config ured the new clip will appear in either the Clip List or playlist You can use this to flatten or consoli date an entire track consisting of multiple clips into one continuous audio file that resides in the same place as the original individual clips The audio is unaffected by Pro Tools volume or pan automation or by any real time plug ins that may be in use on the track as inserts The original audio file clips are merely rewritten in place to a single duplicate file The Duplicate plug in works nondestructively You cannot choose to overwrite files To duplicate an audio selection 1 Select the audio you want to duplicate 2 Choose AudioSuite gt Other gt Duplicate 3 Ensure that Use In Playlist is enabled 4 Click Render Gain The Gain plug in boosts or lowers a selected clip s amplitude by a specific amount Use it to smooth out undesired peaks and other dynamic inconsis tencies in audio material Gain Specifies the gain amount Set this value by manually adjusting the Gain slider by entering a numeric decibel value or by entering a percentage Analyze When clicked displays the peak amplitude value of the current selection RMS Peak Toggle Switches the calibration of gain adjustment between Peak or RMS modes Peak mode adjusts the gain of the input signal to the maximum
320. n also use it in audio post produc tion for pull up and pull down conversions as well as for adjusting audio to specific time or SMPTE durations for synchronization purposes Time Shift Controls Time Shift controls are organized in the following four sections Audio Use the controls in the Audio section to se lect the most appropriate time compression and ex pansion algorithm mode for the type of material you want to process and to attenuate the gain of the processed audio to aid clipping Time Use the controls in the Time section to spec ify the amount of time compression or expansion you want to apply Formant or Transient Use the controls in the For mant or Transient section to adjust either the amount of formant shift or the transient detection depending upon which mode you have selected in the Audio section The Formant section is only available when Monophonic is selected as the Au dio Mode The Transient section is available with slightly different controls depending on whether Polyphonic or Rhythmic is selected as the Audio Mode Pitch Use the controls in the Pitch section to apply pitch shifting Pitch shifting can also be formant correct if you select the Monophonic audio setting Time Shift Chapter 24 Time Shift 138 Time Shift Audio Controls The Audio section of Time Shift provides controls for specifying the type of audio you want to process and gain attenuation of the processed signal to avoid clip
321. n and Time Adjuster see the Pro Tools Reference Guide Chapter 35 TimeAdjuster 216 While phase inversion controls have been used for many years by engineers as creative tools for ad justment of frequency response between multiple microphones sample level delay adjustments pro vide far more control Creative use of this control can provide a powerful tool for adjusting frequency response and timing relationships between audio signals recorded with multiple microphones Using TimeAdjuster for Manual Delay Compensation DSP and host based processing in all digital sys tems incurs delay of varying amounts You can use the TimeAdjuster plug in to apply an exact number of samples of delay to the signal path of a Pro Tools track to compensate for delay incurred by specific plug ins TimeAdjuster provides presets for com mon delay compensation scenarios To compensate for several plug ins in line use the delay times from each settings file as references and add them together to derive the total delay time Alternatively look up the delay in samples for the plug ins you want to compensate for then apply the appropriate amount of delay To manually compensate for DSP induced delays try one of the following methods Phase inversion Comb filter effect cancellation Phase Inversion If you are working with phase coherent track pairs or tracks recorded with multiple microphones you can invert the phase to negate th
322. n at the top of the Space window When selected Picture Preview mode shows pictures as sociated with the IR For an IR provided with Space this will usually include a photograph of the location and an image with technical details such as microphones used or an overview of the micro phone setup Thumbnails of images are displayed in the right hand column In this mode the IR browser can be used to view the associated pictures without loading the IR itself Space Snapshot Mode Snapshot mode is selected using the Snapshot icon at the top of the Space window Space provides up to ten snapshots that are available at all times Each snapshot stores a separate IR waveform and all control settings Snapshots are optimized for quick loading into the convolution processor and switch ing between snapshots is considerably faster than loading a new IR Snapshot mode allows all ten snapshots to be viewed as well as the option to se lect rename copy paste and clear snapshots The name of the currently selected snapshot is al ways displayed in the Info bar at the bottom of the display area and can be automated This lets you switch reverb settings during playback and is use ful for post production sessions where the reverb setting may change as the scene changes The active snapshot can be selected in one of two ways At any time a snapshot can be selected by using the snapshot menu in the Info bar Alterna tively when the display ar
323. n incoming signal peak is too fast for the current compression setting the cursor temporarily leaves the gain transfer curve LFE Enable button Compressor Limiter III shown The LFE Enable button is not available if the plug in is not inserted on an applicable track Dynamics graph display Gain transfer curve and cursor showing amount of compression Input signal level x axis Output signal level y axis Threshold Chapter 12 Dynamics III 62 The cursor changes color to indicate the amount of compression applied as shown in the following ta ble Dynamics III Side Chain Section For information on using the Side Chain section of the Compressor Limiter or Expander Gate see Using Dynamics III Key Input for Side Chain Processing on page 73 Compressor Limiter III The Compressor Limiter plug in applies either compression or limiting to audio material depend ing on the ratio of compression used About Compression Compression reduces the dynamic range of signals that exceed a chosen threshold by a specific amount The Threshold control sets the level that the signal must exceed to trigger compression The Attack control sets how quickly the compressor re sponds to the front of an audio signal once it crosses the selected threshold The Release control sets the amount of time that it takes for the com pressor s gain to return to its original level after the input signal drops below the selected
324. n processes audio by reducing its sample rate and bit resolution It is ideal for emulating the grungy quality of 8 bit samplers Lo Fi Controls Sample Rate The Sample Rate slider adjusts an audio file s play back sample rate in fixed intervals from 700 Hz to 33 kHz in sessions with sample rates of 44 1 kHz 88 2 kHz or 176 4 kHz and from 731 Hz to 36 kHz in sessions with sample rates of 48 kHz 96 kHz or 192 kHz Reducing the sample rate of an audio file has the effect of degrading its audio quality The lower the sample rate the grungier the audio quality The maximum value of the Sample Rate control is Off which effectively means bypass Lo Fi The range of the Sample Rate control is slightly different at different session sample rates because Lo Fi s subsampling is calcu lated by integer ratios of the session sample rate Chapter 45 Lo Fi 282 Anti Alias Filter The Anti Alias control works in conjunction with the Sample Rate control As you reduce the sample rate aliasing artifacts are produced in the audio These produce a characteristically dirty sound Lo Fi s anti alias filter has a default setting of 100 automatically removing all aliasing artifacts as the sample rate is lowered This control is adjustable from 0 to 100 letting you add precisely the amount of aliasing you want back into the mix This slider only has an effect if you have reduced the sample rate with the Sample Rate control
325. n these alterations in the waveform of the audio material Time Compression Expansion plug in Normalizing a selection before using Time Compressing Expansion may produce better results Audio Plug Ins Guide 344 Crossfade length affects the amount of smoothing performed on audio material This prevents audio artifacts such as clicks from occurring Long cross fade times may over smooth a signal and its tran sients This may not be desirable on drums and other material with sharp transients Use the Crossfade slider to manually adjust and optimize crossfade times if necessary For audio material with sharper attack transients use smaller crossfade times For audio material with softer at tack transients use longer crossfade times Min Pitch Sets the minimum or lowest pitch that will be used in the plug in s calculations during the Time Compression Expansion process The con trol has a range of 40 Hz to 1000 Hz This control should be set lower when processing bass guitar or audio material with a low frequency range Set this control higher when processing higher frequency range audio material Accuracy Prioritizes the processing resources al located to audio quality Sound or timing Rhythm Moving the slider towards Sound generally results in better sonic quality and fewer audio artifacts Moving the slider towards Rhythm puts the emphasis on keeping the tempo consistent When you are working with au
326. nabled the Re lease control is overridden and the Release value display is grayed out Channel Linking Pro Limiter provides four different options for de termining how limiting processing is applied to greater than stereo multi channel formats Stereo Pairs When selected limiting is only ap plied to the Left and Right stereo pairs only when either the Left or Right incoming signal exceeds the Threshold setting Similarly limiting is only ap plied to the Left Surround and Right Surround ste reo pairs only when either the Left Surround or Right Surround incoming signal exceeds the Threshold setting The processing of the Center channel if present is applied separately only when the incoming signal on for the Center channel ex ceeds the Threshold setting All w LFE When selected limiting is applied to all incoming channels whenever any channel exceeds the Threshold setting All no LFE When selected limiting is applied to all incoming channels whenever any channel except the LFE channel exceeds the Threshold set ting Front Back When selected limiting is only applied to the Left Center if present and Right channels only when the incoming signal on any front chan nel input exceeds the Threshold setting Limiting is only applied to the Surround channels only when the incoming signal on any surround or back channel exceeds the Threshold setting Auto Release When Auto Release is enabled Pro Limiter over
327. nal program material A full color histogram plots input dB history during playback and provides visual feedback for setting threshold level A user adjustable ceiling lets material be level optimized for recording Dither for noise shaping during final mixdown Online Help provides descriptions of each con trol shown by holding the pointer over any of the elements in Maxim s display Maxim Chapter 17 Maxim 92 About Peak Limiting Peak limiting is an important element of audio pro duction It is the process of preventing signal peaks in audio material from clipping by limiting their dynamic range to an absolute user selectable ceil ing and not letting them exceed this ceiling Limiters let you select a threshold in decibels If an audio signal peak exceeds this threshold gain re duction is applied and the audio is attenuated by a user selectable amount Limiting has two main uses in the audio production cycle Adjusting the dynamic range of an entire final mixdown for premastering purposes Adjusting the dynamic range of individual in struments for creative purposes Limiting a Mix The purpose of applying limiting during final mix down is to flatten any large peaks remaining in the audio material to have a higher average signal level in the final mix By flattening peaks that would oth erwise clip it is possible to increase the overall level of the rest of the mix This results in
328. nce Guide Settings menu Librarian menu Chapter 44 Eleven 264 Master Section The Master section includes plug in I O input output and noise gate controls the Amp Type selector and the Cab Type selector The Master section doesn t change when you switch amps Master section settings are stored and recalled with plug in presets Input LED The Input LED shows green yellow orange or red to indicate whether you are under or over driving the plug in The Input LED is before the Input sec tion of the Master section To learn more about the Input LED within the Eleven signal chain see Eleven Signal Flow Notes on page 280 Input The Input knob provides input trim boost for tone and distortion control The Input range is 18 dB to 18 dB The Input knob provides a great way to increase or decrease gain with amp models that don t have a separate preamp control It also provides a way to trim or boost the level of pre recorded tracks you want to treat with Eleven The setting of the Input knob is saved and restored with Settings files presets Output The Output control sets the output gain after pro cessing letting you make up gain or prevent clip ping on the channel where the plug in is being used Output range is 60 dB to 18 dB Amp Type Amp Type selects which amplifier model to use see Amp Types on page 265 Cab Type This selector lets you select which speaker cabinet model
329. nd Mid Freq The Mid and Mid Freq controls allow you to adjust mid frequencies from 500Hz to 3 5KHz 11 Treble The Treble control adjusts high frequencies 11 Gain The Gain control allows you to adjust the out put level 11 JOEMEEK Meequalizer VC5 EQ Chapter 7 Pultec Plug Ins 33 Chapter 7 Pultec Plug Ins The Pultec plug ins are a set of EQ plug ins that are available in DSP Native and AudioSuite formats The following plug ins are included Pultec EQP 1A Pultec EQH 2 Pultec MEQ 5 The Pultec plug ins support 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates The Pultec plug ins operate as mono multi mono or stereo plug ins Pultec EQP 1A The Pultec EQP 1A provides smooth sweet EQ and an extremely high quality tube audio signal path Use it on individual tracks critical vocals or even across a stereo mix for mastering applications Built in the early 1960s the Pultec EQP 1A offers gentle shelving program equalization on bass and highs and offers a variable bandwidth peak boost control A custom and secret filter network pro vides all its equalization functionality Quality transformers interface it to real world studio equip ment A clean and well designed tube amplifier provides a fixed amount of make up gain Pultec EQP 1A Controls Low Frequency Section Adjust low frequencies using the Boost and Atten knobs and the Low Fre quency switch located at th
330. ndicate if the left or right channel has clipped The contents of this browser can be sorted in as cending and descending order by any column sim ply by clicking on a column one or more times The time information displayed in this browser is relative to where the transport started The Offset field can be used to adjust the timecode values if MasterMeter is being used for historical metering but the session was started from a point other than the beginning If MasterMeter is being used in real time the timecode information in this column can be ignored Signal Clip Events browser Oversampled Clip Events browser Chapter 56 MasterMeter 331 At the bottom of the browser the Peak field dis plays the highest dB value of the oversampled au dio received so far The Events field shows the his torical total of clip events in the oversampled audio signal Once MasterMeter reaches 2000 clip events it ceases to record additional events Al though the meters remain active and the Peak field continues to be updated new events will not be added to the browsers The Events field flashes 2000 to indicate this condition The Oversampling field displays the current over sampling factor in use by the DSP processing This will vary between 2x 4x and 8x oversampling de pending on the session sample rate The information in this browser is cleared using the Clear button or is cleared automatically whenever the Pro Tools transport is
331. nee adjustment Cursor indicates Depth adjustment Detection mode options Smart mode selected Chapter 18 Pro Compressor 102 Attenuation Listen Mode Attenuation Listen mode lets you isolate the gain reduction part of the processed audio signal This can help you hear what parts of the input signal are triggering compression which in turn can help you better understand the characteristics of the compressor with the current settings To enable or disable Attenuation Listen mode Click the Attenuation Listen button the speaker icon at the top right of the dynamics graph so that it is highlighted The button flashes while Attenuation Listen mode is enabled To disable it click the button again so that it is not high lighted Pro Compressor Controls Threshold The Threshold control sets the level that an input signal must exceed to trigger compression A signal will be compressed if its level exceeds this setting If the signal level falls below this value no com pression will occur Ratio The Ratio control sets the compression ratio the amount of compression applied as the input signal exceeds the threshold For example a 2 1 compres sion ratio means an input level that is 2 dB above the threshold will be attenuated resulting in an out put level that is 1 dB over the threshold The com pression ratio ranges from 1 0 1 to 20 0 1 Once the Ratio control hits 21 0 1 it displays LMTR The LMTR setting
332. net Bypass to bypass Cabinet and microphone processing 7 Open one of the Eleven plug ins on any of the three selected Aux Input tracks and Opt Shift click Mac or Alt Shift click Windows the Amp Bypass switch 8 Solo the first Aux Input track Three tracks selected Chapter 44 Eleven 275 9 Click to open the Eleven plug in window on the first Aux Input and do any of the following Choose a cabinet Choose a mic and its position Adjust Speaker Breakup 10 When you re done close the plug in window and then unsolo the track 11 Solo the next Aux Input track and repeat to con figure its cabinet and mic settings 12 Repeat for other Aux Input tracks to configure their cabinet and mic settings 13 When you have set your cabinet tones make sure to unsolo all the Aux Inputs and begin play ing so you can hear the combined tone of all three cabinet channels 14 Do the following to continue Balance the tracks using the volume faders on the Aux Input tracks Try different pan positions for each Aux Input track Evaluate the phase relationships of the combined signals and adjust accordingly see Phase Con siderations with Blending in Eleven on page 277 If You Plan on Blending Cabinets The Eleven plug in emulates the variation in cabi net response that is unique to each amp cab combi nation In the physical world these variations are the result of the distinct loads p
333. new to guitar or new to Pro Tools you might want to know about a few simple effects you can add to your Eleven guitar tracks using nothing more than a few of the plug ins included with Pro Tools Bussing and Submixing Not so much a plug in or effect as a standard oper ating procedure multiple guitar tracks are often submixed to stereo Aux Input for centralized level control of those tracks This is especially useful for applying compression or limiting creating stem mixes and many other practical uses See your Pro Tools Reference Guide for mixing and submix ing setups and suggestions and try them out while exploring some of the following effects sugges tions Dynamics Compression limiting expansion and gating are all useful effects for guitar Different results can be achieved using each of the different types of dy namics processing in combination with signal routing for individual discrete versus submix shared resource processing Here are a few exam ples If all you seek is the taming of occasional dy namic aberrations within a track meaning you just need to clamp a couple overs try putting a lim iter on the individual track after Eleven To glue multiple rhythm tracks or tones to gether bus them to a stereo Aux Input and apply heavy compression or limiting to that Aux Input Experiment with different dynamics plug ins such as Dyn 3 or any of Avid s classic compressor pro cess
334. ng is shown on the right side of the Input meters from 0 dB to 36 dB To better see the Gain Reduction meters you can dim the Input meters see Dim Input Meter Tog gle on page 119 Gain Reduction meters shown with Input meters dimmed Dim Input Meter toggle Gain Reduction meters Input meters dimmed Chapter 20 Pro Limiter 120 Pro Limiter Output Section The Output section provides output metering and controls for adjusting the level of the output signal Output Meters The Output meters show peak signal levels after processing The Output meter scale is shown on the right side of the Output meters from 90 dB to 6 dB Output Sample Peak Hold Display Pro Limiter provides a numerical display for the output sample peak hold value in dB above the output meters To reset the value click it Pro Limiter Controls Pro Limiter provides controls for setting the Threshold Ceiling Character soft saturation limiting and Release time Threshold The Threshold control sets the level an input signal must exceed to trigger limiting Signals that fall be low the Threshold setting are unaffected To adjust the input threshold do one of the following Click in the Threshold field to type a value 30 dB to 0 dB Click the Threshold control and drag up or down Click the Threshold control on the Input meters and drag up or down Ceiling The Ceiling control sets the max
335. ng on the type of track mono stereo or multichannel on which the plug in has been inserted When Side Chain Listen is enabled the Output meter only displays the levels of the side chain signal See Dynamics III Side Chain Listen on page 71 Input left and Output right meter buttons Chapter 12 Dynamics III 61 Dynamics III LFE Enable Pro Tools HD Only The LFE Enable button located in the Options sec tion is on by default and enables plug in process ing of the LFE low frequency effects channel on a multichannel track formatted for 5 1 6 1 or 7 1 surround formats To disable LFE processing de select this button Dynamics III Graph Display The Dynamics Graph display used with the Com pressor Limiter and Expander Gate plug ins shows a curve that represents the level of the input signal on the horizontal x axis and the level of the output signal on the vertical y axis The orange vertical line represents the threshold Use this graph as a visual guideline to see how much dynamics processing you are applying The Compressor Limiter and Expander Gate plug ins also feature an animated multi color cursor in their gain transfer curve displays The gain transfer curve of the Compressor Limiter and Expander Gate plug ins shows a moving ball cursor that shows the amount of input gain x axis and gain reduction y axis being applied to the in coming signal To indicate overshoots when a
336. ng to the current amp but does alter the value of that unused control If you switch to a different amp that does include that previously unused control the new amp inherits the altered setting which can lead to sudden jumps in gain or other set tings Using MIDI and MIDI Learn with Eleven Eleven supports MIDI Control Change CC mes sages meaning that the Master section amp cabi net and mic parameters can be controlled remotely by any CC capable MIDI device This includes MIDI controllers mixers and instruments as well as the 003 in MIDI Mode MIDI Learn lets you quickly map plug in controls to a MIDI foot pedal switch fader knob or other CC compatible trigger You can also manually as sign controls to specific MIDI CC values MIDI control assignments are saved and restored with the Pro Tools session in which they are de fined Settings files presets for Eleven do not store or recall MIDI Learn assignments Previous arrow top and Next arrow bottom Amp Type shown You can control the Amp Cab and Mic Type selectors with MIDI See Using MIDI and MIDI Learn with Eleven on page 262 See the Pro Tools Reference Guide for more information on plug in automation Chapter 44 Eleven 263 To map a MIDI controller to a parameter 1 Make sure your external MIDI device is con nected to your system and recognized by your MIDI Studio Setup Windows or Audio MIDI Setup Mac 2 Create a MIDI tra
337. nge Low High Shelf 0 1 to 2 0 Q Range Peak Notch 0 1 to 10 0 Q Default All 1 0 Gain Range Low High Shelf 12 dB to 12 dB High Peak Gain Range 18 dB to 18 dB 1 Band EQ set to High Pass Filter 1 Band EQ set to Notch Filter 1 Band EQ set to High Shelf EQ Chapter 4 EQ III 19 Low Shelf EQ The Low Shelf EQ boosts or cuts frequencies at and below the Frequency setting The amount of boost or cut is determined by the Gain setting The Q set ting determines the shape of the shelving curve Peak EQ The Peak EQ boosts or cuts a band of frequencies centered around the Frequency setting The width of the affected band is determined by the Q setting Low Pass Filter The Low Pass filter attenuates all frequencies above the cutoff frequency setting at the selected rate 6 dB 12 dB 18 dB or 24 dB per octave while letting all frequencies below pass through No gain control is available for this filter type 1 Band EQ set to Low Shelf EQ 1 Band EQ set to Peak EQ 1 Band EQ set to Low Pass Filter Chapter 4 EQ III 20 7 Band EQ III The 7 Band EQ has the following available bands High Pass Low Notch Low Pass High Notch Low Shelf Low Peak Low Mid Peak Mid Peak High Mid Peak and High Shelf High Peak All seven bands are available for simultaneous use In the factory default setting the High Pass Low Notch and Low Pass High Notch bands are out of circuit the Low Shelf and High Shelf bands are
