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1. 0000 31 57 70 87 Reverb algorithm 0ee 56 63 Reverb decay time eee 55 63 Reverberation 54 97 103 109 RMS Mode siria tds 32 S Sampling rate eener 113 SAVE prese minsin yaan e auaa 19 Selector DULLON ooccoccccccccnocccnnccnnnnnnos 14 Shelf filterna a a n Sa 27 SIDECHAIN ciirsa a an 34 43 SL aii 62 63 120 Se Sii ais 15 SIOPE nata ta traia 100 T A vest eane a a aka 37 40 TLOXE CNY ici 13 Threshold 31 57 73 84 86 Time response cooocccccnccnccnncnnnnnnnnannnnnns 60 Tools MONU nssssssssssssssssserrrrrrrrrerrrrren 20 WPA CKPIU Darna a a emeestions 23 TEUNCA ION cita aaa eee des 71 U OO Tani EEn nE iad AE vacant 17 User ManUal oooocccnccnccncconnnnnnnancnnnnnnos 21 User presetan rs nates 19 V Vintage Mode ooccccccccnccnncnnncnnncnnnannnaos 33 Ww Wet delay rires aaa 64 Wet dry MIX ccc cee cee eect eee ee teen nena 64 O thn tibody ttma stele tad 107 Index 121
2. Click on the knob Click on the knob drag up down drag up down Shift click drag Right click drag up down up down Or Shift click drag up down Increase Decrease a parameter value rotate clockwise counterclockwise Fine adjustment increase decrease PT Command click PT Ctrl click Control click Or Double click Command click Or Double click Reset knob to default value PT Option click PT Alt click 2 2 Text Entry 20 0 dj Many value displays are editable text A text field is editable if your mouse cursor changes to an beam when moved over the text Following is a table that fully describes how to use the text editing features Function Mac Enter text entry mode Click in the display Select text Click drag Select entire text Double click Delete character to left of cursor Delete Delete character to right of cursor Click in the display Click drag Double click Backspace 13 Wave Arts Power Suite Move the cursor left right Left Right arrow Left Right arrow keys keys Shift click drag Shift click drag or or AAN Shift left right Shift left right arrow keys arrow keys ESC ESC or or Click outside value Click outside value Exit text entry mode pox por or or Tab Tab or or Return Enter Return E
3. RTA Post EQ to run the RTA after the EQ processing is applied Referring to the TrackPlug routing diagram the EQ can run either before or after the compressors based on the EQ pre post mode Because the RTA can run either just before or just after the EQ there are four possible insert points for the RTA When the EQ display is set to one of the dynamics sidechains the RTA runs either just before or just after the corresponding sidechain EQ In this case the Pre mode is useful for seeing what frequencies can be used to trigger the dynamics processors for example to visualize the frequency range of a vocal ess sounds before setting the sidechain EQ parameters Show Hide RTA axis Select Show RTA Axis to display the RTA vertical axis instead of the EQ axis The RTA axis is drawn in blue to distinguish it from the white EQ axis In this mode when an EQ edit is made the axis temporarily reverts to the EQ axis during the edit then changes back to the RTA axis when the edit is complete Select Hide RTA Axis to hide the RTA axis at all times only the EQ axis will be displayed 39 Wave Arts Power Suite Dynamics TrackPlug s dynamics user interface section is shown above At the top are the three overview sections for the Gate Comp1 and Comp2 Dynamics interface overview section Each overview section shows the enable button the preset control input and gain meters and a threshold slider control The overview
4. Wet Dry Mix 0 to 100 Late Early Mix 0 to 100 Output Gain 18 to 18 dB Diffusion 0 to 100 5 5 Parameter Descriptions Following is an explanation of MasterVerb s parameters Enable Turns MasterVerb on and off This internal parameter is controlled by the Bypass button in the MasterVerb menu bar When the Bypass button is lit the internal enable is off and vice versa Reflection Type Sets the early reflection pattern The early reflection pattern consists of a set of discrete echoes typically spanning no more than two hundred milliseconds A number of reflection patterns are provided including halls rooms chambers and plates The halls and rooms are reflection patterns derived from actual acoustical spaces The chambers and plates are dense collections of reflections derived from synthetic responses The Anechoic type is a single unit gain echo at time 0 which passes the input signal unchanged althoughit is still processed by the diffuser Other special effect echo types are included Reflection Size Scales the size of the early reflection pattern ranging from x0 5 50 to x1 5 150 Smaller sizes will sound like a smaller space and larger sizes will sound like a larger space Early Damping Parameters The early damping EQ affects all signals going into the wet path of the reverb therefore it is used to change the equalization of both the early reflections and the late reverb It could also be called Wet Equali