338. ngth in lower gain sounds With high gain amp models the Middle control has a more dramatic ef fect and can noticeably shape the sound of the amp at both the minimum and extreme settings The de fault setting is 5 0 The Middle range is from 0 to 10 Treble In most amp models the Treble control is the stron gest of the three tone controls Its setting deter mines the blend and strength of the Bass and Mid dle controls When Treble is set to higher values it becomes the dominant tone control minimizing the effect of Bass and Middle controls When Treble is set to lower values the Bass and Middle have more effect making for a darker amp tone The default setting is 5 0 The Treble range is from 0 to 10 Presence The Presence control provides a small amount of boost at frequencies above the treble control Pres ence is applied at the end of each amp model pre amp stage acting as a global brightness control that is independent of other tone controls The default setting is 3 0 The Presence range is from 0 to 10 Master The Master control sets the output volume of the pre amp acting as a gain control for the power am plifier In a standard master volume guitar amp as the Master volume is increased more power tube distortion is produced The default setting is 5 0 Master range is from 0 to 10 Tremolo Tremolo is achieved through the use of amplitude modulation multiplying the amplitude of the pre amp output by a
339. nic mode adjust the Formant Shift control 6 If using Polyphonic or Rhythmic mode adjust the Transient controls 7 Make sure Time Shift is set to 0 unless you also want to change the duration of the section 8 Adjust the Pitch Shift control to set the amount of pitch change Pitch change is measured in semitones and cents or as a percentage of the original 9 Click Render Changing the Time and Pitch Using Time Shift To change the time and pitch of a selected audio clip 1 Select AudioSuite gt Pitch Shift gt Time Shift 2 Select Varispeed from the Audio Mode pop up menu 3 Adjust either the Time Shift or Pitch Shift con trol to set the amount of time and pitch change in terms of a percentage of the original 4 Click Render Using the Monophonic Polyphonic or Rhythmic modes you can adjust both the Time Shift and Pitch Shift controls inde pendently before processing Chapter 24 Time Shift 144 Post Production Pull Up and Pull Down Tasks with Time Shift The table below provides information on TCE settings for common post production tasks Type the corre sponding TCE represented to 10 decimal places in the table in the Time Shift field for the corresponding post production task and the process the selected audio Pull Up or Pull Down TCE to 10 Decimal Places Frames Pal to Film 4 tfx 96 0 25 to 24 30 PAL to NTSC 4 1 tfx 95 9040959041 25 to 23 976 29 97 Film to PAL 4 1667
340. nnels You can also use Link mode to help you maintain a relative offset between control settings on the two channels To maintain an offset between channels 1 Deselect the Link button 2 Select a channel button left or right and adjust the controls for that channel 3 Select Link mode and adjust the same controls for the opposite channel D2 will maintain the relative offset between the two channels Left Channel Right Channel and Link buttons To copy the control settings of the active channel to the opposite channel Option click Mac or Alt click Windows while linking channels Chapter 6 JOEMEEK VC5 Meequalizer 32 Chapter 6 JOEMEEK VC5 Meequalizer The JOEMEEK VC5 Meequalizer is an EQ plug in that is available in DSP Native and AudioSuite formats VC5 supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates VC5 operates as a mono multi mono or stereo plug in The VC5 offers simple controls with incredibly warm musical results Among countless other achievements Joe Meek built custom gear to get the sounds in his head onto tape One device was a treble and bass circuit with a sweepable mid con trol built into a tiny tobacco tin The Meequalizer VC5 virtually recreates the exact circuitry used by Joe Meek in this device JOEMEEK Meequalizer Controls Operation of the Meequalizer is simple and to the point Bass The Bass control adjusts low frequencies 11 Mid a
341. nnels always show the input level pre fader for the channel regard less of the Source Level setting Downmix The Downmix section of the Down Mixer plug in provides output meters and a single fader to adjust the output level of the Down Mixer from 45 dB to 12 dB Part XI Instrument Plug Ins Chapter 53 Click II 307 Chapter 53 Click II Click II is a metronome plug in The Click II plug in creates an audible click during session playback that you can use as a tempo reference when performing and recording The Click II plug in receives its tempo and meter data from the Pro Tools application letting it follow any changes in tempo and meter in a session The Click II plug in is a Native mono only plug in Several click sound presets are included Click II supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Click II Controls and Displays Click II provides displays for the session Meter and Tempo settings which can be set manually in the Transport window or can follow the conductor rul ers for Meter and Tempo in the session Timeline Beat Display The Beat display shows the number of beats in a bar as determined by the Meter for the session If the session contains meter changes the Beat dis play shows the number of beats in a bar for the Me ter at the current location of the Playback Cursor Follow Meter When the Follow Meter option is enabled high lighted Click II follows t
342. no Stereo L C R L C R S Quad 5 0 5 1 Cutting or boosting an EQ frequency band Setting the EQ crossover frequency Setting the rear cut frequency Chapter 29 ReVibe II 169 ReVibe II Input and Output Meters The Input and Output meters indicates the input and output signal levels These meters range from 0 dB to 96 dB The number of input and output meters that operate simultaneously ranges from a single meter for mono input and output up to five input and output meters for 5 0 and 5 1 multichan nel processing The number of meters displayed de pends on the channel format of the track on which the plug in is inserted Clip Indicators A red channel clip indicator appears at the top of each meter The clip indicator lights when the sig nal level exceeds 0 dB and stays lit until cleared Clicking a meter s clip indicator clears that meter ReVibe II Controls ReVibe II has a variety of controls for producing a wide range of reverb effects Controls can be ad justed by dragging their sliders typing values di rectly in their text boxes and adjusted on the Decay Color and EQ graph displays Room and Reverb Type ReVibe II lets you select the type of Room and Re verb modeled Each Room and Reverb type models early reflection characteristics for specific types of rooms or effects devices Each
343. nput for the side chain while Key Listen lets you monitor the key input Input Level Output Level External Key and Key Listen toggles See Using the Side Chain Input in D3 on page 84 for detailed information on external side chain processing External Key Key Listen Chapter 14 Focusrite D3 81 Meters The meters indicate gain reduction the top meter and output level the bottom meter The Gain Re duction meter indicates the amount of gain reduc tion in dB The Output Level meter indicates the output signal level in dB In Stereo mode two Output Level meters appear one for each channel However a single Gain Re duction meter is used for both channels since the D3 RMS detector uses a composite control signal A red Clip Indicator appears to the right of the out put meter s Clicking on the Clip Indicator clears it Option clicking Mac or Alt clicking Win dows clears both channels when the plug in is used in stereo The following metering indications are used Green nominal levels Yellow pre clipping at 6 dB below full scale signal Red full scale signal clipping D3 Compressor Controls The Compressor icon which represents a compres sion curve acts as a three state switch for enabling disabling or bypassing the compressor Its current state is indicated by the icon s color White indicates enabled In this state the com pressor is active and using av
344. nse information display Impulse Response IR Loading and Organization Features Scrollable IR browser makes finding impulse responses easy Browser supports user defined IR groups on any local drives Browser keyboard shortcuts IR favorites function Automatically recognizes common IR formats for one click loading Quick browser buttons allow rapid IR loading and preview Automation and Ease of Use Features Snapshot mode supports rapid changes between ten predefined reverb scenes Picture preview mode allows you to view image files stored with impulse responses Impulse responses stored directly in Pro Tools presets and sessions for easy session sharing New impulse responses can be copied to system and loaded without closing Space iLok support for quick and easy relocation to other Pro Tools systems Surround and Post Production Features Full input and output surround metering on screen at all times Separate front center and rear levels Independent front and rear decay Snapshot mode ideal for post automation requirements Seamless snapshot switching Automatic phantom channel creation IR Library A wide variety of both real and synthetic reverb spaces and effects Mono stereo and surround formats All reverb impulse responses stored in WAV file format Chapter 30 Space 183 Space Overview The following sections provide
345. nse library divided into the following categories Category Description Halls Halls and auditoriums Churches Churches and chapels Rooms Large and small rooms Chambers Traditional studio reverb chambers Plates Classic electromechanical reverb plates Springs Classic electromechanical reverb springs Digital Reverbs Classic and contemporary digital reverb units Post Production Post production impulses Tiny Spaces Small reverbs from everyday objects Pure Spaces A selection of Pure Space impulses in multiple categories Effects Non reverb effects for sound design in multiple categories Colors Sound coloring and positioning Cosmic Spacey smears and washes Impressions Smears and washes that evoke an image Industrial Heavy machinery Periodic table Better living through chemistry Part VI Delay Plug Ins Chapter 31 Mod Delay III 204 Chapter 31 Mod Delay III Mod Delay III provides mono multi mono mono to stereo and stereo modulating delay effects Mod Delay III is available in DSP Native and Au dioSuite plug in formats Mod Delay III supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Mod Delay III Controls Mod Delay III provides separate sections in the plug in window for Input and Output metering De lay and Modulation controls and for the Wet Dry Mix control Stereo and mono to stereo versions provide meters and controls for e
346. ntrols 9 Dragging Plug In Controls 9 Editing Control Values 10 Dragging in Graphic Displays 10 Adjusting Controls with Fine Resolution 10 Resetting Controls to Default Values 10 Part II EQ Plug Ins Chapter 4 EQ III 12 EQ III Configurations 12 Adjusting EQ III Controls 13 EQ III I O Controls 15 EQ III EQ Band Controls 16 1 Band EQ III
347. o pan The track s channel width changes from mono to 5 0 2 Select a 5 0 output path from the track s Output selector 3 In the AutoPan Plug In window select a clock wise or counter clockwise Path 4 Adjust the Spread and Width sliders 5 From the LFO Waveform selector select Half Sine 6 Adjust the Rate slider 7 Play back the session to hear the mosquito fly ing around your head Using the Side Chain Input The Side Chain Input option in AutoPan lets you direct audio from another track in your Pro Tools session to the Envelope Detector This is achieved by sending the audio from a channel to a bus and setting the side chain input on AutoPan to the same bus Try automating Spread and Width to alter the positioning of the panned sound Try automating the Manual control instead of using the LFO to create a more erratic pan ning of the mosquito sound Try automating Rate to alter the speed of the panned sound over time For more information on using the Side Chain Input see the Pro Tools Reference Guide Chapter 52 Down Mixer 304 Chapter 52 Down Mixer Avid Down Mixer can be used to automatically mix greater than stereo multichannel tracks such as 5 1 down to stereo Pro Tools HD only or ste reo tracks down to mono Down Mixer is available in DSP and Native formats Down Mixer supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Down Mixer support
348. o accommodate the legal range of digital audio on a PCM sampled system The goal of MasterMeter is to allow an engineer to use a DSP model of the reconstruction process to monitor the reconstructed waveform for potential clipping at the final mix and mastering stages Us ing MasterMeter engineers can compare regular and intersample peaks over time and make appro priate adjustments without sacrificing overall level or dynamic range Utilizing an oversampled peak meter in the digital audio studio that represents the reconstruction filters in digital to analog converters is the first step toward an improvement in audio quality in music releases Using MasterMeter MasterMeter uses the DSP power of Pro Tools to model the conversion process found in typical con sumer devices In technical terms the MasterMeter algorithm uses a 31 tap Blackman Harris win dowed sync conversion with oversampling ratios from 2x to 8x depending on the session sample rate The output of this DSP algorithm is then displayed visually This assists engineers in highlighting po tential distortion which may be introduced on play back of mixes especially mixes which have been processed to be particularly loud or hot MasterMeter can be used in two different ways during a session Real Time Metering or Historical Metering Real Time Metering MasterMeter can be used to monitor live signal lev els even if the Pro Tools transport is stopped This can be
349. o frequencies below the Frequency setting rolling off at a slope of 12 dB per octave Side Chain Listen button Side Chain Listen is not saved with other plug in presets HF and LF Filter Enable buttons Band Pass filter Low Pass filter Chapter 12 Dynamics III 72 Dynamics III Side Chain HF Frequency Control The HF frequency control sets the frequency posi tion for the Band Pass or Low Pass filter and ranges from 20 Hz to 20 kHz Dynamics III Side Chain Low Frequency LF Filter Type The LF filter section lets you filter lower frequen cies out of the side chain signal so that only certain bands of low frequencies or higher frequencies are allowed to pass through to trigger dynamics pro cessing The LF side chain is switchable between Band Pass and High Pass filters Band Pass Filter Makes triggering of dynamics processing more sensitive to frequencies within the narrow band centered around the Frequency set ting and rolling off at a slope of 12 dB per octave High Pass Filter Makes triggering of dynamics processing more sensitive to frequencies above the Frequency setting rolling off at a slope of 12 dB per octave Dynamics III Side Chain LF Frequency Control The Frequency control sets the frequency position for the Band Pass or High Pass filter and ranges from 20 Hz to 20 kHz HF frequency controls Band Pass filter High Pass filter LF frequency controls Chapter 12 Dynamics III 73 Using D
350. o keep the Ex pander Gate in effect for longer periods of time with a single crossing of the threshold It can also be used to prevent gate chatter that may occur if varying input levels near the threshold cause the gate to close and open very rapidly This control ranges from 5 ms shortest hold to 4 seconds longest hold Expander Gate III Release Control The Release control sets how long it takes for the gate to close after the input signal falls below the threshold level and the hold time has passed This control ranges from 5 ms fastest release time to 4 seconds slowest release time Expander Gate III Range Control The Range control sets the depth of the Ex pander Gate when closed Setting the gate to higher range levels allows more and more of the gated au dio that falls below the threshold to peek through the gate at all times This control ranges from 80 dB lowest depth to 0 dB highest depth Expander Gate III Side Chain Section The side chain is the split off signal used by the plug in s detector to trigger dynamics processing The Side Chain section lets you toggle the side chain between the internal input signal or an exter nal key input and tailor the equalization of the side chain signal so that the triggering of dynamics processing becomes frequency sensitive See Dy namics III Side Chain Input on page 70 Chapter 12 Dynamics III 68 De Esser III The De Esser reduces sibilant
351. oSuite version of this plug in any settings for this parameter will be active BPM Sync controls Tempo Rate Triplet button On button Dot button Note Value display display Chapter 39 Reel Tape Flanger 231 Reel Tape Flanger Tips To achieve a flanging effect set the Range con trol within the Flange range and adjust the LFO Depth control to a value that is greater than the off set so that the variable delay crosses the zero point To achieve an ADT doubling effect set the Range control within either of the ADT ranges and adjust the LFO Depth control to a value that is smaller than the offset so that the variable delay does not cross the zero point To achieve a manual flanging effect set the LFO Depth control to 0 and vary or automate the Range control within the Flange range For fine control hold Control Windows or Command Mac while varying the Range control To add complexity to flanging or ADT effects turn up the Wow Flutter control to introduce more fluctuation in the variable delay Use Reel Tape Flanger in a send return configu ration to mix the dry signal with an aggressively driven flanged signal to control the amount of grunge in the final mix When you start playback the LFO sweep al ways starts at the bottom of the cycle so each time you start playback from the same location for ex ample at a bar line the ef
352. ocessing to the signal but not Exp Gate or you may want to only apply only a high pass filter To enable effects in the Dynamics or EQ Filters section Click the Enable Disable button for the effect you want to enable so that it is highlighted To disable effects in the Dynamics or EQ Filters section Click the Enable Disable button for the effect you want to disable so that it is not highlighted Listen Mode The Side Chain tab in the Dynamics section and the EQ and Filter tabs in the EQ Filter section pro vide a Listen button When enabled for the Side Chain Listen mode lets you hear the input signal that feeds the dynamic section This can be either the external key input or the internal side chain including the applied filter When enabled for any of the EQ bands Listen solos the corresponding EQ band and temporarily inverts the EQ Type so that you can tune the Fre quency and the Q for that EQ band When enabled for either of the Filter effects Listen solos the enabled Filter band and inverts the Filter This allows you to hear only hear the portion of the audio signal that is being removed by the filter To enable or disable Listen on the Side Chain effect EQ band or a Filter effect Click the Listen button for the Dynamics or EQ Filter tab you want so that it is highlighted Click it again so that it is not highlighted to dis able it Dynamics section Exp Gate disabled
353. oduction Pull Up and Pull Down Tasks with Time Shift on page 144 Time Shift Time section Chapter 24 Time Shift 140 Time Shift Formant Controls The Formant section of Time Shift lets you shift the formant shape of the selected audio independently of the fundamental frequency This is useful for achieving formant correct pitch shifting It can also be used as an effect For example you can formant shift a male vocal up by five semitones and it will take on the characteristics of a female voice The Formant section is only available when Mono phonic is selected as the Audio Type The Formant section provides a single control for transposing the formants of the selected audio by 24 00 semitones 2 octaves to 24 00 semitones 2 octaves with fine resolution in cents Adjust the Formant Shift control or click the Shift field and type a value About Formants Audio with a fundamental pitch has an overtone se ries or set of higher harmonics The strength of these higher harmonics creates a formant shape which is apparent if viewed using a spectrum ana lyzer The overtone series or harmonics have the same spacing related to the pitch and have the same general shape regardless of what the fundamental pitch is It is this formant shape that gives the audio its overall characteristic sound or timbre When pitch shifting audio the formant shape is shifted with the rest of the material which can result in an unnatura
354. of the low and high frequency ranges Low and high crossover points define the cut and boost points of three frequency ranges For best results set crossover points at least one oc tave higher than the frequency you want to boost or cut To boost a signal at 200 Hz for example set the crossover to 400 Hz EQ display Color display Chapter 29 ReVibe II 176 Low Frequency Crossover Control The Lo Crossover control sets the crossover fre quency at which transitions from low frequencies to mid frequencies take place in the reverberation filter The range of this control is from 50 0 Hz to 1 5 kHz Low Frequency Ratio Control The Lo Ratio control sets cut or boost ratios for the decay times of the low and mid frequency bands of the reverberation filter The range of this control is between 1 16 0 and 4 0 1 High Frequency Crossover Control The Hi Crossover control sets the crossover fre quency at which transitions from mid frequencies to high frequencies take place in the reverberation filter The range of this control is from 1 5 kHz to 20 0 kHz High Frequency Ratio Control The Hi Ratio control sets cut or boost ratios for the decay times of the mid and high frequency bands of the reverberation filter The range of this control is between 1 16 0 and 4 0 1 ReVibe II Contour Display The Contour display shows the current reverb shape and early reflections graphically Both front and rear reverb tail shapes and
355. on depends on many factors including the signal level of the source material these presets are just starting points With some experimentation Reel Tape Sat uration can yield warmer sounding results than conventional digital compression Bass Drum Rounds out and adds consistency to bass drum hits Bass Gtr Adds consistency and warmth to bass guitar sound while avoiding compression artifacts Snare Drum Reduces harsh peaks resulting from EQ boosted snare drum or rim shots Chapter 48 SansAmp PSA 1 289 Chapter 48 SansAmp PSA 1 SansAmp PSA 1 is a guitar amp simulator plug in Punch up existing tracks or record great guitar sounds with the SansAmp PSA 1 Capture bass or electric guitar free of muddy sound degradation and dial in the widest range of amplifier harmonic generation cabinet simulation and equalization tone shaping options available Tube sound speaker simulation warm equalization and cool lo fi textures no wonder thou sands of records feature the classic sounds of SansAmp SansAmp PSA 1 is available in DSP Native and AudioSuite formats SansAmp PSA 1 supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates SansAmp PSA 1 operates as a mono multi mono or stereo plug in B Andrew Barta of Tech 21 Inc introduced the SansAmp Classic in 1989 A guitar player with both a trained ear and electronics expertise An drew and Tech 21 pioneered the market for tube amplifier emulation
356. onal character and can be used for a variety of effects beyond pure reverb applications Depending on the capture technique used the IR may be suitable for use with mono stereo surround or a combination of those formats For example a capture setup with a single sound source and two microphones is ideal for a mono to stereo IR Multiple IRs may be taken of a physical space where the sound source has been moved to differ ent physical locations Each resulting IR may be used to create individual reverbs for separate in struments This effectively allows an engineer to place each instrument in the reverb sound field as if the instruments were physically arranged in the space Space IR Library Installation You can download IR Libraries from Avid s Space Online IR Library For more information on down loading and installing IR Libraries from the Space Online IR Library see Installing Space IR Pack ages on page 196 Using Third Party IRs in Space Space reads a wide range of IR formats automati cally including WAV and AIFF file formats al lowing you to import a variety of IRs Space sup ports IR sample rates from 22 kHz up to 96 kHz in bit depths from 16 to 32 bits In addition Space supports the display of JPEG format picture files stored with IRs To use third party IR libraries with Space 1 In the IR Browser select Edit gt Import Other IR Folder 2 Locate and select the library on your hard drive 3 C
357. onic imprint on guitar piano vocals and drums Because it s so easy to control you ll be getting classic tones in no time with the BF 3A BF 3A Controls The Peak Reduction and Output Gain controls combine with the Comp Limit switch to determine the amount and sound of the compression The fol lowing controls and meters are provided Peak Reduction Peak Reduction controls the amount of signal entering the side chain The more Peak Reduction you dial in the more squashed and compressed the sound will be Too little peak reduction and you won t hear any compression ac tion too much and the sound becomes muffled and dead sounding Output Gain Output Gain provides makeup gain to make the signal louder after passing through the peak reduction BF 3A Chapter 9 BF 3A 41 Comp Lim The Comp Limit switch affects the compression ratio The common setting for audio production is Comp which provides a maximum compression ratio of approximately 3 1 In Limit mode the unit behaves more like a broadcast lim iter with a higher threshold and compression ratio of approximately 15 1 Meter Both Gain Reduction and Output metering are provided The Meter knob operates as follows When set to Gain Reduction the meter needle moves backward from 0 to show the amount of compression being applied to the signal in dB When set to Output the needle indicates the out put level of the signal The meter is calibrated with 0
358. ons 111 Filter Type options 116 Side Chain Processing Graph display 116 Source selector 114 Threshold control 112 Pro Limiter plug in 117 Ceiling control 120 Character control 121 Input Level 118 Input meters 119 Release control 121 Threshold control 120 Pultec EQH 2 plug in 34 Audio Plug Ins Guide 356 Pultec EQP 1A plug in 33 phase EQ filters 35 program equalization 33 smooth EQ 33 Pultec MEQ 5 plug in 34 pumping 76 Purple Audio MC77 plug in 125 R Recti Fi plug in 283 Alternating Rectification 284 Alt Max Rectification 284 Mix control 285 negative excursion period of waveform 284 Negative Rectification 284 Output Meter 285 Positive Excursion 284 Positive Rectification 284 Post Filter 285 Pre Filter 283 Rectification 284 Subharmonic synthesis 285 zero crossing 284 Reel Tape Delay plug in 209 Bass control 211 Feedback control 210 Mix control 211 Noise parameter 211 presets 212 Speed control 210 synchronizing to session tempo 212 Treble control 211 Wow Speed parameter 211 Wow Flutter control 211 Reel Tape Flanger plug in 227 Feedback control 229 Invert parameter 229 LFO Depth control 229 LFO Rate control 229 Mix control 229 Noise parameter 230 presets 231 Range control 228 synchronizing to session tempo 230 Wow Flutter control 229 Reel Tape plug ins 227 Drive control 210 228 286 Output control 210 228 286 Tape Formula control 210 228 287 Tape Machine control 210 228 286 Reel Tape Saturation plug i
359. ooger plug ins Analog Delay 207 Moogerfooger 12 Stage Phaser 222 Moogerfooger Lowpass Filter 219 Moogerfooger Ring Modulator 225 Moogerfooger Ring Modulator plug in 225 Carrier Oscillator 226 LFO 225 metallic 226 oscillator 225 sinusoidal 225 sum and difference frequencies 225 tremolo 225 vibrato 225 N Normalize plug in 348 P phase 277 Pitch II Input Polarity 134 Pitch II plug in 133 Effects controls Delay 136 Feedback 136 Low Pass Filter 136 Mix 136 Input and Transient controls Clip indicator 134 Follow 135 Input 134 Level indicator 135 Range 134 Threshold 135 Window 135 Pitch Shift controls Coarse 135 Fine 135 keyboard 135 Link 135 Ratio 135 Pitch transposition 134 POW r Dither plug in 294 bit resolution for Dither plug in 294 Noise Shaping 294 Pro Compressor plug in 97 Attack control 103 Attenuation meters 99 Depth control 104 Dry Mix control 104 Dynamics Graph display 100 Input Level 98 Input meters 98 Knee control 103 Makeup control 104 Ratio control 102 Release control 103 Side Chain Detection options 101 Filter Type options 106 Side Chain Processing Graph display 106 Source selector 104 Threshold control 102 Pro Expander plug in 107 Attack control 113 Attenuation meters 109 Depth control 113 Dry Mix control 113 Dynamics Graph display 109 Hold control 113 Hysteresis control 113 Input Level 108 Input meters 108 Knee control 112 Ratio control 112 Release control 113 Side Chain Detection opti
360. op of the Output meters indicate clipping at the input or output stage of the plug in Clip indicators can be cleared by clicking the indicator Toggling Multichannel Input and Output Meters With multichannel track types LCRS and higher both Input and Output meters cannot be shown at the same time Click either the Input or Output but ton to display the appropriate level meter The In put Output meters display is toggled to Output by default Gain Reduction Meter The Gain Reduction GR meter indicates the amount the input signal is attenuated in dB and shows different colors during dynamics process ing Light Orange Indicates that gain reduction is within the knee and has not reached the full ratio of compression Dark Orange Indicates that gain reduction is being applied at the full ratio for example 2 1 Threshold Arrow The orange Threshold arrow next to the Input meter indicates the current threshold and can be dragged up or down to adjust the threshold When a multi channel instance of the plug in has been configured to show only the Output meter the Threshold ar row is not displayed Phase Invert The Phase Invert button at the top of the Levels sec tion inverts the phase polarity of the input signal to help compensate for phase anomalies that can occur either in multi microphone environments or because of mis wired balanced connections The Input and Output meters display differently dependi
361. or bass guitar Settings in the 12 millisecond range work well on drums or percussion Decay Rate The Decay Rate control determines how much of the decay from a transient is heard in the processed audio when time stretching When time stretching using the Rhythmic setting the re sulting gaps between the transients are filled in with audio and Decay Rate determines how much of this audio is heard by applying a fade out rate Decay Rate is only available when Rhythmic is se lected as the Audio Type Adjust the Decay Rate up to 100 to hear the audio that is filling the gaps created by the time stretching with only a slight fade or adjust down to 1 0 to completely fade out between the original transients Time Shift Pitch Controls The Pitch section of Time Shift provides controls for pitch shifting the selected audio Use the Pitch control to transpose the pitch from 24 00 semi tones 2 octaves to 24 00 semitones 2 oc taves with fine resolution in cents Transpose Displays the transposition amount in semitones You can transpose pitch from 24 00 semitones 2 octaves to 24 00 semitones 2 oc taves with fine resolution in cents Adjust the Pitch control or click the Transpose field and type a value Shift Displays the pitch shift amount as a percent age You can pitch shift from 25 00 2 octaves to 400 00 2 octaves Adjust the Pitch control or click the Shift field and type a value The default va
362. or Alt click Windows the knob to rotate it in the opposite direction or click the lettering to select a specific setting Voce Chorus Vibrato Chapter 41 Voce Plug Ins 236 Voce Chorus Vibrato Tips and Tricks The classic setting for organ is C3 but you ll find other settings useful on a variety of instruments Some of our favorites include Electric Pianos Many electric pianos feature built in vibrato But if the sound you re using doesn t provide a realistic vibrato perhaps you re wrestling with a sampler track dry and apply the effect later Guitar A certain popular guitar amp has a knob that says Vibrato but it s really just Tremolo Tremolo is amplitude modulation the sound gets louder and quieter Vibrato in contrast imparts pitch change A select few highly sought after 50s Magnatone guitar amps feature a true tube vibrato one even does stereo You can approximate this sound by recording guitar direct or starting with a clean miked sound applying Voce Chorus Vibrato then using SansAmp PSA 1 or Eleven Voce Spin Voce Spin provides the most accurate simulation of the well loved rotating speaker 15 classic record ing setups feature horn resonance speaker cross over varying microphone placement even the Memphis sound with the lower drum s slow mo tor unplugged Don Leslie invented the rotating speaker in 1937 His design is simple and elegant an internal 40
363. original input sig nal Activate the Phase switch and observe the change in the sound characteristics of the Big Bottom Pro effect For most applications leave the Phase switch in the Off position Activate the AutoTrace switch and observe the change in the sound characteristics Also notice that the compression level in the dynamic pro cessor shown by the Comp meter is affected as well Readjust the Mix control as desired to experi ence the benefits of the Big Bottom Pro plug in Remember that a little Big Bottom Pro effect goes a long way Chapter 44 Eleven 255 Chapter 44 Eleven Eleven is a guitar amplifier plug in that is available in DSP Native and AudioSuite formats Eleven gives you stunning guitar amplifier cabinet and microphone models of the best of the best vintage and con temporary gear Eleven Free is a free version of Eleven that comes with every Pro Tools system with a reduced feature set Eleven Free comes in Native and AudioSuite formats only Eleven Plug in Chapter 44 Eleven 256 Eleven Free Plug In Features Two custom amp models from Avid Two speaker cabinet models Amps and cabinets can be mixed and matched Noise Gate to control any unwanted noise Settings files presets to store and recall factory and custom tones Support of any compatible work surface or MIDI controller MIDI Learn provides effortless map ping to any continuous contro
364. ors to find one that works best for the material Don t be afraid to use extreme compression ratios to achieve this effect EQ Simple EQ processing can be used to soften hot spots in the playing range of some guitars Using any of the included EQ plug ins you can also try applying drastic shelving or band limiting as a spe cial effect or automate a filter sweep to simulate a wah style effect Echo and Delay To add echo to the guitar track bus an Eleven track to an Aux Input and put a Delay plug in on the Aux Try other delay plug ins to unlock the secrets of multi tap ping pong and other specialized appli cations Chapter 44 Eleven 280 Eleven Signal Flow Notes The following figure shows the signal flow through Eleven from its input source to its output destination Plug Ins are Pre Fader Keep in mind that inserts plug ins in Pro Tools are post disk live input but pre fader The track fader does not affect the signal into any plug ins in serted on that same track This is the same for all Pro Tools inserts not just Eleven Input LED before the Input Knob The Input LED is before the Input section of the Master section which is prior to the first input stage of each amp This lets you determine whether you re clipping a signal before it enters the Eleven signal chain The Input LEDs will light red when the signal has clipped the input If this occurs in sert the Trim plug in before Eleven and use i
365. ossible to achieve a substantial increase in bass energy with out significantly increasing the peak level output For optimal performance keep the peak hold meter of the Drive meter inside the yellow area Optimizing Big Bottom Pro Effects When using Big Bottom Pro the output signal level is equal to the input signal levels plus the bass en hanced Big Bottom Pro effect The dynamic char acteristics of Big Bottom Pro are based on a com plex algorithm that includes the signal peak level the average steady staid level as well as the dy namic characteristics Unlike a bass EQ which adds a constant boost in the low end Big Bottom Pro enhancement is added into the input signal dy namically Starting with the factory settings experiment with the controls on Big Bottom Pro to hear how this plug in effects the low end frequencies of your source material as follows If the Drive meter is clipping in the yellow area adjust the Drive control for optimal operation Activate the Solo switch to listen to only the Big Bottom Pro side chain effect Vary the Tune control to hear the low pass filter isolate the low end bandwidths of the original in put signal De activate the Solo switch and continue to vary the Tune control until you find the optimal set ting Adjust the Mix control to set the amount of Big Bottom Pro effect Use the In Out switch for an A B comparison with the output signal and the
366. ou are applying to the incoming audio signal Dynamics Graph Gain Reduction Resolution Channel Strip lets you view the gain reduction scale on the Dynamics Graph display either in 3 dB increments from 0 dB to 18 dB or in 6 dB incre ments from 0 dB to 36 dB To change the Dynamics Graph Gain Reduction resolution Click the Graph Resolution toggle Using the Dynamics Graph to Adjust Controls You can drag in the Dynamics Graph display to ad just the corresponding Expander Gate and Com pressor Limiter controls The cursor updates to show which control is being adjusted Expander Gate Ratio Expander Gate Knee Expander Gate Threshold Gate Depth Hysteresis Compressor Limiter Ratio Compressor Limiter Knee Compressor Limiter Threshold Limiter Depth Dynamics graph display Input signal level x axis Expander Gate Threshold Compressor Limiter Threshold Output signal level y axis Graph Resolution toggle For the Expander Gate and Compressor Lim iter effects adjusting a control in the Dynam ics Graph display automatically shows the pane that includes the adjusted control if it is not already shown except when the All tab is shown Chapter 11 Channel Strip 51 Expander Gate Controls Threshold The Threshold Thresh control sets the level be low which an input signal must fall to trigger ex pansion or gating Signals that fall below the thr
367. ow initial buildup This control interacts with the Size and Decay controls to affect the overall reverb density High settings of diffusion can be used to enhance percussion Use low or moderate settings for clearer and more natural sounding vocals and mixes Decay Control Decay controls the rate at which the reverb decays after the original direct signal stops The value of the Decay setting is affected by the Size and Algo rithm controls This control can be set to infinity on most algorithms for infinite reverb times Pre Delay Control Pre Delay determines the amount of time that elapses between the original audio event and the onset of reverberation Under natural conditions the amount of pre delay depends on the size and construction of the acoustic space and the relative position of the sound source and the listener Pre Delay attempts to duplicate this phenomenon and is used to create a sense of distance and volume within an acoustic space Long Pre Delay settings place the reverberant field behind rather than on top of the original audio signal Hi Frequency Cut Hi Frequency Cut controls the decay characteristic of the high frequency components of the reverb It acts in conjunction with the Low Pass Filter control to create the overall high frequency contour of the reverb When set relatively low high frequencies decay more quickly than low frequencies simulat ing the effect of air absorption in a hall The maxi mum
368. ower Q values represent wider bandwidths With the low band EQ set to Shelf the Q control changes the Q of the shelving filter Higher Q val ues represent steeper shelving curves Lower Q val ues represent broader shelving curves You can press the Shift key while clicking and dragging an EQ control point up or down to adjust the Gain setting without changing the Frequency Likewise press the Shift key while clicking and dragging an EQ control point left or right to adjust the Frequency set ting without changing the Gain setting Option Shift click Mac or Alt Shift click Windows an EQ control point to invert its Gain setting You can also Control click Mac or Start click Windows and drag a control point up or down to increase or decrease the Q setting EQ Filters section Low Frequency tab Chapter 11 Channel Strip 57 Low Mid Frequency EQ Controls The LMF tab provides controls for the low mid fre quency band of the EQ This band is a peak EQ Frequency The Frequency control lets you set the center fre quency for the peak low mid frequency band Gain The Gain control lets you boost or attenuate the corresponding frequencies for the low mid fre quency band Q The Q control changes the width of the low mid peak EQ band Higher Q values represent narrower bandwidths Lower Q values represent wider band widths High Mid Frequency EQ Controls The HMF tab provides controls for the high mid fre
369. owever the default of 25 milliseconds should work well for most material X Form Pitch Section Controls The Pitch section provides controls for pitch shift ing the selected audio Use the Pitch control to transpose the pitch from as much as 36 00 semi tones 3 octaves to 36 00 semitones 3 oc taves with fine resolution in cents depending on which Range button is enabled 2x 4x or 8x X Form also lets you transpose the formant shape in dependently of the fundamental frequency Transpose The Transpose setting displays the transposition amount in semitones You can transpose pitch by as much as 36 00 semitones 3 octaves to 36 00 semitones 3 octaves with fine resolution in cents depending on which Range button is en abled Adjust the Pitch control or click the Trans pose field and type a value Shift The Shift setting displays the pitch shift amount as a percentage Pitch can be shifted by as much as 12 50 3 octaves to 800 00 3 octaves de pending on which Range button is enabled 2x 4x or 8x Adjust the Pitch control or click the Shift field and type a value Formant Audio with a fundamental pitch has an overtone se ries or set of higher harmonics The strength of these higher harmonics creates a formant shape which is apparent if viewed using a spectrum ana lyzer The overtone series or harmonics have the same spacing related to the pitch and have the same general shape rega
370. perates as a fast attack compressor with a high compression ratio It does not attack in stantaneously or look ahead in order to attack ahead of time but instead uses a very fast 1 millisecond attack time As such the D3 is not a brick wall limiter but limits the overall dynamic range of sig nals in a sonically pleasing way Like the Compressor the Limiter is activated when the signal exceeds the user selected threshold The Limiter then compresses any signal above the se lected threshold down to the threshold limit that you have set To enable the limiter 1 Disable the compressor by Control clicking Mac or Start clicking Windows its icon 2 Click the Limiter icon D3 Side Chain Processing Compressors and limiters generally use the de tected amplitude of the input signal as a control source Other signals can also be used as a control source by using a key input With de essing for ex ample a frequency modified version of the input signal is used as a trigger This is known as side chain processing Side chain processing allows the D3 compression or limiting to be controlled by another independent audio signal In this way you can compress or limit one track s audio using the dynamics of a different track s audio Using D3 in Stereo In stereo configurations all D3 controls except the Input Level affect both channels of the stereo sig nal The D3 RMS detector which derives the con trol signal th
371. ping Mode The Audio Mode pop up menu determines the fol lowing types of TCE and pitch shift algorithm for processing audio Monophonic Select Monophonic for processing monophonic sounds such as a vocal melody Polyphonic Select Polyphonic for processing com plex sounds such as a multipart musical selection Rhythmic Select Rhythmic for processing percus sive sounds such as a mix or drum loop Rhythmic mode uses transient analysis for time shifting If you select audio with no apparent tran sients or set the Transient Threshold control to a setting above any detected transients Time Shift assumes a virtual transient every three seconds to be able to process the file Consequently the file should be 20 bpm or higher one beat every three seconds to achieve desirable results For material that has no apparent transients use Monophonic or Polyphonic mode Varispeed Select Varispeed to link time and pitch change for tape like pitch and speed change effects and post production workflows Range The Audio Range pop up menu determines the fol lowing frequency ranges for analysis Low For low range material such as a bass guitar select Low Mid For mid range material such as male vocals select Mid In Monophonic mode Mid is the de fault setting and is usually matches the range of most monophonic material High For material with a high fundamental fre quency such as female vocals select high Wi
372. plate of sheet metal suspended by springs inside a soundproof en closure An adjustable damping plate allows con trol of the reverb decay time and piezoelectric pick ups attached to the plate provide the return reverb signal to the console An alternative and less ex pensive analog reverb system is the spring reverb most commonly seen in guitar amplifiers beginning in the 1960s Similar to the plate reverb in opera tion the spring reverb uses a transducer to feed the signal into a coiled steel spring and create vibra tions These are then captured via a pickup and fed back into an amplifier Since the advent of digital audio technology in the 1980s artificial reverberation has been created pri marily by digital algorithms that crudely mimic the physics of natural reverb spaces by using multiple delay lines with feedback Digital synthetic re verb units offer a new level of realism and control unavailable with older analog reverb systems but still fall short of the actual reverb created by a real space Components of Reverb Reverberation sound in a normal space usually has several components For example the sound of a single hand clap in a large cathedral will have the following distinct parts The direct sound of the hand clap is heard first as it travels from the hand directly to the ear which is the shortest path After the direct sound the first component of reverb heard by a listener is reflected sound from the
373. played here Once the audio selection is displayed you can load replacement samples and adjust their trigger thresholds while viewing the waveform peaks Trigger markers then appear in the waveform indi cating the points at which the samples will be trig gered The color of each marker indicates which thresh old replacement sample will be triggered The blue Trigger Envelope shows the waveform slope that determines the trigger points The Zoomer lets you increase or decrease waveform magnification here to help accurately set trigger thresholds If you change the audio selection on the source track SoundReplacer automatically updates the waveform display each time you make a new selec tion or begin playback Waveform display with trigger markers shown Chapter 58 SoundReplacer 337 SoundReplacerTrigger Threshold The color coded Trigger Threshold sliders set a to tal of three amplitude zones one for each replace ment audio file for triggering replacement sam ples The yellow slider represents amplitude zone 1 the lowest level trigger The red slider represents amplitude zone 2 the middle level trigger The blue slider represents amplitude zone 3 the highest level trigger With a replacement sample loaded drag the Threshold slider to set the amplitude level Color coded trigger markers will appear in the Waveform at points where the source audio signal exceeds the threshold set for that amplitu
374. plug in that is available in DSP Native and AudioSuite formats Signal Generator supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Signal Generator Controls The Signal Generator plug in provides the following controls Frequency Sets the frequency of the signal in hertz Values range from a low of 20 Hz to a high of 20 kHz in a 44 1 kHz session The upper limit of the frequency range for this setting will increase to match the Nyquist frequency half the sample rate in 96 kHz and 192 kHz sessions HD series sys tems only Level Sets the amplitude of the signal in decibels Values range from a low of 95 dB to a high of 0 0 dB Signal These buttons select the waveform Choices are sine square sawtooth triangle white noise and pink noise Peak Generates signal at the maximum possible level without clipping RMS Generates signal at levels consistent with the RMS Root Mean Square value or the effective average level of the signal Signal Generator Refer to the guide for your audio interface for instructions on using Signal Generator to cal ibrate that interface Signal Generator produces a tone as soon as it is inserted on a track To mute the Signal Generator use the Bypass button The Signal Generator plug in is not intended for rigorous test purposes it is a simple level calibration tool Chapter 57 Signal Generator 334 AudioSuite Processing with Signal Genera
375. pole or the entire filter 4 pole 2 pole is brighter while 4 pole has a deeper mellow quality Drive The Drive control sets the input gain Use it to adjust the input to the filter and envelope fol lower LED Indicators Three LEDs down the center of the unit provide vi sual feedback Level Level glows green when signal is present to the envelope circuit Env Env envelope glows redder in response to the envelope tracking of the input Bypass Bypass glows either red bypassed or green not bypassed to show whether or not the ef fect is in the signal path Moogerfooger Lowpass Filter Tips and Tricks Try inserting an LFO ahead of the Moogerfooger Lowpass Filter to produce a cool auto wah effect Use this in conjunction with Voce Spin s rotating speaker for a swirling auto wah effect Chapter 37 Moogerfooger 12 Stage Phaser 222 Chapter 37 Moogerfooger 12 Stage Phaser Moogerfooger 12 Stage Phaser combines a 6 or 12 stage phaser with a wide ranging variable LFO Start with subtle tremolo or radical modulation ef fects then crank the distortion and resonant filters for unbelievable new tones all featuring classic MOOG sound Moogerfooger 12 Stage Phaser is available in DSP Native and AudioSuite formats Moogerfooger 12 Stage Phaser supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Moogerfooger 12 Stage Phaser operates as a mono multi mono or stereo plug in
376. ponding to the con trol that you want to adjust 2 Do any of the following Type a new value For controls that support val ues in kilohertz typing k after a numeric value will multiply the value by 1000 To increment the value scroll up with a mouse or scroll wheel or press the Up Arrow key To decrement the value scroll down with a mouse or scroll wheel or press the Down Arrow key 3 Do one of the following Press Enter on the numeric keyboard to input the value and remain in keyboard editing mode Press Return Mac or Enter Windows on the alpha keyboard to enter the value and leave key board editing mode To move forward through control text boxes in a plug in Press the Tab key To move backward through control text boxes in a plug in Press Shift Tab Dragging in Graphic Displays Some plug ins have graphic displays with control points that you can drag to adjust the corresponding controls Adjusting Controls with Fine Resolution Controls and control points can be adjusted with fine resolution by holding the Command key Mac or the Control key Windows while adjusting the control Resetting Controls to Default Values You can reset any on screen control to its default value by Option clicking Mac or Alt clicking Windows directly on the control or on its corresponding text box Typing a control value EQ III Dragging a control point EQ III