5. General Drums Guitar Default Industrial Kick Another Top Ten Kick Drum Punchy Kick Tight Bright Snare Good Ol Snare Hit Snare Big Attack Snare Snare 1 Great Toms Beastly Drum Loop Beastly Drum Loop 2 Rock Drum Room 1 Rock Drum Room 2 Distorted Drum Loop Nook the Drum Room Filter Guitar Funk From Pod to Metal God Acoustic Gtr Teeny Toy Guitar Ukelelephone Bass Bass Channel Walkin Jazz Bass Keyboard Piano Pop Jazz Piano Ballad Horns Sax Solo Vocals Vocal Showstopper Warm Rap Control Angry Rapper Songstress in my Ear Sexy Pop Diva Female R B Ballad 48 4 TrackPlug Effected Vox 1 Mix Aggressive Rock 2 Mix Very Nice 2 Mix Compress TrackPlug also ships with TrackPlug v4 presets listed below Category Preset name Drums Drums v4 Drums Squisher v4 Drums Squasher v4 Drums Tighter v4 Kick v4 Snare 2 v4 Snare 3 v4 Toms v4 Guitar Hi hats Cymbals v4 Acoustic Guitar v4 Acoustic Gtr 2 Bass Bass v4 Bass Acoustic v4 Bass Attack v4 Bass Attackier v4 Horns Keyboard Piano v4 Horns v4 Sax v4 Vocals Vocal Female v4 Vocal Male v4 Voiceover Female v4 Voiceover Male v4 Mix Mix Dance v4 Mix Rock v4 Master Plug v4 10 Ba
6. by the head Both ear signals are also subject to a complicated filtering process caused by acoustical interaction with the torso head and in particular the pinna external ear The various folds in the pinna modify the frequency content of the signals reinforcing some frequencies and attenuating others in a manner that depends on the direction of the incident sound Thus an ear acts like a complicated tone control that is direction dependent We unconsciously use the time delay amplitude difference and tonal information at each ear to determine the location of the sound These indicators are called sound localization cues 8 2 2 Head Related Transfer Functions HRTFs The transformation of sound from a point in space to the ear canal can be measured accurately the measurements are called Head Related Transfer Functions HRTFs The measurements are usually made by inserting miniature microphones into the ear canals of a human subject or a manikin A measurement signal is played by a loudspeaker and recorded by the microphones The recorded signals are then processed by a computer to derive a pair of HRTFs for the left and right ears corresponding to the sound source location This process is diagrammed in the figure below 92 8 Panorama microphones in ears speaker HRTF HRTF Figure 8 1 HRTF measurement for a single source location Each HRTF typically consisting of several hundred numbers describes the time
7. compress a signal However short release times also cause the gain to modulate more deeply during steady audio signals causing distortion This is because with aggressive settings the peak limiter will be compressing each oscillation of the input signal ducking the gain to allow the peak oscillation to pass then quickly restoring the gain as the signal passes through 0 70 6 FinalPlug In contrast long release times prevent distortion during steady audio signals This is because the gain will be ducked on the onset of the signal and will stay ducked because the release happens so slowly Hence the amount of gain modulation with each oscillation of the input is very small and hence the distortion is greatly reduced compared to using a fast release time However the disadvantage of using a long release time is that it takes a long time for the gain to increase after loud sounds in the input This can cause an audible pumping of the background level of the audio each peak in the audio causes the background to be reduced in volume and then the background level increases slowly until the next peak Furthermore long release times do not permit aggressive compression FinalPlug features an auto release mode which dynamically chooses the optimum release time based upon the program content FinalPlug chooses the release time based on the crest factor of the input signal The crest factor is an indicator of the peakiness of a signal it is obtained