377. portion of the reverb To clear the Clip LED click it Chapter 29 ReVibe II 167 Chapter 29 ReVibe II ReVibe II is a studio quality reverb and acoustic environment modeling plug in available in DSP Native and AudioSuite formats Using ReVibe II ReVibe II makes it possible to model extremely re alistic acoustic spaces and place audio elements within a mix ReVibe II supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates ReVibe II works with mono and stereo formats and LCR LCRS quad 5 0 and 5 1 greater than stereo multichannel formats In general when working with stereo and greater than stereo tracks use the multichannel version of ReVibe II ReVibe II Greater than stereo formats are only available with Pro Tools HD Chapter 29 ReVibe II 168 ReVibe II supports the following combinations of track types and plug in insert formats Dragging in the Graphic Display to Adjust Controls In addition to dragging controls and typing control values you can adjust settings on the Decay Color amp EQ graphic displays by dragging control points on the graph To cut or boost a particular EQ band Drag a control point up or down To adjust EQ frequency crossover Drag the control point right or left To adjust high frequency rear cut Drag the control point right or left Track Type Plug in Insert Format Mono Stereo L C R L C R S Quad 5 0 5 1 Mo
378. pports tempo values from 30 300 bpm When slaved to a ReWire client application Pro Tools playback will be re stricted to this range even if the client appli cation s tempo is outside this range Addi tionally some ReWire client applications such as Reason may misinterpret Pro Tools meter changes resulting in mismatched lo cate points and other unexpected behavior To prevent this avoid using meter changes in Pro Tools when using Reason as a ReWire client Chapter 54 ReWire 317 Looping Playback with ReWire Because Pro Tools does not offer separate loop markers as found in other third party applications such as Reason if you want to loop playback do one of the following To loop playback in Pro Tools 1 In the Pro Tools Timeline select the time range that you want to loop 2 Begin playback by pressing the Spacebar or clicking the Play button in the Transport To loop playback within a ReWire client sequencer With playback stopped specify the loop within the ReWire client application and begin play back Automating Input Switching with ReWire ReWire supports automation for switching inputs during playback To automate switching inputs during playback 1 Set the track s automation to write 2 Do one of the following Change the input link pop up menu manually Draw the automation in the Edit window If you create a playback loop by making a se lection in the Pro Tools Timel
379. process as well as making it easier for engineers to generate test tones for musicians to tune with InTune Chapter 55 InTune 320 InTune Controls and Displays InTune Auto Button Click the Auto button to toggle Automatic Mode on and off When Automatic mode is active InTune will detect the note played and automatically show the pitch for that note To enable Automatic mode Click the Auto button to enable Automatic mode The Auto button highlights To tune to a single note and turn off Automatic mode Click the button for a note This turns off automatic mode InTune will now display pitch relative to the selected note only InTune Test Tone Menu Selector InTune will generate both sine wave and triangle wave test tones as shown in the tone menu The Audible tuning tone modulates the input signal against the reference tone To hear a test tone 1 Select Sine Triangle or Audible from the Test Tone selector 2 Click the Note button for a note 3 Adjust the Tone Volume slider When a test tone is playing Tone Playing appears in the information display InTune Edit Button Clicking the Edit button displays the Tuner Pro gramming screen where you can create custom ized tuning presets that display note selections for specific instruments and tunings See Creating In Tune Tuning Presets on page 322 Selecting a note Selecting a test tone Chapter 55 InTune 321 InT
380. processing letting you make up gain or prevent clipping on the channel where the Pro Compression plug in is being used To adjust the output level do one of the following Click in the Output Level field and type a value INF dB to 12 dB Click the Output Level control and drag up or down to adjust the Output Level setting Output Meters The Output meters show peak signal levels after processing Attenuation Meters The Output meter can be switched to show Attenu ation metering for the processed signal from 0 dB to 36 dB To toggle between the Attenuation and Output meters Click the Output Attenuation toggle in the top right hand corner of the Output section Output section with Meters and Output Level control Output Attenuation toggle Attenuation meters shown Chapter 18 Pro Compressor 100 Pro Compressor Dynamics Graph The Dynamics Graph display shows a curve that represents the level of the input signal on the hor izontal x axis and the amount of gain reduction applied on the vertical y axis The display shows a vertical line representing the Threshold setting for the Compressor The Dynamics Graph display also features an ani mated red ball in the gain transfer curve display This ball shows the amount of input gain x axis and gain reduction y axis being applied to the in coming signal at any given moment To indicate overshoots when an incoming signal peak is too fa
381. produce in finite reverberation Level Control Level controls the output level of the reverb tail When set to INF minus infinity no reverb tail is heard and the reverb effect consists entirely of the early reflections if enabled The range of this con trol is from INF to 6 0 dB See ReVibe II Room Types on page 177 for a list of room presets When specifying reverb size keep in mind that the size of a reverberant space in meters is approximately equal to its longest dimen sion In general halls range from 25 m to 50 m large to medium rooms range from 15 m to 30 m and small rooms range from 5 m to 20 m Similarly a Room Size setting of 20m corresponds roughly to a 4x8 plate Chapter 29 ReVibe II 171 Diffusion Control Diffusion controls the rate that the sound density of the reverb tail increases over time The control ranges between 50 and 50 At 0 diffusion is set to an optimal preset value Positive Diffusion settings create a longer initial buildup of echo den sity At negative settings the buildup of echo den sity is slower than at the optimal preset value Spread Control Spread controls the rate at which reverberation builds up Spread works in conjunction with the At tack Shape control to determine the initial contour and overall ambience of the reverberation enve lope At low Spread settings there is a rapid onset of re verb at the beginning of the reverberation envelope Hig
382. put With side chain filters you can make dynamics processing more or less sensitive to certain fre quencies For example you might configure the side chain so that certain lower frequencies on a drum track trigger dynamics processing Source The Source selector lets you set the source for side chain processing Int Stereo Pairs Ext All w LFE Int All no LFE or Int Front Rear Internal Source Stereo Pairs When Int StereoPairs is selected processing is triggered for both the Left and Right channel when the input signal reaches the threshold on either the Left or the Right channel With greater than stereo multichannel processing the input signal for each stereo pair affects only those same channels and likewise mono channels are affected only by their own input signal For example with an LCR multi channel format the processing for the Center chan nel is only triggered when the Center channel input signal reaches the threshold However when the input signal reaches the threshold on the Left or the Right channel processing is triggered for both the Left and the Right channel External All with LFE When Ext All w LFE is selected the plug in uses the amplitude of a separate reference track or exter nal audio source to trigger dynamics processing Dynamics processing is applied equally to all chan nels when the input signal reaches the threshold on any input channel With greater than stereo multi channel p
383. put Dial in a highpass frequency from 50 Hz to 500 Hz Frequencies below the setting are filtered from the feedback loop HPF On Off The HPF Off HPF On enables or dis ables the highpass filter HPF Drive The Drive control sets the input gain Mix The Mix control blends the original input sig nal with the delayed signal LED Indicators Three LEDs down the center of the unit provide vi sual feedback Input Level The Input Level LED glows green when signal is present HPF The HPF LED turns green when the highpass filter is enabled Bypass The Bypass LED glows either red by passed or green not bypassed to show whether or not the effect is in the signal path Moogerfooger Analog Delay Tips and Tricks Infidelity Because analog delay chips offer only a fixed num ber of cells the extended delay times store a lower fidelity version of the input signal Try the Long delay setting when going for cool lo fi sounds and textures Echo Swarms By carefully adjusting the Feedback Drive and Highpass controls you can use the Moogerfooger Analog Delay as a sound generator Simply pulse the delay unit with a short piece of audio even a second will do and adjust the Delay Time knob Set correctly the unit will generate cool timbres for hours all by itself Chapter 33 Reel Tape Delay 209 Chapter 33 Reel Tape Delay Reel Tape Delay is part of the Reel Tape suite of tape simulation effects plug ins Ree
384. quency band of the EQ This band is a peak EQ Frequency The Frequency control lets you set the center fre quency for the peak high mid frequency band Gain The Gain control lets you boost or attenuate the corresponding frequencies for the high mid fre quency band Q The Q control changes the width of the high mid peak EQ band Higher Q values represent narrower bandwidths Lower Q values represent wider band widths EQ Filters section Low Mid Frequency tab EQ Filters section High Mid Frequency tab Chapter 11 Channel Strip 58 High Frequency EQ Controls The High Frequency EQ tab provides controls for the high frequency band of the EQ Filter Type The High Frequency band can be set to be a Peak or High Shelf EQ Frequency The Frequency control lets you set the center fre quency for the high frequency band Peak or Shelf EQ Gain The Gain control lets you boost or attenuate the corresponding frequencies for the high frequency band Q With the high band EQ set to Peak the Q control changes the width of the EQ band Higher Q values represent narrower bandwidths Lower Q values represent wider bandwidths With the high band EQ set to Shelf the Q control changes the Q of the shelving filter Higher Q val ues represent steeper shelving curves Lower Q val ues represent broader shelving curves Filter 1 and Filter 2 Controls The Filter 1 and Filter 2 tabs provide the same set of controls fo
385. r The input Drive peak meter indicates the actual peak level to the Big Bottom Pro side chain Compression Meter The Compression Comp meter indicates the ac tual amount of compression taking place in the Big Bottom Pro side chain If the Comp meter is not showing any activity the input level is too low Ad just the Level and Drive controls accordingly Out Meter The Output peak meter indicates the actual peak level after mixing the Big Bottom Pro side chain with the original input signal Block Diagram of Aphex Big Bottom Pro BLOCK DIAGRAM OF BIG BOTTOM PRO INPUT Attenuation Mix 12 dB Tune 40 to 400 Hz AUTO TRACE On Off Phase On Off Programmed Gain Buffer Solo In Out Level 0 dB Drive 12 dB MAIN AUDIO PATH BIG BOTTOM SIDE CHAIN Summing System Bypass closed System Bypass open Attenuation Dynamic Compressor OUTPUT Low Pass Filter Audition the loudest or peak sections of your audio material to avoid Big Bottom Pro output clipping Use the Out Meter to check for clipping Chapter 43 Aphex Big Bottom Pro 252 Rotary Controls Level Control The Level control sets the attenuation of the input signal For normal operation set the Level control on Max 0 dB In the event you are not generating enough bass en hancement even when the Mix control is also set on Max lower the Level control This will give the plug in more headroom
386. r 12 Stage Phaser Moogerfooger Lowpass Filter Chapter 1 Audio Plug Ins Overview 4 Pro Compressor Pro Expander Pro Limiter Purple Audio MC77 Reel Tape plug ins Reel Tape Saturation Reel Tape Delay Reel Tape Flanger Reverb One ReVibe II Smack SoundReplacer Space Tel Ray Variable Delay Voce Spin Voce Chorus Vibrato X Form Using Plug Ins in Pro Tools See the Pro Tools Reference Guide for information on working with plug ins including Inserting plug ins on tracks Plug In Window controls Adjusting plug in controls Automating plug ins Using side chain inputs Using plug in presets Clip indicators Chapter 1 Audio Plug Ins Overview 5 Conventions Used in Pro Tools Documentation Pro Tools documentation uses the following conventions to indicate menu choices keyboard commands and mouse commands The names of Commands Options and Settings that appear on screen are in a different font The following symbols are used to highlight important information System Requirements and Compatibility for Plug Ins To use Pro Tools plug ins you need the following An Avid qualified system running Pro Tools or Pro Tools HD Software An iLok USB key iLok for plug ins that can be purchased or rented Avid can only assure compatibility and provide s
387. r each filter Filter Type Both Filter 1 and Filter 2 can be set independently Select from the following Filter Type options High Pass Low Pass Band Pass and Notch Frequency The Frequency control lets you set the center fre quency for the selected Filter Type from 20 Hz to 21 0 kHz Slope When the Filter Type is set to Low Pass or High Pass the Slope control is available The Slope con trol lets you set the slope for the filter from the se lected Frequency to INF 12 dB O or 24 dB O Q When the Filter Type is set to Band Pass or Notch the Q control is available The Q control changes the width of the filter around the center frequency band Higher Q values represent narrower band widths Lower Q values represent wider band widths EQ Filters section High Frequency tab EQ Filters section Filter 1 tab shown Chapter 12 Dynamics III 59 Chapter 12 Dynamics III Dynamics III is a suite of three dynamics plug ins available in DSP Native and AudioSuite formats Compressor Limiter see Compressor Limiter III on page 62 Expander Gate see Expander Gate III on page 65 De Esser see De Esser III on page 68 Dynamics III supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates The Compressor Limiter and Expander Gate mod ules support mono stereo and greater than stereo multichannel formats up to 7 1 The De Esser module supports mono
388. r is patented in the United States Japan and most of Europe Others may claim they are doing the same thing but they can only resort to some form of EQ amplitude correction or expan sion phase scrambling and or filtering They can only increase peak levels causing clipping feed back tape distortion and listener fatigue Aural Exciter is available in DSP Native and Au dioSuite formats Aural Exciter supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Aural Exciter operates as a mono multi mono or stereo plug in Block Diagram of Aphex Aural Exciter Type III BLOCK DIAGRAM OF AURAL EXCITER TYPE III INPUT Attenuation Mix 6 dB 18 dB Peaking Programmed Gain Buffer Solo Aural Exciter Level 0 dB SPR On Off Drive 6 dB 18 dB MAIN AUDIO PATH AURAL EXCITER SIDE CHAIN Summing Bypass closed Bypass open Tune Null FIll Attenuation Dynamic Compressor OUTPUT Spectral Phrase Refracter State Variable Filter Harmonics Harmonics Generator Density Timbre Waveform Generator Chapter 42 Aphex Aural Exciter Type III 243 Meters Drive Meter The Drive meter monitors the peak level to the har monic generator It works in conjunction with the Drive switch For optimal performance keep the peak hold meter of the Drive meter inside the yellow area The harder you drive the Exciter the more Exciter e
389. r stereo plug in Reverb One features include Editable Reverb EQ graph Editable Reverb Color graph Reverb Contour graph Dynamic control of reverb decay Chorusing Early reflection presets Extensive library of reverb presets About Reverb Digital reverberation processing can simulate the complex natural reflections and echoes that occur after a sound has been produced imparting a sense of space and depth the signature of an acoustic environment When you use a reverberation plug in such as Reverb One you are artificially creating a sound space with a specific acoustic character This character can be melded with audio material with the end result being an adjustable mix of the original dry source and the reverberant wet signal Reverberation can take relatively lifeless mono source material and create a stereo acoustic envi ronment that gives the source a perceived weight and depth in a mix Creating Unique Sounds In addition digital signal processing can be used creatively to produce reverberation characteristics that do not exist in nature There are no rules that need to be followed to produce interesting treat ments Experimentation can often produce striking new sounds Acoustic Environments When you hear live sound in an acoustic environ ment you generally hear much more than just the direct sound from the source In fact sound in an anechoic chamber devoid of an acou
390. ray the External Key is disabled and the plug in uses the amplitude of the input signal to trigger dynam ics processing Compressor Limiter and Expander Gate Side Chain External Key button Chapter 12 Dynamics III 71 Dynamics III Side Chain Listen When enabled this control lets you listen to the in ternal or external side chain input by itself as well as monitor its levels with the Output meter This is especially useful for fine tuning the plug in s filter settings or external key input Dynamics III Side Chain HF and LF Filter Enable Buttons The HF Filter Enable and LF Filter Enable buttons toggle the corresponding filter in or out of the side chain When this button is highlighted the filter is applied to the side chain signal When this button is dark gray the filter is bypassed and available for activation Dynamics III Side Chain High Frequency HF Filter Type The HF filter section lets you filter higher frequen cies out of the side chain signal so that only certain bands of high frequencies or lower frequencies pass through to trigger dynamics processing The HF side chain filter is switchable between Band Pass and Low Pass filters Band Pass Filter Makes triggering of dynamics processing more sensitive to frequencies within the narrow band centered around the Frequency set ting and rolling off at a slope of 12 dB per octave Low Pass Filter Makes triggering of dynamics pro cessing more sensitive t
391. rb bal ance until you get the distant sound you re looking for Spin into Moogerfooger Lowpass Filter Try using the amplitude modulation effects of Spin as an LFO driving the Moogerfooger Lowpass Fil ter Distortion and Spin To simulate overdriving the tube amp powering the rotating speaker apply distortion before Spin since in the real world signal path the amp distorts the signal before the speakers throw the sound around Among tons of other great distortion sounds the SansAmp PSA 1 plug in provides dis tortion presets for both the model 122 and model 147 rotating speakers Organ Signal Path Likewise when going for classic organ sounds route through the Voce Chorus Vibrato before Spin as that s the signal path in the B 3 organ The John Lennon Vocal Thing In what seems like a particularly dangerous Beatles studio experiment a Leslie speaker cabinet was dismembered a microphone was affixed to the rap idly spinning upper rotor and John Lennon at tempted to sing into it Fortunately the deafening wind noise captured by the microphone put a stop to the proceedings before anyone was hurt Feel free just to run the vocal through the rotating speaker that s what they wound up doing Reverse Spin Those reverse vocal and reverse guitar tricks are even more fun when you run em through Spin Try reversing the vocal and putting it through Spin as well as putting the vocal through Spin then revers ing t
392. rdless of what the fundamental pitch is It is this formant shape that gives the audio its overall characteristic sound or timbre When pitch shifting audio the formant shape is shifted with the rest of the material which can result in an unnatural sound Keeping this shape constant is critical to formant correct pitch shifting and achiev ing a natural sounding result The Pitch section of X Form lets you pitch shift the formants of the selected audio independently of the fundamental frequency This is useful for achiev ing formant correct pitch shifting It can also be used as an effect For example you can formant shift a male vocal up by five semitones and it will take on the characteristics of a female voice To enable or disable formant shifting Click the In button The In button lights when formant shifting is enabled The Formant field displays the amount of formant pitch shifting from 36 00 semitones 3 octaves to 36 00 semitones 3 octaves with fine resolu tion in cents Adjust the Formant control or click the Formant field and type a value X Form Pitch section Chapter 26 X Form 151 X Form AudioSuite Input Modes X Form supports the Pro Tools AudioSuite Input Mode selector for use on mono or multi input pro cessing Mono Mode Processes each audio clip as a mono file with no phase coherency maintained with any other simultaneously selected clips Multi Input Mode Processes up to 48 input ch
393. re client appli cation You can also edit the MIDI CC data in Pro Tools until you achieve the best results To play back MIDI CC data over ReWire 1 From the MIDI track s MIDI Output selector select the ReWire client application device you want to control the same device from which you recorded the MIDI CC data 2 Start playback in Pro Tools 3 Switch to the ReWire client application Notice that the corresponding control changes accord ing to the MIDI CC data from Pro Tools Quitting ReWire Client Applications When quitting Pro Tools sessions that integrate Re Wire client applications quit the client application first then quit Pro Tools Session Tempo and Meter Changes and ReWire Pro Tools transmits both Tempo and Meter data to ReWire client applications allowing ReWire com patible sequencers to follow any tempo and meter changes in a Pro Tools session With the Pro Tools Conductor button selected Pro Tools always acts as the Tempo master using the tempo map defined in its Tempo Ruler With the Pro Tools Conductor button deselected the ReWire client acts as the Tempo master In both cases playback can be started or stopped in either application If you quit Pro Tools before quitting ReWire client applications a warning dialog may ap pear stating that one or more ReWire appli cations did not terminate To avoid this quit all ReWire client applications before quitting Pro Tools Pro Tools su
394. reatively controlling the character of the re verb in a mix Reverse Reverses the IR waveform and controls the total length As the IR waveform is recalcu lated it is re displayed in the Waveform display The value shown is measured in Beats Per Minute to let you easily match the tempo of the music When loading an IR from an audio file Space relies on the user to define which part of the IR is the early portion of the waveform If the Early Length is set to zero controls in the Early group will not affect the IR Reverb controls Chapter 30 Space 201 Space Decay Section Controls The Decay group controls allow the user to control the decay of the low mid and high frequency por tions of the IR Use the controls to tailor the re verb s character for a mix or for creative possibili ties not found in traditional reverb processors Low Decreases or increases the rate at which low frequencies decay Low Xover Adjusts the frequency point that divides the IR into low and mid frequency portions Mid Decreases or increases the rate at which mid frequencies decay High Xover Adjusts the frequency point that di vides the IR into mid and high frequency portions High Decreases or increases the rate at which high frequencies decay Front Rear In quad and 5 0 channel output modes Front and Rear independently control the decay for front and rear channels Using Space This section addresses some common scenarios in
395. reducing noisy low level signals than the typically abrupt cutoff of a gate For more information on the LFE channel refer to the Pro Tools Reference Guide Expander Gate III Chapter 12 Dynamics III 66 Expander Gate III Input Output Level Meters The Input and Output meters show peak signal lev els before and after dynamics processing See Dy namics III Levels Section on page 59 for more in formation Expander Gate III Dynamics Graph Display The Dynamics Graph display lets you visually see how much expansion or gating you are applying to your audio material See Dynamics III Graph Dis play on page 61 Expander Gate III Look Ahead Button Normally dynamics processing begins when the level of the input signal crosses the threshold When the Look Ahead button is enabled dynamics processing begins 2 milliseconds before the level of the input signal crosses the threshold The Look Ahead control is useful for avoiding the loss of transients that may have been otherwise cut off or trimmed in a signal Expander Gate III Threshold Control The Threshold Thresh control sets the level be low which an input signal must fall to trigger ex pansion or gating Signals that fall below the threshold will be reduced in gain Signals that are above it will be unaffected An orange arrow on the Input meter indicates the current threshold and can also be dragged up or down to adjust the threshold setting The
396. reen instructions to remove the plug in Chapter 3 Adjusting Plug In Controls 9 Chapter 3 Adjusting Plug In Controls You can adjust plug in controls by dragging on screen controls by editing control values or by dragging in graphic displays Dragging Plug In Controls Rotary Controls Some plug ins have rotary controls that can be ad justed by dragging over them horizontally or verti cally To adjust a rotary control 1 Click on the control 2 Do any of the following Drag up or to the right to increment the control Drag down or to the left to decrement the control Slider Controls Some plug ins have slider controls that can be ad justed by dragging horizontally Some sliders are bipolar meaning that their zero position is in the center of the slider s range Drag ging to the right of center yields a positive value and dragging to the left of center yields a negative value To adjust a slider control 1 Click on the control 2 Do any of the following Drag to the right to increment the control Drag to the left to decrement the control Adjusting a rotary control by dragging EQ III Adjusting a slider control by dragging ReVibe II Chapter 3 Adjusting Plug In Controls 10 Editing Control Values Some controls have text boxes that display the cur rent control value You can edit the control value directly To edit control values 1 Click in the text box corres
397. requency of the Freak Mod effect Resonator and Resonator Add a resonant fre quency tone to the audio signal This frequency is determined by the Effect Frequency control The difference between these two modules is that Res onator reverses the phase polarity of the effect producing a hollower sound than Resonator The Resonator can be used to produce metallic and flanging effects that emulate the sound of classic analog flangers Sci Fi Effect Amount Effect Amount controls the mix of the processed sound with the original signal The range of this control is from 0 100 Sci Fi Effect Frequency Effect Frequency controls the modulation fre quency of the ring modulator and resonators The frequency range is dependent on the effect type For Ring Mod the frequency range of this control is from 0 Hz to 22 05 kHz For Freak Mod the fre quency range is from 0 Hz to 22 05 kHz For Res onator the frequency range is from 344 to 11 025 kHz For Resonator the frequency range is from 172 Hz to 5 5 kHz You can also enter a frequency value using key board note entry Sci Fi Mod Type Controls The four Mod Type buttons determine the type of modulation applied to the frequency of the selected effect Depending on the type of modulation you select here the sliders below it will change to pro vide the appropriate type of modulation controls If the Mod Amount is set to 0 no dynamic modula tion is applied to the au
398. ressor Limiter tab Compressor Limiter Ratio set to LMTR Compressor Limiter Ratio set to a negative value Chapter 11 Channel Strip 53 Release The Release control sets the length of time it takes for the Compressor Limiter to be fully deactivated after the input signal drops below the threshold Release times should be set long enough that if sig nal levels repeatedly rise above the threshold the gain reduction recovers smoothly If the release time is too short the gain can rapidly fluctuate as the compressor repeatedly tries to recover from the gain reduction If the release time is too long a loud section of the audio material could cause gain re duction that continues through soft sections of pro gram material without recovering Knee The Knee control sets the rate at which the com pressor reaches full compression once the threshold has been exceeded As you increase this control it goes from applying hard knee compression to soft knee compres sion With hard knee compression compression be gins when the input signal exceeds the threshold This can sound abrupt and is ideal for limiting With soft knee compression gentle compression begins and increases gradually as the input signal approaches the threshold and reaches full com pression after exceeding the threshold This cre ates smoother compression Gain The Gain control lets you boost overall output gain to compensate
399. reverb contour and the rear channel early reflec tions on or off within the Contour display When the Rear button is illuminated the initial reverber ation envelope and early reflections for the rear channels are displayed When the Rear button is not illuminated they are not displayed ReVibe II Room Types ReVibe II comes with over 200 built in Room Type presets in 14 Room Type categories These Room Type presets contain complex early reflections and room coloration characteristics that define the sound of the space The Room Type categories and their presets are as follows Studios Large Natural Studio 1 Large Natural Studio 2 Large Live Room 1 Large Live Room 2 Large Dense Studio 1 Large Dense Studio 2 Medium Natural Studio 1 Medium Natural Studio 2 Medium Natural Studio 3 Medium Natural Studio 4 Medium Live Room 1 Medium Live Room 2 Medium Dense Studio 1 Medium Dense Studio 2 Small Natural Studio 1 Small Natural Studio 2 Small Natural Studio 3 Small Natural Studio 4 Small Natural Studio 5 Small Dense Studio 1 Small Dense Studio 2 Vocal Booth 1 Vocal Booth 2 Vocal Booth 3 Vocal Booth 4 Rooms Large Bright Room 1 Large Bright Room 2 Large Neutral Room 1 Large Neutral Room 2 Large Dark Room 1 Large Dark Room 2 Large Boomy Room
400. reverbs has been the heavy CPU processing requirement which can result in convolution re verbs with unacceptable latency Many early soft ware convolution reverbs did not offer adequate control over traditional reverb parameters such as Pre Delay EQ or decay time Space redefines reverb processing in Pro Tools by offering zero and low latency convolution with the full set of controls provided by traditional synthetic reverbs Impulse Response sample Chapter 30 Space 185 Space System Design Space uses advanced DSP algorithms to deliver convolution reverb processing The following figure shows the internal system design of Space and demonstrates how Space processes the audio signal The impulse computer is an internal module of Space that provides extensive control over the currently loaded impulse response waveform When you adjust the parameters shown below the IR is automatically recalculated by the impulse computer and reloaded into the convolution processor Space internal system design Audio In IR Level Impulse Response Low and High Convolution Processor Impulse Computer Shelf EQ Dry Level Balance Wet Level Out Level Audio Out Chapter 30 Space 186 The following figure shows the internal functions of the impulse computer as it processes the waveform and loads it into the convolution processor Space internal functions of the impulse computer Impulse Response Width Front Rear
401. rmance filenames should always be suffixed with type to avoid Space having to open the file to determine audio format Input Output Channel Order File Format Mono Mono Mono file Mono Stereo L R One 2 channel file or two mono files Mono Quad L R Ls Rs One 4 channel file or four mono files Mono 5 0 L C R Ls Rs One 5 channel file or five mono files Stereo Stereo L R Four mono files Stereo Quad L R Ls Rs Eight mono files Stereo 5 0 L C R Ls Rs Ten mono files Chapter 30 Space 189 The following examples show how various multi mono IR files could be named Stereo to Stereo IR Cathedral 1 L wav Cathedral 1 R wav Cathedral 2 L wav Cathedral 2 R wav Stereo to 5 0 IR Cathedral 1 L wav Cathedral 1 C wav Cathedral 1 R wav Cathedral 1 Ls wav Cathedral 1 Rs wav Cathedral 2 L wav Cathedral 2 C wav Cathedral 2 R wav Cathedral 2 Ls wav Cathedral 2 Rs wav Mono to Quad IR Cathedral L wav Cathedral R wav Cathedral Ls wav Cathedral Rs wav Stereo to Quad IR Cathedral 1 L wav Cathedral 1 R wav Cathedral 1 Ls wav Cathedral 1 Rs wav Cathedral 2 L wav Cathedral 2 R wav Cathedral 2 Ls wav Cathedral 2 Rs wav Channel Compatibility and Space Space works best with IRs that match your current channel configuration For example if Space is in stantiated in a mono to stereo configuration stereo IRs will be highlighted in the IR browser The IR
402. ro Limiter Chapter 20 Pro Limiter 118 Pro Limiter Metering Pro Limiter uses sample peak meters for Input and Output signals The Peak Hold value is displayed numerically at the top of the meter and the Peak Hold indicator ap pears as a thin orange line in the meter This pro vides highly accurate visual metering correlation with the audio signal Input and Output meters use the following color coding Dark Blue Indicates nominal levels from 90 dB to 20 dB Light Blue Indicates pre clipping levels from 20 dB to 0 dB Yellow Indicates full scale levels from 0 dB to 6 dB Gain Reduction meters are orange for the entire dy namic range displayed Pro Limiter Input Section The Input section provides input metering and con trols for adjusting the level of the input signal and the display of the Input meters Input Trim The Input Trim control sets the input gain of the plug in before processing letting you boost or at tenuate gain at the plug in input stage To trim the level of the input signal do one of the following Click in the Input Level field to type a value 30 0 dB to 30 0 dB Click the Input Trim control and drag up or down to adjust the Input Trim setting Input section with Meters Gain Reduction Peak Hold display Dim Input Meter toggle Input Sample Peak Hold display Threshold control Input Trim Chapter 20 Pro Limiter 119 Dim Input Meter Toggle
403. rocessed signal to avoid clipping Normalizing a selection before using X Form may produce better results X Form X Form Audio section Chapter 26 X Form 148 Type The Audio Type determines the type of TCE and pitch shift algorithm for processing audio Poly phonic Monophonic or Poly Faster Polyphonic Use for processing complex sounds such as a multipart musical selection Monophonic Use for processing monophonic sounds such as a vocal melody Poly Faster Use for faster previewing and pro cessing but with slightly reduced audio quality Gain The Gain control attenuates the input level to avoid clipping Adjust the Gain control from 0 0 dB to 6 0 dB to avoid clipping in the processed signal Clip Indicator The Clip indicator indicates clipping in the processed signal When using time compres sion or pitch shifts above the original pitch it is possible for clipping to occur The Clip indicator lights when the processed signal is clipping If the processed signal clips undo the AudioSuite pro cess and attenuate the input gain using the Gain control Then re process the selection Level Indicator The Level indicator displays the level of the output signal using a plasma LED which uses the full range of plasma level metering colors X Form Time Section Controls The Time section of X Form provides controls for specifying the amount of time compression or ex pansion as well as the timebase used
404. rocessing the key signal triggers dynam ics processing for all processed audio channels equally Side Chain section Selecting the Source for side chain processing Chapter 19 Pro Expander 115 The reference track used for side chain processing is selected using the Plug In Key Input selector in the Plug In window header Internal All No LFE When Int All no LFE is selected dynamics pro cessing is applied equally to all channels when the input signal reaches the threshold on any input channel except for the LFE channel if present The LFE channel is processed independently based on its own input signal Internal Front Rear For LCRS or greater channel formats when Int Front Rear is selected dynamics processing is applied based on front channel inputs LCR and surround channel inputs S independently For 1 formats the LFE channel is processed inde pendently based on its own input signal Side Chain Listen Mode Listen mode lets you hear the input signal for the side chain to the compressor This can be either the external key input or the internal side chain in cluding the applied filter To enable or disable Listen mode on the side chain Click the Listen button in the top right corner of the Side Chain section so that it is highlighted To disable it click the button again so that it is not highlighted Configuring a key input for side chain processing Side Chain Listen mode enable
405. rve for dynamic expansion and gating as well as a frequency graph for side chain filtering Additionally the dynamics graph can be used to graphically edit the Threshold Ratio Knee and Depth settings Pro Expander Metering Pro Expander provides combined meters that show both sample peak metering and averaged metering Pro Expander uses sample peak meters over aver age metering for Input and Output signals Attenu ation metering uses sample peak metering only The Peak Hold value is displayed numerically at the top of the meter and the Peak Hold indicator ap pears as a thin orange line in the meter This pro vides highly accurate visual metering correlation with the audio signal Pro Expander also displays averaging metering with an integration time of ap proximately 400 ms Input and Output meters use the following color coding Dark Blue Indicates nominal levels from 90 dB to 20 dB Light Blue Indicates pre clipping levels from 20 dB to 0 dB Yellow Indicates full scale levels from 0 dB to 6 dB Greater than stereo formats are only available with Pro Tools HD Pro Expander Chapter 19 Pro Expander 108 Attenuation meters show dark blue for the entire dynamic range displayed Peak Indicators The Input and Output meters provide graphical rep resentation of transient peaks as well as graphical and numerical display of the last greatest regis tered peak The Attenuation meter provides similar
406. s Section The Dynamics section of Channel Strip provides Expander Gate Compressor Limiter and Side Chain processing all in one This section also pro vides a dynamics graphic display for the Compres sor Limiter and Expander Gate plug ins The dis play shows a curve that represents the level of the input signal on the horizontal x axis and the amount of gain reduction applied on the vertical y axis The vertical line represents the threshold Showing the FX Chain Process Order FX Chain FILT bypassed Dynamics section All tab shown Chapter 11 Channel Strip 50 Dynamics Graph The Dynamics Graph display used with Ex pander Gate and Compressor Limiter processing shows a curve that represents the level of the input signal on the horizontal x axis and the amount of gain reduction applied on the vertical y axis The display shows two vertical lines representing the Threshold setting for the Expander Gate and Com pressor Limiter respectively The Dynamics Graph display also features an ani mated red ball in the gain transfer curve display This ball shows the amount of input gain x axis and gain reduction y axis being applied to the in coming signal at any given moment To indicate overshoots when an incoming signal peak is too fast for the current compression setting the cursor temporarily leaves the gain transfer curve Use this graph as a visual guideline to see how much dynamics processing y
407. s Set the release time fast enough that you don t hear unnatural dy namic changes but slow enough that you don t hear unnecessary pumping between two loud pas sages in rapid succession Chapter 11 Channel Strip 44 Chapter 11 Channel Strip Avid Channel Strip is available in DSP Native and AudioSuite formats Channel Strip provides EQ Dynamics Filter and Gain effects Channel Strip processing algorithms are based on the award win ning Euphonix System 5 console channel strip ef fects Channel Strip supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Channel Strip supports mono stereo and greater than stereo multichannel formats up to 7 1 In addition to standard knob and fader controls Channel Strip also provides a graph to track the gain transfer curve for the Expander Gate Com pressor Limiter and Side Chain effects and a Fre quency Graph display that shows the response curve for the current EQ setting on a two dimen sional graph of frequency and gain The frequency graph display also lets you modify frequency gain and Q settings for individual EQ bands by dragging their corresponding points in the graph Channel Strip provides different sections for signal metering and gain adjustment signal path ordering dynamics processing and equalization and filter ing Channel Strip Compressor Limiter tab shown Greater than stereo formats are only available with Pro Tools HD
408. s and other high fre quency noises that can occur in vocals voice overs and wind instruments such as flutes These sounds can cause peaks in an audio signal and lead to distortion The De Esser reduces these unwanted sounds using fast acting frequency dependent compression The Threshold control sets the level above which compression starts and the Frequency Freq con trol sets the frequency band in which the De Esser operates Using De Essing Effectively To use de essing most effectively insert the De Es ser after compressor or limiter plug ins The Frequency control should be set to remove sib ilants typically the 4 10 kHz range and not other parts of the signal This helps prevent de essing from changing the original character of the audio material in an undesired manner Similarly the Range control should be set to a level low enough so that de essing is triggered only by sibilants If the Range is set too high a loud non sibilant section of audio material could cause un wanted gain reduction or cause sibilants to be over attenuated To improve de essing of material that has both very loud and very soft passages automate the Range control so that it is lower on soft sections De Esser III Level Meters These controls let you track input output and gain reduction levels Input and Output Meters The Input and Output meters show peak signal lev els before and after dynamics processing Green
409. s and signal routing can vary depending on what you want to do while recording and mixing with Eleven This sec tion gives you a few specific examples of some of the many different ways you can choose to work Recording Dry Monitor Through Eleven on page 270 Recording Wet Record Eleven Processed Track to Disk on page 271 Recording Dry and Eleven Simultaneously on page 272 Processing Pre Recorded Tracks Through Eleven on page 273 Blending Eleven Cabinets and Amps on page 274 Recording Dry Monitor Through Eleven This workflow lets you record dry clean while the recorded signal is processed through Eleven let ting you hear it but without committing the track to that tone forever The flexibility to audition and compare different settings and combinations of amps cabinets and microphones is a very creative and powerful tool for mixing and arranging To record dry and monitor through Eleven 1 Choose Track gt New and configure the New Track to create one mono Audio Track 2 Set the track input to the audio interface input your guitar is plugged in to such as In 1 Mono 3 Insert Eleven on the track see Inserting Eleven on Tracks on page 261 Mic Axis switch in the Cabinet section Chapter 44 Eleven 271 4 Choose a Settings file preset or adjust Eleven s parameters to get your tone see Eleven Settings Presets on page 263 5
410. s highlighted in the Waveform display If Early length is set to zero then the Early setting have no effect on the au dio Otherwise when changes are made to any con trol in the Early group the IR waveform is recalcu lated and displayed in the Waveform display Length Adjusts the length of the Early reflections from zero to 500 ms When set to zero other con trols in the Early group have no effect on the audio The Early Length control adjusts the point in the impulse where the early portion ends and the late portion or tail begins For the most realistic reverb results Early Length should be adjusted while viewing the waveform display The early portion of a reverb IR is typically seen as a series of discrete spikes at the beginning of the waveform Early Length can however be ad justed to any value to explore other creative possi bilities Early controls Early controls indicated in IR waveform Chapter 30 Space 200 Size Changes the size of the Early reflections from 50 to 200 Early Size expands or contracts the reflections in the early portion of the IR as speci fied by the Early Length control Reduce the Early Size to give the space a smaller tighter sound In crease the Early Size to give the space a larger roomier sound Lo Cut Early Lo Cut controls the frequency of a highpass filter applied to the early portion of the IR as specified by the Early Length control The de fault setting of zero disables
411. s sets the first stage of your gain structure for Eleven To prepare your guitar and Pro Tools host based hardware for input calibration 1 In Pro Tools choose Setup gt Playback Engine and set your Hardware Buffer to a low enough setting to reduce monitor latency 2 On your guitar select the highest output pickup or position and set the volume and tone controls to 10 maximum 3 Strum full chords your loudest expected play ing while watching the Input indicators on your audio hardware 4 Adjust the Input Gain on your audio interface high enough to indicate a strong signal on the hardware Input LED but not overloading the input To prepare your guitar and HD hardware for input calibration 1 On your guitar select the highest output pickup or position and set all volume and tone controls to the maximum 2 Strum full chords your loudest expected play ing while watching the Input indicators on your audio hardware 3 Adjust your pre amp input gain until you see a strong signal on your audio interface Input me ters but not overloading the input Set Up a Pro Tools Track In this step you ll create and configure an audio track to use for the final stage of input calibration To set up and check Track level all systems 1 Choose Tracks gt New and create one mono Audio track 2 In the Mix window click the track Input selector and choose your guitar input 3 Click the track Insert selector and se
412. s the following greater than stereo multichannel formats LCR LCRS 5 0 5 1 7 1 SDDS 7 1 When inserting Down Mixer on a compatible greater than stereo multichannel track the channel format of the track output changes to stereo When inserting Down Mixer on a stereo track the channel format of the track output changes to mono Down Mixer 5 1 to Stereo Avid Down Mixer Stereo to Mono Chapter 52 Down Mixer 305 Source The Source section of the Down Mixer plug in pro vides controls that let you mute invert the phase and adjust the level of each input channel to the Down Mixer Mute When enabled the Mute button mutes the channel input to the Down Mixer Phase When enabled the Phase button inverts the phase of the channel input to the Down Mixer Level You can adjust the level of the channel input to the Down Mixer from 45 dB to 12 dB For stereo to mono down mixing both the Left and Right chan nels are mixed to summed mono For greater than stereo multichannel down mixing the following rules apply All left channel sources L Lc Ls Lss Lsr feed to the left channel L of the down mixer All right channel sources R Rc Rs Rss Rsr feed to the right channel R of the down mixer The center channel C and low frequency chan nel LFE are panned center into the stereo field of the down mixer Meter The level meters for source cha
413. s the tubes are biased further into Class B operation Fairchild 660 Tubes wires and iron Chapter 13 Fairchild Plug Ins 76 Fairchild 660 Controls Adjust the Input Gain and Threshold controls to gether until you get the sound you want Like many classic compressors after a little bit of tweaking you ll know immediately when you get it right Input Gain Input Gain sets the input level to the unit and the compression threshold just like the In put control on an 1176 Full clockwise is loudest Threshold Threshold adjusts the gain to the side chain just like the Peak Reduction control on an LA 2A Time Constant Selects the attack and release times for the compressor One is fastest and six is slow est Seven and eight are custom presets Fairchild 660 Tips and Tricks 5 6 7 8 The Fairchild manual documents Time Constant settings 5 and 6 as user presets although you have to go inside with a soldering iron to change them We used the factory default values Bonus Settings Settings 7 and 8 do not exist on real world units well at least most of them These settings are taken from a real world Fairchild modification invented by Dave Amels many years before he designed the plug in version What do they do Settings 7 and 8 offer versions of Time Constant 2 with a gentler release useful for compressing vocals and other program material where you desire more subtlety in the compression action Giv
414. s to 1 8 of the original duration depending on which Range button is enabled 2x 4x or 8x The Shift field only displays up to 2 decimal places but lets you type in as many decimal places as you want up to the IEEE standard While the display rounds to 2 decimal places the actual time shift is applied based on the number you typed This is es pecially useful for post production pull up and pull down factors see Using X Form for Post Produc tion Pull Up and Pull Down Tasks on page 153 2x 4x and 8x Range Buttons The 2x 4x and 8x Range buttons set the possible range for the Time Shift Pitch Shift and Formant Shift controls 2x Lets you apply Time Shift Pitch Shift and For mant Shift from 50 00 to 200 00 where 50 00 is 2 times the original duration and 200 00 is 1 2 of the original duration 4x Lets you apply Time Shift Pitch Shift and For mant Shift from 25 00 to 400 00 where 25 00 is 4 times the original duration and 400 00 is 1 4 of the original duration 8x Lets you apply Time Shift Pitch Shift and For mant Shift from 12 50 to 800 00 where 12 50 is 8 times the original duration and 800 00 is 1 8 of the original duration X Form Transient Section Controls The Transient section provides controls for setting the sensitivity for transient detection and for adjust ing the analysis window size Sensitivity The Sensitivity setting controls how X Form deter mines and interprets