8. contribution from the indicated image source Additional image sources are shown that 98 8 Panorama correspond to other reflective paths between the source and listener First order image sources are labeled X1 second order image sources are labeled X2 Based on the positions of the source listener and the reflective surfaces walls floor ceiling it is easy to use the above geometrical procedure to calculate the time and direction of all early reflections Each reflection can then be rendered using 1 a delay line to delay the sound according to the total travel time along the reflected path 2 an attenuation or filter to approximate the transmission and reflection losses and 3 a binaural synthesizer to properly localize the reflection Panorama models the first order reflections in a rectangular room using the above procedure There are six first order reflections corresponding to the left right front and back walls and the ceiling and floor The ceiling and floor are called the top and bottom walls in Panorama Panorama models the wall reflections using filters that closely approximate the sound absorption properties of common building materials The absorption data were obtained from standard acoustical texts 8 2 11 Late Diffuse Reverberation The early reflection model does not address the late portion of the reverberation which contains millions of reflections traveling in all directions Late reverberation is usua
9. noise shapers push the quantization noise to higher frequencies to which the ear is less sensitive However there are many variations of noise shaping techniques in use The user must decide which shape is best for the task at hand In this spirit FinalPlug provides a wide variety of noise shapes to choose from Some of the shapes are similar to other noise shapers in common use It should be noted that dithering and noise shaping may offer fewer practical benefits than the marketing from many manufacturers would have you believe This is because acoustic recordings typically contain much higher levels of noise than required to properly dither a signal Hence adding dithering will have no effect except to slightly increase the existing noise level A study by D W Fostle Audio Magazine March 1996 showed that very few recorded CDs have low enough noise floors to benefit from proper dithering and noise shaping The advantages of dithering are even more tenuous when considering 24 bit media like DVD audio Dithering and noise shaping will have benefits only when the original material has a noise floor below the quantization level of the final medium This is unlikely when dealing with acoustically recorded material However purely synthetic material more common these days may indeed have very low noise floors and may benefit from proper dithering and noise shaping As shown in the above schematic truncation and dithering occurs after peak l
10. rectangular room The rear wall is drawn in purple the other walls are drawn in blue The listening position which is at the origin of the coordinate system is drawn as three short white lines pointing along the positive X Y and Z axes The source position is drawn as a small blue and white circle Gray crosshair lines are drawn through the source position along each of the axes However the crosshairs 104 8 Panorama are limited to the interior of the room in order to more easily visualize when the source goes outside of the room boundaries Clicking and dragging on the 3D room view allows you to change the viewpoint The 3D room view is useful for visualizing the dimensions of the room as controlled by the reflection parameters described later 8 4 Parameters All Panorama parameters and their possible values are listed below as they would be displayed by a generic parameter value style user interface Parameter name Azimuth Elevation Distance X Y Z Coordinate System Direct Enable Reflection Enable Reverb Enable Direct Gain Reflection Gain Reverb Gain Direct Slope Reflection Slope Reverb Slope Direct Filter Length Reflection Filter Length Left Enable Right Enable Front Enable Back Enable Top Enable Bottom Enable Left Distance Right Distance Front Distance Back Distance Top Distance Bottom Distance Left Material Right Material Front Material Back Material Top Material Values 180 to 1