415. s to the signal being compressed This signal path is derived from the input signal or Key Input depending on the user s selection When enabled the Side Chain EQ filter lets the user tailor the equalization of the side chain signal so that the compression becomes frequency sensi tive The Side Chain EQ filter has the following set tings High Pass Makes the compressor s detector less sensitive to low frequencies in the input signal or Key Input by rolling off at a rate of 6 dB per octave For example you might use this setting on a mix to prevent a bass guitar or bass drum from causing too much gain reduction Band Emphasis Makes the compressor s detector more sensitive to mid to high frequencies in the in put signal or Key Input by boosting those frequen cies in the side chain signal For example you might use this setting to reduce sibilance in vocal tracks Combined Enables the High Pass and peak settings simultaneously to make the compressor s detector more sensitive to high frequencies and less sensi tive to low frequencies Off Disables the Side Chain EQ control See Using the Smack Side Chain Input on page 131 for more information on using the Side Chain EQ on a Key Input High Pass Side Chain EQ Band Emphasis Side Chain EQ Combined Side Chain EQ Chapter 22 Smack 130 Smack Distortion Control When enabled Distortion adds subtle second order and third order harmonic distortion to the outpu
416. ser III 68 Dynamics III Side Chain Input 70 Chapter 13 Fairchild Plug Ins 75 Fairchild 660 75 Fairchild 670 77 Chapter 14 Focusrite D3 78 D3 Compressor 78 D3 Limiter 79 D3 Side Chain Processing 79 Using D3 in Stereo 79 D3 Common Controls 80 D3 Compressor Controls
417. ser Tips and Tricks More Harmonics More Fun The richer the harmonic content of the sound the more there is to filter and sweep Try adding distor tion using the SansAmp PSA 1 before the phaser it s a cool variation on the common signal path used when putting a phaser in front of a guitar amp Aggressive and Extreme Dr Moog apparently took these mantras of early 21st Century recording science to heart when he designed the Rate knob on his phaser Flick the Rate switch to Hi and let the party begin Try mut ing a track and mixing in bits of extremely phase swept material It s an Effect Play with It All the controls on the Moogerfooger 12 Stage Phaser are fully independent of one another This means you can set them in any combination that you wish There is no such thing as a wrong com bination of settings so you can experiment all you like to find new exciting effects for your music Chapter 38 Moogerfooger Ring Modulator 225 Chapter 38 Moogerfooger Ring Modulator Moogerfooger Ring Modulator provides a wide range carrier oscillator and dual sine square wave form LFO Add motion to rhythm tracks and achieve radical lo fidelity textures you set the limits Moogerfooger Ring Modulator is available in DSP Native and AudioSuite formats Moogerfooger Ring Modulator supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates The Moogerfooger Ring Modulator operates as a
418. shion Chapter 30 Space 191 Space Controls and Displays The Space interface is divided into the following sections 1 Display area See Space Display Area on page 192 2 IR Browser See Space IR Browser on page 195 3 Primary controls See Space Primary Controls on page 197 4 Group Selectors and Controls See Space Group Selectors and Controls on page 198 The Space interface 1 2 3 4 Chapter 30 Space 192 Space Display Area The display area of Space operates in the following four modes indicated by the Display Mode selec tors at the top right hand corner of the Space win dow Waveform mode Picture Preview mode Snapshot mode Preferences mode The Display area changes based on the selected mode Info Bar At all times the Info bar at the bottom of the dis play area window shows the following controls and information Snapshot Menu A pop up menu allowing quick se lection or automation of a snapshot IR Name Displays the folder and file name of the currently loaded IR Quick Browser Controls The Quick browser con trols allow the IR to be quickly changed even when the IR browser is closed automatically loading each IR sequentially The Waveform icons step backwards and forwards through IRs and automat ically load the IR file The Folder icons step back wards and forwards through folders The Quick browser requires an IR to be currently loa
419. sion Output Gain Output Gain provides makeup gain after compression Slope Slope is similar to the compression ratio controls found on other compressors However on the JOEMEEK the actual ratio varies based on program material so the term Slope is used instead In practice 1 is very gentle compression and 2 or 3 are typically right for voice and submixes The higher numbers are better for instruments and ex treme sounds At the suggestion of the original de signers the 5 setting found on the later model JO EMEEK SC2 2 were added Use 5 to create severe pumping effects Attack Attack sets the time that the compressor takes to act Slower attacks are typically used when the sound of the compression needs to be less obvi ous Release Release sets the time during which signal returns to normal after compression With longer release times the compression is less noticeable JOEMEEK SC2 Compressor Chapter 16 JOEMEEK SC2 Compressor 90 JOEMEEK Compressor Tips and Tricks Not Perfect Just Right Standard engineering practice says that a compres sor should work logarithmically For a certain in crease of volume the output volume should rise proportionally less with a result that the more you put in the more it s pushed down The JOEMEEK compressor doesn t work this way As volume increases at the input a point is reached where the compressor starts to work and the gain through the amplifier is reduced If the
420. so RetroLoop Reverb One ReVibe Revolution rS9 rS18 RTAS Salesview Sci Fi Scorch ScriptSync SecureProductionEnvironment Shape to Shape ShuttleCase Sibelius SimulPlay SimulRecord Slightly Rude Compressor Smack Soft SampleCell Soft Clip Limiter SoundReplacer SPACE SPACEShift SpectraGraph SpectraMatte SteadyGlide Streamfactory Streamgenie StreamRAID SubCap Sundance Sundance Digital SurroundScope Symphony SYNC HD SYNC I O Synchronic SynchroScope Syntax TDM FlexCable TechFlix Tel Ray Thunder TimeLiner Titansync Titan TL Aggro TL AutoPan TL Drum Rehab TL Everyphase TL Fauxlder TL In Tune TL MasterMeter TL Metro TL Space TL Utilities tools for storytellers Transit TransJammer Trillium Lane Labs TruTouch UnityRAID Vari Fi Video the Web Way VideoRAID VideoSPACE VTEM Work N Play Xdeck X Form and XMON are either registered trademarks or trademarks of Avid Technology Inc in the United States and or other countries Bonjour the Bonjour logo and the Bonjour symbol are trademarks of Apple Computer Inc Thunderbolt and the Thunderbolt logo are trademarks of Intel Corporation in the U S and or other countries This product may be protected by one or more U S and non U S patents Details are available at www avid com patents Product features specifications system requirements and availability are subject to change without notice Guide Part Number 9329 65427 00 REV A 06 1
421. ss option to apply a high pass filter to the side chain processing at the selected frequency Notch Select the Notch option to apply a notch fil ter to the side chain processing at the selected fre quency Band Pass Select the Band Pass option to apply a band pass filter to the side chain processing at the selected frequency Side Chain Processing Graph The Side Chain Processing Graph display shows the frequency curve for the selected Filter Type at the selected Filter Frequency Selecting the Source setting for Side Chain processing Chapter 11 Channel Strip 55 Channel Strip EQ Filters Section The EQ Filters section of Channel Strip provides a high quality 4 band parametric equalizer for adjusting the frequency spectrum of audio material EQ Filters Graph The EQ Filters section provides an interactive Frequency Graph display that shows the response curve for the current EQ settings on a two dimensional graph of frequency and gain The Frequency Graph display also lets you modify frequency gain and Q settings for individual EQ bands by dragging their correspond ing points in the graph The Frequency Graph display also plots the frequency Q and filter shape of the two filters when either or both are enabled Frequency Graph Gain Resolution Channel Strip lets you view the gain scale on the Frequency Graph display either in 3 dB increments from 12 dB to 12 dB or in 6 dB increments from 24 dB to 24 dB To chan
422. st for the current compression setting the ball temporarily leaves the gain transfer curve Use this graph as a visual guideline to see how much dynamics processing you are applying to the incoming audio signal Using the Dynamics Graph to Adjust Controls You can drag in the Dynamics Graph display to adjust the corresponding Compressor controls The cursor updates to show which control is being ad justed Threshold Ratio Knee Depth To adjust the Threshold setting using the Dynamics graph Position the cursor over the vertical Threshold line in the graph and drag left or right to make the adjustment To adjust the Ratio setting using the Dynamics graph Position the cursor over the ratio curve in the graph and drag up or down or left or right to make the adjustment Dynamics graph display Input signal level x axis Threshold Gain reduction amount y axis Cursor indicates Threshold adjustment Cursor indicates Ratio adjustment Chapter 18 Pro Compressor 101 To adjust the Knee setting using the Dynamics graph Position the cursor over the knee of the curve in the graph and drag up or down or left or right to make the adjustment To adjust the Depth setting using the Dynamics graph Position the cursor over the horizontal Depth line in the graph and drag up or down or left or right to make the adjustment Detection Modes Pro Compressor prov
423. stic space s character can sound harsh and unnatural Each real world acoustical environment from a closet to a cathedral has its own unique acoustical character or sonic signature When the reflections and reverberation produced by a space combine with the source sound we say that the space is ex cited by the source Depending on the acoustic en vironment this could produce the warm sonic char acteristics we associate with reverberation or it could produce echoes or other unusual sonic char acteristics Reverb Character The character of a reverberation depends on a num ber of things These include proximity to the sound source the shape of the space the absorptivity of the construction material and the position of the listener Reflected Sound In a typical concert hall sound reaches the listener shortly after it is produced The original direct sound is followed by reflections from the ceiling or walls Reflections that arrive within 50 to 80 milli seconds of the direct sound are called early reflec tions Subsequent reflections are called late rever beration Early reflections provide a sense of depth and strengthen the perception of loudness and clar ity The delay time between the arrival of the direct sound and the beginning of early reflections is called the pre delay Chapter 28 Reverb One 160 The loudness of later reflections combined with a large pre delay can contribute to the perception of largen
424. t greatest regis tered peak Peak Hold The Attenuation meter pro vides similar graphical and numeric representa tions for the amount of compression applied to the input signal The numerical display for the Peak value turns orange when the signal exceeds 0 dB on the meters You can click the numerical display to reset the dis played value Pro Compressor Input Section The Input section provides input metering and con trols for adjusting the level of the input signal Input Level The Input Level control sets the input gain of the plug in before processing letting you boost or at tenuate gain at the plug in input stage To adjust the level of the input signal do one of the following Click in the Input Level field to type a value 36 0 dB to 36 0 dB Click the Input Level control and drag up or down to adjust the Input Level setting Input Meters The Input meters show peak signal levels before processing Peak indicators in the Output meters Peak Hold value Peak Hold indicator Sample Peak indicator Average meter Input section with Meters and Input Level control Chapter 18 Pro Compressor 99 Pro Compressor Output Section The Output section provides output metering and controls for adjusting the level of the output signal The Output meters can also be toggled to show post processing gain attenuation meters Output Level The Output Level control sets the output level after
425. t signal Odd harmonics produce waveforms that are more square shaped and are often described as harsh sounding Even harmonics produce waveforms with more rounded edges and are often described as smooth sounding The amount of distortion that Smack applies to the input signal depends on both the level of the input signal and the amount of compression being ap plied Odd Applies mostly odd and some even harmon ics to the distortion Even Applies mostly even and some odd harmon ics to the distortion O E Applies an equal blend of odd and even har monic distortion Smack HPF Toggle Switch When enabled the HPF high pass filter toggle switch gently rolls off audio frequencies lower than 60 Hz in the output signal at a rate of 6 dB per oc tave This is especially useful for removing thumps or pops from vocals bass or kick drums Smack VU Meter The VU meter displays the amount of input level output level or gain reduction from compression depending on the current Meter Mode button set ting It is calibrated to a reference level of 14 dBFS 0 VU Meter Mode Button and Clip Indicators The Meter Mode button toggles between display ing three display modes as follows In Displays the input signal level referenced to 14 dBFS 0 VU Out Displays the output signal gain referenced to 14 dBFS 0 VU GR Displays the amount of gain reduction applied by the
426. t is available in DSP Native and AudioSuite formats Smack supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Smack supports mono stereo and greater than stereo multichannel formats up to 7 1 The Smack compressor limiter plug in has the fol lowing features Three modes of compression Norm mode emulates FET compressors which can have faster attack and release times than electro optical compressors This mode lets you fine tune compression precisely by adjusting the attack release and ratio con trols Warm mode is based on Norm mode but has release characteristics more like those of elec tro optical limiters Opto mode emulates classic electro optical limiters which tend to have gentler attack and release characteristics than FET compressors The attack release and ratio controls are not adjustable in this mode Key Input side chain processing which lets you trigger compression using the dynamics of another signal Side Chain EQ filter which lets you tailor the compression to be frequency sensitive High Pass filter which lets you remove thumps or pops from your audio Distortion control which lets you add different types of subtle harmonic distortion to the output signal Greater than stereo formats are only available with Pro Tools HD Smack Smack has no control to directly adjust the threshold level the lev
427. t the Output Volume setting Output Meters The Output meters show peak signal levels after processing Dark Blue Indicates nominal levels from INF to 12 dB Light Blue Indicates pre clipping levels from 12 dB to 0 dB White Indicates full scale levels from 0 dB to 6 dB which can result in distortion and clipping Output section 5 1 channel format shown Chapter 11 Channel Strip 49 Channel Strip FX Chain Channel Strip lets you determine the signal path through the available Equalizer EQ Filter FILT Dynamics DYN and Volume VOL processing modules This way you can determine the best signal path for the type of processing you want To set the FX Chain 1 Click the FX Chain show hide button to reveal the Process Order options 2 Click an effects chain ordering option to select it The available options include EQ gt FILT gt DYN EQ gt DYN gt FILT DYN gt EQ gt FILT FILT gt DYN gt EQ 3 Select PRE or POST to place the Output Volume control at the beginning or at the end of the ef fects signal chain Bypassing or Unbypassing Individual Effects Modules In the FX Chain display you can deselect or select individual effects modules to bypass or unbypass the effect To bypass an effect module Click the module so that it is not highlighted To unbypass an effect module Click the module so that it is highlighted Channel Strip Dynamic
428. t the output level do one of the following Click in the Output Level field to type a value INF dB to 12 dB Click the Output Volume control and drag up or down to adjust the Output Volume setting Output Meters The Output meters show peak signal levels after processing Attenuation Meters The Output meter can be switched to show Attenu ation metering for the processed signal from 0 dB to 36 dB To toggle between the Attenuation and Output meters Click the Output Attenuation toggle in the top right hand corner of the Output section Pro Expander Dynamics Graph The Dynamics Graph display shows a curve that represents the level of the input signal on the hor izontal x axis and the amount of processing ap plied on the vertical y axis The display shows a vertical line representing the Threshold setting for the Expander Output section with Meters and Output Level control Output Attenuation toggle Attenuation meters shown Chapter 19 Pro Expander 110 The Dynamics Graph display also features an ani mated red ball in the gain transfer curve display This ball shows the amount of input gain x axis and gain reduction y axis being applied to the in coming signal at any given moment To indicate overshoots when an incoming signal peak is too fast for the current compression setting the ball temporarily leaves the gain transfer curve Use this graph as a visual guideline to see
429. ted the LFO will cycle through the waveform once only and then stop LFO section When the Panner section is set to Envelope Env the controls in the LFO section have no effect on panning Selecting the LFO Waveform LFO Triggers Chapter 51 AutoPan 301 Beat Clock When the Beat Clock trigger is se lected the LFO synchronizes to MIDI Beat Clock TL AutoPan receives Beat Clock signal every 64th note The Duration menu determines how often the Beat Clock signal triggers TL AutoPan ranging from every 16th note to every 4 bars When Beat Clock signal is received the Beat Clock trigger light blinks brightly Using the Beat Clock function enables TL AutoPan to produce consistent panning results ensuring that the LFO is always in the same state at each beat Envelope When the Envelope trigger is selected the LFO is triggered directly by the Envelope De tector which analyzes the amplitude of the audio signal If the Side Chain Input selector in the Enve lope section is activated then the side chain audio signal is used instead When activated the Enve lope light blinks brighter when an audio signal is detected The threshold level can be adjusted using the Threshold control in the Envelope section If the Envelope Detector is completely released due to previous portions of the audio signal going above threshold a trigger occurs the next time the audio goes above the threshold level Another trig ger will not happen unt
430. ter output levels for all channels The range of this control is from 0 dB of attenuation to 40 dB of gain Impact Ext Control Side Chain External On Off enables and disables side chain processing With side chain processing you can trigger compression from a separate reference track or external audio source The source used for trig gering side chain processing is referred to as the Key Input Impact Listen On Off Control Key Listen On Off enables and disables audition ing of the Key Input the reference track or external audio source used for triggering side chain pro cessing This is useful for fine tuning Impact s compression settings to the Key Input Impact Gain Reduction Meter The Gain Reduction meter is an analog style meter that indicates the amount of gain reduction in dB The range of this meter is from 0 dB to 40 dB The gain reduction meter displays the amount of gain reduction linearly from 0 20 db and non linearly from 20 40 dB Applying large amounts of Make Up gain will boost the level of any noise or hiss present in audio material making it more audible See Using the Impact Compressor on page 87 for instructions on setting up and us ing a key input Chapter 15 Impact 87 Impact Meters The Input Output meters indicate input and output signal levels in dB When Impact is used in mono or stereo both input and output meters are dis played When Impact is used in a multichannel for
431. tes enabled In this state the limiter is active and using available DSP resources Black indicates disabled In this state the limiter is not using DSP resources Gray indicates bypassed In this state the limiter is not active but is still using available DSP re sources To disable the limiter Control click Mac or Start key click Win dows the icon When the limiter is disabled the icon is black To re enable the limiter Click the icon When the limiter is enabled the icon is white To bypass the limiter Click the icon a second time When the limiter is bypassed the icon is gray If you are using the Compressor Limiter plug in which allows you to use either the compressor or the limiter but not both simultaneously you must disable one module by Control clicking Mac or Start clicking Windows the icon before you can enable the other Limit LED When lit the Limit LED indicates that limiting is being applied When unlit limiting is not being ap plied Threshold This sets the threshold level Signals that exceed this level will be limited Signals that are below it will be unaffected A setting of 0 dB is equivalent to no limiting The range of this control is from 24 dB to 0 dB Limiter controls Limiter In Out icon Limit LED Threshold Chapter 14 Focusrite D3 84 Using the Side Chain Input in D3 The side chain is the split off signal used by a plug in
432. th Quad instances of Dither will have all of their dither noise un correlated However any additional in stances of the Dither plug in will begin to re peat the dithering noise Chapter 50 POW r Dither 294 Chapter 50 POW r Dither POW r Dither is a dither generation plug in The POW r Dither plug in is an advanced type of dither that provides optimized bit depth reduction It is designed for final stage critical mixdown and mas tering tasks where the highest possible fidelity is required when reducing bit depth POW r Dither is available in DSP and Native formats POW r Dither supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates POW r Dither operates as a mono multi mono or stereo plug in POW r Dither Controls POW r Dither provides a variety of controls for ad justing plug in parameters Bit Resolution Use this pop up menu to choose either 16 or 20 bit resolutions for POW r Dither processing Set this control to the maximum bit resolution of your des tination 16 bit Recommended for output to digital devices with a maximum resolution of 16 bits such as DAT and CD recorders 20 bit Recommended for output to devices that support a full 20 bit recording data path Noise Shaping Noise shaping can further improve audio perfor mance and reduce perceived noise inherent in dith ered audio Noise shaping uses filtering to shift noise away from frequencies in the middle of the
433. th a default value of 0 dB Tape Machine The Tape Machine control lets you select one of three tape machine types emulated by the plug in each with its own sonic characteristics US Emulates the audio characteristics of a 3M M79 multitrack tape recorder Swiss Emulates the audio characteristics of a Studer A800 multitrack tape recorder Lo Fi Simulates the effect of a limited bandwidth analog tape device such as an outboard tape based echo effect Tape Formula The Tape Formula control lets you select either of two magnetic tape formulations emulated by the plug in each with its own saturation characteris tics Classic Emulates the characteristics of Ampex 456 exhibiting a more pronounced satura tion effect Hi Output Emulates the characteristics of Quantegy GP9 exhibiting a more subtle saturation effect Reel Tape Flanger Controls In addition to the Drive Output Tape Machine and Tape Formula controls Reel Tape Flanger has the following controls Range The Range control adjusts the overall magnitude of the variable delay which determines the offset be tween the two modeled tape machines A center or zero setting results in no offset Range is contin uously adjustable from 20 ms to 20 ms and is divided into two types of effects flanging and au tomatic double tracking Flange Range settings within the narrow center band around zero simulate tape flanging with a phase cancellation e
434. th selected Panner section mono to stereo left to right path selected Panner section mono to 5 0 left to right path selected Chapter 51 AutoPan 299 Spread The Spread slider opens or constricts the field of panning At 100 the spread of the panning field is at its greatest At 0 the spread of the panning field is completely constricted and the sound is centered and stationary left right and front back The Spread slider is only available with mono to quad and mono to 5 0 formats and a circular path clockwise or counterclockwise selected Panning Source Click LFO or Env to select the source for panning When the Source is set to LFO panning is con trolled by the LFO and its controls see AutoPan LFO Controls on page 300 When the Source is set to Envelope Env panning is controlled by the Envelope Detector and its controls see AutoPan Envelope Controls on page 302 The Envelope Detector can be triggered by the panned audio sig nal or by a side chain input see Using the Side Chain Input on page 303 Panning Display The Panning display graphically represents the panning field and the location of the sound source within that field Sound Location Indicator This bright yellow light indicates the location of the sound source Panning Field Indicator This is the grey line on which the yellow Sound Location indicator travels and indicates the panning field Path The Path sele