11. whereas digital audio editors typically use 32 bit floating point sample format containing at least 24 bit sample resolution Hence the 24 bit resolution samples must be reduced to 16 bit prior to burning the CD This is done by discarding the least significant bits which is the definition of bit depth truncation Truncation will cause the addition of quantization noise and quantization distortion to the resulting signal For low level signals i e signals that are only a few bits in amplitude the quantization distortion can sound very objectionable Dithering is a technique used to eliminate quantization distortion by adding a small amount of random noise to the signal prior to truncation The added noise dithers 71 Wave Arts Power Suite the signal between the various quantized output levels in such a way that the quantization distortion is converted to pure broadband noise Hence dithering adds noise to the signal in order to convert the objectionable quantization distortion to less objectionable noise Dithering is thus converting distortion to hiss FinalPlug uses TPDF triangular probability distribution function noise for dithering Mathematically this is known to be the optimal type of noise to convert quantization distortion to noise that is uncorrelated with the original signal Noise shaping is a technique used to shape the spectrum of the quantization noise in order to make it less perceptible to the human ear Typically
12. Band HiGain 18 to 18 dB Band Ratio 1 to 50 Band Knee O Soft 1 Medium 2 Hard Band Att 0 1 to 1000 msec Band Rel 1 to 2000 msec In the above parameter names Band will display as Band1 through Band6 for the six frequency bands Global parameters Parameter name Values Enable O Off 1 On Output Level 18 to 18 dB Lookahead Delay O Off 1 1 msec 2 2 msec 3 5 msec Crossover Slope O 18 dB oct 1 30 dB oct 2 6 dB oct Compression Mode O Clean 1 Vintage AAX 1 Peak 2 RMS 85 Wave Arts Power Suite 7 5 Parameter Descriptions Band Enable This parameter defines the enable state of each band The user interface handles this internal parameter through the Add Delete Mute and Solo buttons If a band is deleted its Band Enable is set to Off If a band is muted its enable is set to Mute If a band is on and not muted or soloed its enable is set to On If a band is soloed its enable is set to Solo If there are N enabled bands these must occupy bands 1 through N if there is only one band enabled it is band 1 Band 1 may not be deleted Band Lo and Band Hi These parameters define the low and high cutoff frequencies for each band The user interface allows the user to edit these internal parameters through the band edge lines on the frequency response display For N active bands Bandl Lo is always 20 Hz and Band N Hi is always 20 kHz The other bands are constrai
13. Click the button to toggle the state Buttons that do not light up are used to activate certain ijcommands S l Output meters show the peak signal power in short time updates in green from 96dB to 6dB in yellow between 6dB and OaB and in red above OdB Peak hold levels are also drawn The meter also stores the overall peak value for each channel and displays these values in the peak indicator boxes above the meter If the detector finds a peak value above 0dB the text color will turn red as a warning Click on either indicator box to reset them back to 96dB Right click PC or Shift click Mac on an indicator box to automatically increase or decrease the output gain so that the peak will be 0 1dB 15 3 Menu Bar and Preset Manager 3 Menu Bar and Preset Manager All Wave Arts plug ins in the Power Suite Bundle have the following menu bar displayed at the top of the plug in Bypass Undo Copy AAE Default gt Preset Tools This section describes the operation of the menu bar preset manager and the other functions available in the menus 3 1 Bypass Clicking on the bypass button bypasses the effect that is audio will pass through the effect without alteration The button is lit when the effect is bypassed 3 2 Undo Clicking the Undo button causes the parameters to revert to their settings prior to the last edit Only one level of undo is available so clicking the undo button again will restore th