435. that have the task of recreating the audio through digital reconstruction filters are clipping repeatedly throughout most CDs on the market The result is that most CDs and DVDs end up dis torting with regularity when they are asked to re construct and play back audio that appears to be completely legal because not a single sample ac tually clipped Intersample peaks D A converter range Chapter 56 MasterMeter 329 Seven consumer CD players were subjected to tests Nielsen 2003 designed to analyze their ability to reproduce and reconstruct signal levels above full scale 0 dBFS All of the players experienced dif ficultly dealing with signal levels this high further showing that while all of the samples can be legal the level can still be hotter than is legal The result is that a CD player can be unable to reproduce the audio accurately In some cases the reconstruction sounds perfect to the mastering engineer be cause the engineer s equipment can actually repro duce the waveforms properly The Red Book format for CDs and the DVD specs both allow for this illegal content and the mastering engineer is still allowed to put out releases that meet the spec while allowing consumers players to distort With an oversampled peak meter the engi neer will be able to know that the music is clipping by how much and where With this knowledge the engineer can then decide with complete informa tion whether or not t
436. the cutoff frequency High Pass and Low Pass filters for the currently selected filter type The 1 Band EQ may be set to any one of six EQ types High Pass Notch High Shelf Low Shelf Peak and Low Pass by clicking the corresponding icon in the EQ Type selector Frequency control Q control Frequency Graph display 1 Band EQ Frequency response curve Control dot Frequency Graph display EQ Type selector Gain Freq and Input Level and Polarity controls Q controls Chapter 4 EQ III 18 Band Controls The individual EQ types have some combination of the following controls as noted below 1 Band EQ III Types High Pass Filter The High Pass filter attenuates all frequencies be low the Frequency setting at the selected rate 6 dB 12 dB 18 dB or 24 dB per octave while letting all frequencies above pass through No gain control is available for this filter type Notch Filter The Notch Filter attenuates a narrow band of fre quencies centered around the Frequency setting No gain control is available for this EQ type The width of the attenuated band is determined by the Q setting High Shelf EQ The High Shelf EQ boosts or cuts frequencies at and above the Frequency setting The amount of boost or cut is determined by the Gain setting The Q setting determines the shape of the shelving curve Control Value Frequency Range All 20 Hz to 20 kHz Frequency Default All 1 kHz Q Ra
437. the high frequen cies from a tone making it sound more mellow or muted The Moogerfooger Lowpass Filter contains a genuine four pole lowpass filter We say genu ine because the four pole filter a major part of the MOOG Sound of the 60s and 70s was first patented by Bob Moog in 1968 The digital version preserves all the character nuances and personal ity of his original classic analog design Moogerfooger Low Pass Filter Chapter 36 Moogerfooger Lowpass Filter 220 An Envelope Follower tracks the loudness contour or envelope of a sound Think of it like this each time you play a note the envelope goes up and then down The louder and harder you play the higher the envelope goes In the Moogerfooger Lowpass Filter the Envelope Follower drives the cutoff fre quency of the Lowpass Filter Since the envelope follows the dynamics of the input it plays the fil ter by sweeping it up and down in response to the loudness of the input signal Moogerfooger Lowpass Filter Controls Envelope Section Amount The Amount knob determines how much the envelope varies the filter When the knob is counterclockwise the envelope signal has no effect on the filter When the knob is fully clockwise the envelope signal opens and closes the filter over a range of five octaves Smooth Fast The Smooth Fast switch determines how closely the envelope tracks the loudness of the input signal Some sounds like guitar chords
438. the threshold However when the input signal reaches the threshold on the Left or the Right channel processing is triggered for both the Left and the Right channel Key If Key is selected the plug in uses the ampli tude of a separate reference track or external audio source to trigger dynamics processing The refer ence track used is selected using the Plug In Key Input selector in the Plug In window header With greater than stereo multichannel processing the key signal triggers dynamics processing for all pro cessed audio channels equally All Linked If All Linked is selected dynamics pro cessing is applied equally to all channels when the input signal reaches the threshold on any input channel except for the LFE channel if present The LFE channel is processed independently based on its own input signal Detection The Detection options include Peak or Avg Aver age Peak Select the Peak option to apply side chain processing according to the detected peak ampli tude Average Select the Average option to apply side chain processing according to the detected average amplitude Filter Frequency The Filter Frequency control lets you set the fre quency for the selected Filter Type Filter Type Four Filter Type options are available for side chain processing Low Pass Select the Low Pass option to apply a low pass filter to the side chain processing at the selected frequency High Pass Select the High Pa
439. tomatic selects the reverb tail type stored with the room type Natural is an average reverb tail type with no ex treme characteristics Smooth is optimized for large rooms Fast Attack can be useful for plate reverbs Dense is similar to smooth and can also be good for a plate reverb Tight is good for small to medium rooms Sparse 1 produces sparse early reflections with a high diffusion buildup Sparse 2 can be useful for a spring reverb Wide is a generic large reverb Small is optimized for small rooms ReVibe II Reverb Section Controls The Reverb section has controls for the various re verb tail elements including level time size spread attack time attack shape rear shape diffu sion and pre delay These determine the overall character of the reverb tail Size Control The Size control adjusts the apparent size of the re verberant space from small to large Set the Size control to approximate the size of the acoustic space you want to simulate Size values are given in meters The range of this control is from 2 0 m to 60 0 m though relative size will change based on the current Room Type Higher Size settings increase both the Time and Spread values Time Control Time controls how long the reverberation continues after the original source signal stops The range of this control is from 100 0 ms to Inf infinity Set ting Time to its maximum value will
440. tor To create an audio clip using the Signal Generator plug in 1 Make a selection in the Edit window 2 Choose AudioSuite gt Signal Generator 3 Enter values for the Frequency Level and Signal controls 4 Click Render in the Signal Generator plug in Select the Create Continuous File option for greater flexibility in making audio selections for use with the Signal Generator plug in You can use the AudioSuite Signal Generator plug in for musical purposes as well as for testing purposes For example you might want to add a little color to a kick drum track by doubling it with a 50 Hz tone using the kick track as the key input signal gating the tone track Chapter 58 SoundReplacer 335 Chapter 58 SoundReplacer SoundReplacer is an AudioSuite only plug in de signed to replace audio elements such as drums percussion and sound effects in Pro Tools tracks with alternate sounds SoundReplacer can quickly and intelligently match the timing and dynamics of original performance material making it ideal for both music and audio post production SoundReplacer features Sound replacement with phase accurate peak alignment Intelligent tracking of source audio dynamics for matching the feel of the original performance Three separate amplitude zones per audio event for triggering different replacement samples ac cording to performance dynamics Zoomable waveform display for precision threshold ampli
441. transients from the original audio Part of X Form s processing relies upon sep arating transient parts of the sample from non transient parts Transient material tends to change its content quickly in time as opposed to parts of the sound which are more sustained Sensitivity is only available when Polyphonic is selected as the Audio Type For highly percussive material lower the Sensitiv ity for better transient detection especially with the Rhythmic audio setting For less percussive mate rial a higher setting can yield better results Exper iment with this control especially when shifting drums and percussive tracks to achieve the best re sults When changing to a smaller Range setting such as switching from 8x to 2x the Time Shift and Pitch Shift settings are constrained to the limits of the new smaller range For example with 8x enabled and Time Shift set to 500 switching to 2x changes the Time Shift value to 200 X Form Transient section Chapter 26 X Form 150 Window The Window setting determines the analysis win dow size You can adjust the Window from 10 0 milliseconds to 100 0 milliseconds Adjust the Window control or click the Window field and type a value Window is only available when Mono phonic is selected as the Audio Type Try larger window sizes for low frequency sounds or sounds that do not have many transients Try smaller window sizes for tuned drums and percus sion H
442. trols Speed The Speed control adjusts the delay time calibrated to tape speed A slower tape speed results in a lon ger delay A faster tape speed results in a shorter delay The displayed tape Speed value corresponds to the delay time resulting from the distance between the record and play heads on an Ampex 440 series tape transport Tape speed is adjustable from approximately 1 7 8 ips 1486 ms delay to approximately 30 ips 93 ms delay with a default value of approxi mately 15 ips 172 ms delay You can synchronize the delay time to the current tempo of the Pro Tools session See Synchroniz ing Reel Tape Delay to Session Tempo on page 212 Feedback The Feedback control adjusts the amount of de layed output fed back into the input allowing gen eration of multiple echoes A higher feedback amount results in more echo regeneration A lower feedback amount results in less echo regeneration Feedback amount is adjustable from 0 to 100 per cent with a default value of 30 percent Chapter 33 Reel Tape Delay 211 Wow Flutter The Wow Flutter control adjusts the amplitude of the tape machine s wow and flutter or the amount of fluctuation in tape speed A higher setting results in wider fluctuations in speed A lower setting re sults in narrower fluctuations in speed Wow Flut ter is adjustable from 0 to 1 percent with a default value of 0 20 percent Wow Speed Plug In Automation Playlist or Contro
443. trols Pro Compressor also provides a dynamics graph to track the gain transfer curve for compression as well as a frequency graph for side chain filtering Additionally the dynamics graph can be used to graphically edit the Threshold Ratio Knee and Depth settings Pro Compressor Metering Pro Compressor provides combined meters that show both sample peak metering and averaged me tering Pro Compressor uses sample peak meters over average metering for Input and Output signals Attenuation metering uses sample peak metering only The Peak Hold value is displayed numerically at the top of the meter and the Peak Hold indicator ap pears as a thin orange line in the meter This pro vides highly accurate visual metering correlation with the audio signal Pro Compressor also dis plays averaging metering with an integration time of approximately 400 ms Input and Output meters use the following color coding Dark Blue Indicates nominal levels from 90 to 20 dB Light Blue Indicates pre clipping levels from 20 dB to 0 dB Yellow Indicates full scale levels from 0 dB to 6 dB Greater than stereo formats are only available with Pro Tools HD Pro Compressor Chapter 18 Pro Compressor 98 Attenuation meters show yellow for the entire dynamic range displayed Peak Indicators The Input and Output meters provide graphical rep resentation of transient peaks as well as graphical and numerical display of the las
444. trols the output level of the reverb tail When set to 0 the reverb effect consists entirely of the early reflections if enabled Time Controls the rate at which the reverberation decays after the original direct signal stops The value of the Time setting is affected by the Size setting You should adjust the reverb Size setting before adjust ing the Time setting If you set Time to its maxi mum value infinite reverberation is produced The HF Damping and Reverb Color controls also affect reverb Time Attack Attack determines the contour of the reverberation envelope At low Attack settings reverberation builds explosively and decays quickly As Attack value is increased reverberation builds up more slowly and sustains for the length of time deter mined by the Spread setting When Attack is set to 50 the reverberation enve lope emulates a large concert hall provided the Spread and Size controls are set high enough To hear examples of reverb tail chorusing load one of the Chorus presets using the Plug In Librarian menu Chapter 28 Reverb One 162 Spread Controls the rate at which reverberation builds up Spread works in conjunctions with the Attack con trol to determine the initial contour and overall am bience of the reverberation envelope Low Spread settings result in a rapid onset of rever beration at the beginning of the envelope Higher settings lengthen both the attack and buildup stages of the
445. ts Trim gain control to attenuate the signal Input Knob and Amp Gain Eleven actually gives you two separate input gain stages to the plug in The Input knob in the Master section which af fects the signal level before entering the amplifier model The gain knob s on each amplifier which con trol the main input stage of that particular amplifier model This makes the Input knob useful for increasing or decreasing gain on amps that don t have a separate preamp Noise Gate After the Input Knob The Noise Gate is keyed triggered from the input signal The gate is applied to the output of the amp when open it lets sound pass from the amp to the cabinet module and when closed it silences amp output to the speaker cabinet Signal flow through Eleven Input LED Input knob Gate Amp Cabinet Mic Output knob Input from Pro Tools track disk or live input Output to Pro Tools output or bus Chapter 45 Lo Fi 281 Chapter 45 Lo Fi Lo Fi provides retro down processing effects Lo Fi features include Bit rate reduction Sample rate reduction Soft clipping distortion and saturation Anti aliasing filter Variable amplitude noise generator Lo Fi is available in DSP Native and AudioSuite formats Lo Fi supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Lo Fi operates as a mono multi mono or stereo plug in Lo Fi dow
446. ts the Delay Time setting When Sync is enabled the Tempo control is un available Duration The Duration setting lets you set the Delay Time based on a rhythmic value Select a note value whole note half note quarter note eight note or sixteenth note Additionally you can select the Dot or Triplet modifier buttons to dot the selected note value or make it a triplet For example select ing a quarter note and then selecting the dot indi cates a dotted quarter note and selecting an eighth note and then selecting the triplet indicates a triplet eighth note Duration buttons Chapter 31 Mod Delay III 206 Groove The Groove control provides fine adjustment of the delay in percentages of a 1 4 subdivision of the beat from 100 to 100 It can be used to add swing by slightly offsetting the delay from the precise beat of the track Modulation Section Rate The Rate control sets the rate of modulation of the delayed signal from 0 00 Hz to 20 0 Hz Depth The Depth control sets the depth of the modulation applied to the delayed signal from 0 to 100 Mix The Mix control sets the balance between the de layed signal wet and the original signal dry If you are using a delay for flanging or chorusing you can control the depth of the effect somewhat with the Mix setting Click the Dry button to set the Mix to 100 dry Click the Wet button to set the Mix to 100 wet Output The Output sect
447. tude zone adjustment Crossfading or hard switching of replacement audio in different amplitude zones for optimum realism and flexibility Online help Audio Replacement Techniques Replacing audio elements during the course of a re cording session is a fairly common scenario In mu sic production it is often done in order to replace or augment an element that lacks punch In film or video post production it is typically done to im prove or vary a specific sound cue or effect In the past engineers and producers had to rely on sampling audio delay lines or MIDI triggered audio samplers methods that had distinct disadvan tages Delay lines for example support only a sin gle replacement sample and while they can track the amplitude of the source events the replacement sample itself remains the same at different ampli tude levels SoundReplacer Chapter 58 SoundReplacer 336 The result is static and unnatural In addition to these drawbacks sample triggers are notoriously difficult to set up for accurate timing Similarly with MIDI triggered samplers MIDI timing and event triggering are inconsistent result ing in problems with phase and frequency response when the original audio is mixed with the triggered replacement sounds The SoundReplacer Solution SoundReplacer solves these timing problems by matching the original timing and dynamics of the source audio while providing three separate ampli tude
448. udio production applications ranging from pitch correction of musical material to sound design Pitch II supports 44 1 kHz 48 kHz 88 2 kHz and 96 kHz sample rates for all plug in formats Pitch II also supports 176 4 kHz and 192 kHz sample rates for Native and AudioSuite plug in formats Pitch II is available in mono mono to stereo and stereo channel formats Pitch II stereo Chapter 23 Pitch II 134 Pitch processing typically uses the technique of varying sample playback rate to achieve pitch transposition Changing audio sample playback rate results in the digital equivalent of vari speed ing with tape This is usually unsatisfactory since it changes the overall duration of the material Pitch transposition with Pitch II involves a much more complex technique Pitch II digitally re aligns portions of the re sampled audio waveform itself while using de glitching crossfades to mini mize undesirable artifacts The result is a processed signal that is transposed in pitch but still retains the same overall duration of the original unprocessed signal Pitch II Controls Input and Transient Controls Input The Input control lets you attenuate the gain of the input signal from 6 0 dB to 0 dB to prevent clipping in the pitch shift algorithm Input Polarity The Input Polarity button inverts the polarity of the input signal to help compensate for phase anomalies occurring in multi microphone environments or because of m
449. ulation resonation and sample amp hold which are typically found on older modular analog synthesizers Sci Fi is ideal for adding a synth edge to a track Sci Fi Controls Sci Fi Input Level Input Level attenuates signal input level to the Sci Fi processor Since some Sci Fi controls such as Resonator can cause extreme changes in signal level adjusting the Input Level is particularly use ful for achieving unity gain with the original signal level The range of this control is from 12 dB to 0 dB Sci Fi Effect Types Sci Fi provides four different types of effects Ring Mod Is a ring modulator which modulates the signal amplitude with a carrier frequency pro ducing harmonic sidebands that are the sum and difference of the frequencies of the two signals The carrier frequency is supplied by Sci Fi itself The modulation frequency is determined by the Ef fect Frequency control Ring modulation adds a characteristic hard edged metallic sound to audio Sci Fi Chapter 40 Sci Fi 233 Freak Mod Is a frequency modulation processor that modulates the signal frequency with a carrier frequency producing harmonic sidebands that are the sum and difference of the input signal fre quency and whole number multiples of the carrier frequency Frequency modulation produces many more sideband frequencies than ring modulation and an even wilder metallic characteristic The Ef fect Frequency control determines the modulation f
450. une Meter Selector The Meter selector lets you use a standard needle style meter or a strobe style display To select the Meter display Select Needle or Strobe from the Meter selector Strobe Display The Strobe display scrolls to the left when the tuned note is flat and to the right when the tuned note is sharp When the tuned note is close to the target note the strobe slows to a stop The information display shows the exact number of cents sharp or flat from the target note InTune Reference Frequency Control You can adjust the tuning reference frequency us ing the arrows inside the information display By default reference frequency is A 440 Hertz InTune Note Buttons The Note buttons provide two functions When in automatic mode clicking on a note but ton will turn off automatic mode and InTune will now display pitch relative to the selected note only When a tone is selected in the test tone menu clicking on a note button will play a test tone for that note Click the note button again to turn off the test tone The number of note buttons will depend on the pre set selected The default chromatic preset will dis play all twelve notes A preset for a six string guitar will only display six notes Octave Buttons The octave range of 0 8 displayed in InTune is based on middle C being equal to C4 In chromatic presets you can select a tuning octave from 0 8 us ing the arrows at each en
451. upport for hardware and software it has tested and approved For complete system requirements and a list of Avid qualified computers operating systems hard drives and third party devices visit www avid com compatibility Third Party Plug In Support For information on third party plug ins for Pro Tools systems refer to the documentation that came with your plug in Convention Action File gt Save Choose Save from the File menu Control N Hold down the Control key and press the N key Control click Hold down the Control key and click the mouse button Right click Click with the right mouse button User Tips are helpful hints for getting the most from your system Important Notices include information that could affect your data or the performance of your system Shortcuts show you useful keyboard or mouse shortcuts Cross References point to related sections in this guide and other Avid documentation Chapter 1 Audio Plug Ins Overview 6 About www avid com The Avid website www avid com is your best on line source for information to help you get the most out of your Pro Tools system The following are just a few of the services and features available Product Registration Register your purchase online Support and Downloads Contact Avid Customer Success technical support download software updates and the latest online manuals browse the Compatibility documents for system requirements searc
452. ured in the early 1960s the LA 2A achieved wide acclaim for its smooth com pression action and extremely high quality audio signal path The BF 2A has been meticulously crafted to capture every nuance of the legendary LA 2A tube studio compressor providing the most authentic vintage compression sound available Originally designed as a limiter for broadcast au dio a Comp Limit switch was added to LA 2A compressors after serial number 572 The subse quent addition of a Comp Compress setting made the LA 2A even more popular for use in audio pro duction However the switch was inconveniently located on the back of the unit next to the terminal strips and tube sockets in the original version In the BF 2A plug in the switch has been placed on the front panel where you can make better use of it The heart of the LA 2A is its patented T4B Electro Optical Attenuator which provides the compression action The T4B consists of a photo conductive cell which changes resistance when light strikes it It is attached to an electrolumines cent panel which produces light in response to voltage Audio voltage is applied to the light source and what happens as the audio converts to light and back to voltage gives the LA 2A its unique compression action BF 2A preserves all the subtle characteristics of this unique electronic circuit After compression gain brings the signal back to its original level The LA 2A s gain comes from a tube a
453. useful in quickly determining the appropri ate level for mixing and mastering When used in real time the timecode information displayed in the browsers should be ignored Historical Metering To gain an overall picture of the levels in an entire session MasterMeter can be inserted on a Master Fader track and the entire session played from be ginning to end This is typically done during final mix and mastering When session playback is complete MasterMeter shows historical peak and event information for the entire session as well as a historical list of events in the browsers for both signal clips and oversampled clips You can then manually examine the relevant parts of the session using the timecode listed in the browsers to determine any appropriate corrective actions Chapter 56 MasterMeter 330 MasterMeter Controls and Displays MasterMeter Browsers Signal Clip Events Browser The Signal Clip Events browser displays historical clip events from the current session The columns displayed show the relevant timecode for the be ginning and ending of a clip event When used in a stereo track the first column shows L or R to indi cate if the left or right channel has clipped The Min and Max values in this browser will always be zero unless the Clip level is set below zero The contents of this browser can be sorted in ascending and de scending order by any column simply by clicking on a column one or more times T
454. ut on page 258 2 For the next steps hold your guitar but don t play it and be sure to leave its volume up You should hear only the noise that we ll soon get rid of 3 To make it easier to hear the effect begin by set ting the Release to its middle 12 o clock posi tion 4 Now raise the Threshold control to its highest setting fully clockwise so that the Gate fully closes you shouldn t hear anything coming through Eleven 5 Slowly lower the Threshold control until the Gate opens again to find the cutoff or thresh old of the noise 6 Raise the Threshold control again slightly in creasing it only enough to once again silence the noise hold Command Mac or Ctrl Win while adjusting to be able to fine tune the setting in tenths of a dB Now you re in the ballpark 7 If you lowered the Release setting as suggested in step 3 make sure to return it to its maximum setting fully clockwise before continuing Amp Types The Amp Type selector lets you choose an amp Available Amp Types in Eleven include the fol lowing 59 Tweed Lux 59 Tweed Bass 64 Black Panel Lux Vibrato 64 Black Panel Lux Normal 66 AC Hi Boost 67 Black Panel Duo 69 Plexiglas 100W 82 Lead 800 100W 85 M 2 Lead 89 SL 100 Drive 89 SL 100 Crunch 89 SL 100 Clean 92 Treadplate Modern
455. ut out by each amp and the way the cabinet handles responds to that particular type of signal Though subtle the effect of this is a unique cabinet resonance In each Eleven plug in you insert on a track the currently selected Amp Type has a similar effect on the sound of its current cabinet even when the amp section itself is bypassed This does not mean that the bypassed amp set tings affect the cabinet tone only the chosen amp type This could bring just the right amount of extra low low mid or mid range response to the cabinet Setup for blending cabinets Amp on Amps bypassed Cabs on Cab bypassed Different amps can also have a different num ber of stages which can affect polarity See Phase Considerations with Blending in Eleven on page 277 for more information Chapter 44 Eleven 276 How Do I Use This Here are a few suggested ways you can pair Eleven s amps and cabinets To accurately capture the sound of one amp split to and driving multiple cabinets make sure the same Amp Type is selected in all the Eleven plug ins all the cabinets as well as the active amp For maximum variety mix and match bypassed amps with active cabinets For realism with the combo amps such as the Tweed Lux and AC Hi Boost make sure to use their default cabinets Blending Eleven Amps You can easily set up tracks and Eleven for multi amp setups To blend multiple amps 1 Set up tracks
456. utput Level Meters Stereo mode Chapter 5 Focusrite D2 28 Frequency Display The frequency display is a visual representation of the current EQ settings As you adjust the controls of any currently active filter the display plots the changes to the EQ curve in real time If you are us ing D2 in stereo the frequency display shows the EQ curve for the right channel in red and the left channel in blue EQ Filter Controls Each of the six different EQ filters has its own con trols and its own icon The icons act as three state switches for enabling disabling or bypassing the specific filter The current state of a filter is indi cated by its color White enabled In this state the filter is active audible and using available DSP resources Black disabled In this state the filter is not us ing any DSP resources and has no effect on au dio Gray bypassed In this state the filter is not ac tive but is still using available DSP resources The effect of the filter is not audible Cartesian Graph To reset all D2 controls to their default settings Option click Mac or Alt click Windows the frequency display To reset controls for both channels when in Stereo mode Option Shift click Mac or Alt Shift click Windows the frequency display Chapter 5 Focusrite D2 29 High Pass Filter The 18 dB octave High Pass Filter provides a ro tary control for adjusting the corner cutoff fre qu
457. vidual controls for early and late reflections and indepen dent front rear and center levels for surround out puts Input Cuts or boosts the input signal level from inf dB to 12 dB Output Cuts or boosts the output signal level from inf dB to 12 dB Early Cuts or boosts the levels of the early reflec tions from inf dB to 12 dB Late Cuts or boosts the levels of the late reflections from inf dB to 12 dB Front Rear Center In quad and 5 0 channel output modes Space provides additional controls to atten uate or boost the Front left and right Rear left and right and Center 5 0 only signal levels from inf dB to 12 dB In 5 0 output mode the level of the center channel is affected by both the Front and Center controls Space Delay Controls The Delays group provides controls for the delay timings of the reverb When changes are made to any control in the Delays group the IR waveform is recalculated and displayed in the Waveform dis play Pre Delay Adjusts length of the Pre Delay from 200 to 200 ms The Pre Delay is the time be tween the direct sound and the first reflection In creasing the Pre Delay often changes the perceived clarity of audio such as vocals Pre Delay adjusts the delay of the overall impulse and affects both the Early and Late portions of the IR equally Pre Delay can be set to negative values to allow for subtle or radical changes to the reverb For exam ple a small
458. w 2 Choose AudioSuite gt Other gt Pro Limiter Loudness Analyzer 3 Click Analyze The Loudness numerical displays update to show the analyzed values for information on the Loud ness numerical displays see Pro Limiter Loud ness Numeric Displays on page 122 Pro Limiter Loudness Analyzer AudioSuite plug in With the Pro Limiter Loudness Analyzer the Preview and Render buttons do not do any thing useful They are simply present as part of the AudioSuite plug in framework Chapter 21 Purple Audio MC77 125 Chapter 21 Purple Audio MC77 Purple MC77 is a dynamics processing plug in that is available in DSP Native and AudioSuite for mats Purple MC77 supports 44 1 kHz 48 kHz 88 2 kHz 96 kHz 176 4 kHz and 192 kHz sample rates Purple MC77 operates as a mono multi mono or stereo plug in The Purple Audio MC77 is a spot on digital replica of Andrew Roberts acclaimed MC77 Limiting am plifier which in turn is an update of his classic MC76 hardware unit Representing a different take on the 1176 style FET limiter the Purple Audio MC77 preserves every audio nuance and sonic sub tlety of the classic originals Purple Audio MC77 Controls Purple Audio MC77 has controls identical in name to those of the BF76 and which function similarly For more information see Chapter 10 BF76 Purple Audio MC77 Chapter 22 Smack 126 Chapter 22 Smack Smack is a dynamics processing plug in tha
459. w pass and high pass filters are in Bypass mode when the 6 Band EQ is first opened Focusrite D2 Chapter 5 Focusrite D2 27 D2 Controls Input Level Input Level allows you to attenuate signal input level to the D2 The range of this control is from 18 dB to 12 dB When you use D2 in stereo each channel has its own separate Input Level knob To adjust input lev els for both channels simultaneously select the Link button then drag either knob Output Level Output Level allows you to adjust the overall out put gain The range of this control is from 18 dB to 12 dB When you use the D2 plug in in stereo each chan nel has its own separate output level knob To ad just output levels for both channels simultaneously select the Link button Meters The D2 high resolution plasma style meters indi cate signal levels and detect clipping at the input algorithm or output stage When D2 is used in ste reo two meters appear one for each channel A Clip Indicator is located above each meter It in dicates clipping by increasing its brightness as suc cessive samples are clipped Click the Clip Indica tor to clear it Option clicking Mac or Alt clicking Windows clears both channels when D2 is used in stereo The following metering indications are used Green nominal levels Yellow pre clipping at 6 dB below full scale signal Red full scale signal clipping input Level O
460. walls floor and ceiling of the cathedral The tim ing of each reflection will vary on the size of the room but they will always arrive after the direct sound For example the reflection from the floor typically occurs first followed by the ceiling and the walls The initial reflections are known as early reflections and are a function of the reflective sur faces the position of the audio source and the rela tive location of the listener A small room may have only a fraction of a second before the first reflections whereas large spaces may take much longer The elapsed time of the early reflections defines the perceived size of the room from the point of view of a listener Space of fers various controls over early reflection parame ters Chapter 30 Space 184 The time delay between the direct sound and the first reflection is usually known as pre delay Space lets you adjust pre delay Increasing the pre delay often changes the perceived clarity of au dio such as vocals Reflections continue as the audio reaches other sur faces in a space and they create more reflections as the sound waves intermingle with one another be coming denser and changing in character depend ing on the properties of the room As the room ab sorbs the energy of the sound waves the reverb gradually dies away This is known as the reverb tail and may last anywhere up to a minute in the very largest of spaces The reverb tail will often vary
461. watt tube amplifier feeds a speaker crossover which splits the signal All frequencies below 800 Hz go to a 15 bass speaker and all frequencies above 800 Hz go to a compression horn driver The large bass speaker is bolted to the cabinet and a foam drum directly below the speaker reflects the bass outward For the high frequencies a treble horn with two bells reflects the sound from the compression horn driver located below Only one bell actually produces sound the other is merely a counterbalance Voce Spin Lower speaker assembly Rotation Direct Sound 15 Low Frequency Loudspeaker Scoop Chapter 41 Voce Plug Ins 237 Then of course it spins Separate belts pulleys and motors drive the upper treble horn and the lower foam drum Adding to the effect the upper horn and lower drum spin in opposite directions Most rotating speakers feature two sets of motors allow ing both slow Chorale and fast Tremolo ro tation speeds Voce Spin Controls Of course all that motion creates a rich sound but then you have to capture it using microphones Spin provides fifteen classic recording setups to choose from giving you the sounds you ve heard on countless records instantly Just choose a preset and click Chorale Tremolo or Off Alternately click and drag the flip switch Short flicks of the wrist land on Off longer flicks toggle between Chorale and Tremolo You may also Alt click
462. waveform of lower frequency Tremolo is not available on all amps Tremolo Speed The Speed control sets the rate of the Tremolo effect The Tremolo Speed LED pulses at the rate of Tremolo Speed The default setting is 5 0 Tremolo Depth The Depth controls the amount of the Tremolo effect The default setting for this con trol is 0 0 which is equivalent to off Some amp models call the Tremolo Depth control Intensity Some might assume a Master volume knob capable of silencing the amp completely Not so Use the Output knob in the Master sec tion to silence the output of the plug in Use Master volume for tone and distortion Eleven does not support Tempo Sync Chapter 44 Eleven 268 Eleven Cabinet Types The Cab Type selector lets you pick a cabinet to use with the current amp The selected cabinet and its controls are displayed directly below the amp con trols Available cabinets include the following 1x12 Black Panel Lux 1x12 Tweed Lux 2x12 AC Blue 2x12 Black Panel Duo 4x10 Tweed Bass 4x12 Classic 30 4x12 Green 25W These models only appear in the full version of Eleven Cabinets are listed by their number and diameter of their speakers For example 1x12 means a cabi net has a single 12 inch speaker Pairing Amps and Cabinets Eleven lets you combine amps and cabinets in tra ditional pairings such as the combo amps through their default cabinets
463. which Space can be used during a Pro Tools ses sion Using Space Presets Space ships with a selection of factory presets for different reverb sounds The presets are designed to give a sample of the various IRs available from the Plug In Presets selector in conjunction with various reverb settings However the presets do not cover the entire IR library Using Space on an Effect Send When Space is used on an Aux Input track as an ef fects send the Dry control should be set to inf dB Automating Space Snapshots Snapshot automation is a powerful method of changing the reverb parameters without having to individually automate each parameter To automate Space Snapshots 1 Insert Space on a track 2 Select Snapshot mode 3 Load an IR into each Snapshot and make any de sired changes to specific Space controls 4 Name each Snapshot as desired 5 Click Auto 6 Add Snapshot to the list of automated controls 7 Select Space gt Snapshot from the automation menu for the track 8 Draw the desired automation on the track with the Pencil tool The names displayed in the auto mation track will match the names entered for each Snapshot If the waveform is reversed using the Reverse control effected audio may continue to play for several seconds after the transport is stopped or audio input finishes Decay controls Chapter 30 Space 202 Space IR Library Categories Space includes an extensive impulse respo
464. www avid com After downloading make sure the installer is uncompressed dmg 2 Ensure that Pro Tools is already installed and has been launched at least once on your com puter 3 If Pro Tools is running quit Pro Tools 4 Locate and double click the plug in installer disk image 5 Drag the plug in aaxplugin to the Plug Ins folder alias in the disk image Installing Paid Plug Ins on Windows To install a plug in on Windows 1 Download the installer for Windows from www avid com After downloading make sure the installer is uncompressed ZIP 2 If Pro Tools is running quit Pro Tools 3 Locate and double click the plug in installer 4 Follow the on screen instructions to complete the installation 5 When installation is complete click Finish Removing Plug Ins If you need to remove a plug in from your Pro Tools system follow the instructions below for your computer platform Removing Plug Ins on Mac To remove a plug in 1 Locate and open the Plug Ins folder on your Startup drive Library Application Support Avid Audio Plug Ins 2 Do one of the following Drag the plug in to the Plug Ins Unused folder Drag the plug in to the Trash and empty the Trash Removing Plug Ins on Windows To remove a plug in 1 Choose Start gt Control Panel 2 Click Programs and Features 3 Select the plug in from the list of installed applications 4 Click Uninstall 5 Follow the on sc
465. xample if Decay Ratio is set to 4 the reverb time is increased by a factor of 4 when the signal is above the threshold level If the ratio is 0 25 reverb time is increased by a factor of 4 when the signal is below the Threshold level Threshold Sets the input level above or below which reverb decay time will be modified Chorus Controls The Chorus section has controls for setting the depth and rate of chorusing applied to a reverb tail Chorusing thickens and animates sounds by adding a delayed pitch modulated copy of an audio signal to itself Chorusing produces a more ethereal or spacey re verb character It is often used for creative effect rather than to simulate a realistic acoustic environ ment Depth Controls the amplitude of the sine wave generated by the LFO low frequency oscillator and the in tensity of the chorusing The higher the setting the more intense the modulation Rate Controls pitch modulation frequency The higher the setting the more rapid the chorusing Setting the Rate above 20 Hz can cause frequency modula tion to occur This will add side band harmonics and change the reverb s tone color producing some very interesting special effects Reverb One Reverb Section Controls The Reverb section has controls for the various re verb tail elements including level time attack spread size diffusion and pre delay These deter mine the overall character of the reverb Level Con
466. xim Controls and Meters 93 Using Maxim 96 Maxim and Mastering 96 Contents vi Chapter 18 Pro Compressor 97 Pro Compressor Metering 97 Pro Compressor Input Section 98 Pro Compressor Output Section 99 Pro Compressor Dynamics Graph 100 Pro Compressor Controls 102 Pro Compressor Side Chain Processing 104 Chapter 19 Pro Expander 107 Pro Expander Metering
467. y the Q setting The Low Pass and High Notch EQ controls and their corresponding graph elements are displayed on screen in gray The following control values are available High Pass filter left and Low Notch EQ right Control Value Frequency Range 20 Hz to 8 kHz Frequency Default 20 Hz HPF Slope Values 6 12 18 or 24 dB oct Low Notch Q Range 0 1 to 10 0 Low Notch Q Default 1 0 High Pass Filter button Frequency control Slope control Frequency control Q control Band Enable button Low Notch EQ button Band Enable button Low Pass filter left and High Notch EQ right Control Value Frequency Range 120 Hz to 20 kHz Frequency Default 20 kHz HPF Slope Values 6 12 18 or 24 dB oct High Notch Q Range 0 1 to 10 0 High Notch Q Default 1 0 Low Pass Filter button Frequency control Slope control Frequency control Q control Band Enable button High Notch EQ button Band Enable button Chapter 4 EQ III 22 7 Band EQ III Low Shelf Low Peak The Low Shelf Peak band is switchable between low shelf EQ and low peak EQ functions By de fault this band is set to Low Shelf Low Shelf EQ Boosts or cuts frequencies at and be low the Frequency setting The amount of boost or cut is determined by the Gain setting The Q setting determines the shape of the shelving curve Low Peak EQ Boosts or cuts a band of frequencies centered around the Frequenc
468. y setting The width of the affected band is determined by the Q setting The Low Shelf and Low Peak Gain controls and their corresponding graph elements are displayed on screen in red The following control values are available 7 Band EQ III Low Mid Peak The Low Mid Peak band boosts or cuts frequencies centered around the Frequency setting The width of the band is determined by the Q setting The Low Mid Gain control and its corresponding graph elements are displayed on screen in brown Low Shelf EQ left and Low Peak EQ right Low Shelf EQ button Frequency control Band Enable button Gain control Q control Low Peak EQ button Frequency control Band Enable button Gain control Q control Control Value Frequency Range 20 Hz to 500 Hz Frequency Default 100 Hz Low Shelf Q Range 0 1 to 2 0 Low Peak Q Range 0 1 to 10 0 Q Default 1 0 Low Shelf Gain Range 12 dB to 12 dB Low Peak Gain Range 18 dB to 18 dB Low Mid Peak EQ Frequency control Band Enable button Gain control Q control Chapter 4 EQ III 23 The following control values are available 7 Band EQ III Mid Peak The Mid Peak band boosts or cuts frequencies cen tered around the Frequency setting The width of the band is determined by the Q setting The Mid Gain control and its corresponding graph elements are displayed on screen in yellow The following control values are available
469. ynamic range so that there is less variation be tween loud and soft hits This is useful if the dy namics of the source material are too extreme The Dynamics button provides a quick means of toggling on and off the Dynamics control When Dynamics is toggled off SoundReplacer will not track changes in the source audio file s dynamics Audio events in the resulting replacement audio file will uniformly be at the amplitude of the re placement samples themselves with no variation in dynamics SoundReplacer Online Help To use online help hold the pointer over any control and an explanation will appear Using SoundReplacer Following are basic guidelines for using SoundReplacer effectively Also see Getting Op timum Results with SoundReplacer on page 340 To use SoundReplacer 1 On the source track select the audio you want to replace Only selected audio will be replaced 2 Choose SoundReplacer from the AudioSuite menu 3 Click the Load Sound icon the icon beneath the yellow slider to import the replacement sound for amplitude zone 1 4 Locate an audio file and click Open 5 Adjust the amplitude zone slider 6 Repeat steps 3 5 to load replacement sounds into amplitude zones 2 and 3 7 To align the amplitude peak in the replacement file s to threshold trigger markers in the source audio enable Peak Align 8 Click Preview to audition the replacement audio 9 Adjust the Threshold sliders to fine tun
470. ynamics III Key Input for Side Chain Processing To use a filtered or unfiltered external key input to trigger dynamics processing 1 Click the Key Input selector and select the input or bus carrying the audio from the reference track or external audio source 2 Click External Key to activate external side chain processing 3 To listen to the signal that will be used to control side chain input click Side Chain Listen to en able it highlighted 4 To filter the key input so that only specific fre quencies trigger the plug in use the HF and LF controls to select a frequency range 5 Begin playback The plug in uses the input or bus that you chose as an external key input to trigger its effect 6 Adjust the plug in s Threshold Thresh control to fine tune external key input triggering Using a Filtered Input Signal for Side Chain Processing with Dynamics III To use the filtered input signal to trigger dynamics processing 1 Ensure the Key Input selector is set to No Key Input 2 Ensure that the External Key button is disabled dark gray Selecting a Key Input External Key Side Chain Listen Key Input selector Side Chain section Chapter 12 Dynamics III 74 3 To listen to the signal that will be used to control side chain input click Side Chain Listen to en able it highlighted 4 To filter the side chain input so that only specific frequencies within the input signal trigger the plug in us
471. you can Process incoming audio signals with plug ins Automate volume pan and plug in controls Bounce To Disk Take advantage of the audio outputs of your Pro Tools audio interfaces ReWire Pro Tools does not support sending audio to ReWire client applications Not all ReWire client applications support automatic launch from a ReWire mixer ap plication For these applications launch the ReWire client app separately and then select it as a plug in insert in Pro Tools Exchange of additional metadata such as controller and note names between Pro Tools and ReWire clients is not sup ported Chapter 54 ReWire 311 Audio and MIDI signal flow between Pro Tools and a ReWire client application Reason shown Audio from ReWire client Reason to Pro Tools MIDI from Pro Tools to ReWire client Reason MIDI from ReWire client Reason to Pro Tools Chapter 54 ReWire 312 ReWire Requirements To use the ReWire plug in you will need An Avid qualified Pro Tools system 64 bit ReWire compatible client software such as Reason from Propellerheads Software ReWire support is also under development for other third party companies For availability check with the manufacturer or visit the Avid website www avid com Track Count with Pro Tools HD With Pro Tools HD the ReWire plug in can be in serted on any kind of track Each channel of audio transmitted through ReWire then uses the sam
472. z sample rates D3 operates as a mono multi mono or stereo plug in D3 features include Compressor Limiter This configuration allows you to use both the compressor and the limiter at the same time The Compressor Limiter plug in requires twice as much DSP as the Compres sor Limiter Compressor Limiter This configuration allows you to use either the compressor or the limiter but not both at the same time The Compressor Limiter plug in uses half as much DSP as the Compres sor Limiter It is provided so that you can conserve DSP since you may not need both compression and limiting at the same time The Compressor Limiter defaults to the compres sor being enabled and the limiter disabled D3 Compressor The D3 compressor reduces the dynamic range of audio signals that exceed a user selectable thresh old by a specific amount This is accomplished by reducing output levels as input levels increase above the threshold The amount of output level reduction that D3 ap plies as input levels increase is referred to as the compression ratio This parameter is adjustable If you set the compression ratio to 2 1 for example for each 2 dB that the signal exceeds the threshold the output level will be reduced to 1 dB above the threshold With a compression ratio of 4 1 an 8 dB increase in input will produce only a 2 dB increase in output Focusrite D3 Chapter 14 Focusrite D3 79 D3 Limiter The D3 limiter o
473. zones per audio event This lets you trigger different replacement samples according to perfor mance dynamics Each replacement sample is assigned its own ad justable amplitude zone Variations in amplitude within the performance determine which sample is triggered at a specific time For example you could assign a soft snare hit to a low trigger threshold a standard snare to a medium trigger threshold and a rim shot snare to trigger only at the highest trigger threshold Replacement samples that are triggered in rapid succession or in close proximity to each other will overlap naturally avoiding the abrupt sound trun cation that occurs on many samplers In addition to its usefulness in music projects SoundReplacer is also an extremely powerful tool for sound design and post production Morphing gun shots changing door slams or adding a Dop pler effect can now be accomplished in seconds rather than minutes with sample level precision Replacement audio events can be written to a new audio track or mixed and re written to the source audio track Sample thresholds can be amplitude switched between the replacement samples or am plitude crossfaded for seamless transitions SoundReplacer Controls SoundReplacer Waveform Display The waveform display shows the audio that you have selected for replacement When you select au dio on the source track then open Sound Replacer the audio waveform will automatically be dis

Audio Plug-Ins Guide v11.2

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