14. For example you may want to compress the vocals in a mix without affecting the low frequencies bass kick drum or high frequencies snare cymbals A multiband dynamics processor can accomplish this by restricting the frequency range of the compression to the mid range vocal frequencies 80 7 MultiDynamics It is also possible to think of multiband dynamics as providing EQ which is dependent on the level of the input sound You may want to reduce the level of a high hat but only when the drums are played softly This can be accomplished by setting up a high frequency band that decreases gain when the input falls below a threshold downward expansion This is more powerful than using a shelving EQ which will change the tonal balance at all levels equally The multiband dynamics processor can have a flat frequency response at high inputs levels and act like a shelving EQ only at low levels Another powerful use of multiband dynamics is noise reduction One can set up multiple bands to do noise gating by setting the gain to decrease when the signal falls below threshold Then the thresholds of each band can be adjusted to be just above the ambient noise level in each band So when the signal in each band is above the ambient noise level it passes through when the signal falls near the ambient noise level it is gated This is a really powerful technique for cleaning up noisy production recordings Multiband dynamics applied to solo inst
15. Q for the vintage shelf the range is 0 707 to 1 414 for the resonant shelf the range is 0 5 to 5 Dynamics Parameters 45 Wave Arts Power Suite The TrackPlug dynamics processor is described in detail in the section on Dynamics earlier in this chapter Dyn Enable When off the dynamics section is bypassed Dyn Knee Sets the dynamics knee shape options are soft medium and hard Dyn Thresh Level above which the compressor is active or below which gate is active Range is 96dB to OdB Dyn Ratio Controls how much gain reduction is applied as input level passes threshold Dyn Gain Dynamics makeup gain Dyn Attack Attack time in msec Dyn Release Release time in msec Dyn Mode Sets the dynamics mode Options are Clean peak Clean RMS Vintage peak Vintage RMS Vintage Warm AAX options are Vintage peak Vintage RMS Vintage Warm Dyn Lookahead Sets the lookahead time Lookahead is useful when gating to provide a bit more time for the gate to restore gain prior to an onset Sidechain Mode Set the sidechain source and mode The sidechain modes are described in detail earlier in this chapter Sidechain Monitor enables sidechain monitoring Sidechain EQ Type Sets the type of the sidechain EQ Options are Lowpass Highpass Bandpass and Notch Sidechain EQ Freq Sets the cutoff or center frequency of the sidechain EQ Sidechain EQ Width Sets the width of the sidec
16. Shape2 Ultra 5 44 1 kHz Shape3 6 44 1 kHz Shape3 Ultra 7 44 1 kHz Shape4 8 44 1 kHz Shapes4 Ultra 9 44 1 kHz Shape5 10 96 kHz 11 96 kHz Ultra 74 6 FinalPlug Mute Dithering O Off 1 On 6 5 Parameter Descriptions The FinalPlug parameters are described below Limiter Enable Turns limiter on and off When off the limiter is bypassed Threshold Sets the level at which the limiter starts to take effect on the signal Ceiling Sets the maximum output level of the signal Generally this is set slightly below 0 0dB the presets use 0 1dB for example Please note that if audio is truncated to a very low bit depth i e 4 or 8 bits it is possible that dither noise can exceed this ceiling causing hard clipping Release When FinalPlug detects a peak it quickly turns down the gain so the peak will not clip After the peak has passed FinalPlug will restore the gain to the nominal level The Release time sets the duration of the gain restoration Long release times will tend to turn down the gain and keep it down Short release times will lead to a louder mix by quickly restoring gain after peaks Auto release When auto release is on the release time is automatically calculated based on the input signal and the Release parameter is ignored When auto release is off the release time is based on the Release parameter Truncate Dither Enable disables the truncate dither section of the plug in
17. The time scale of the display can be changed by clicking on the x1 and x10 controls at the upper right this changes the maximum display time from 250 msec to 2 5 seconds When the envelope is enabled the envelope response is shown in the time display as a blue shaded region This allows the user to visualize the attack hold and release times Frequency response display FREQUENCY RESPONSE 200 20k 2k i pr The reverb frequency decay plot is a three dimensional plot that shows the frequency response of MasterVerb during the reverb decay The plot allows easy visualization of both the overall equalization due to the early damping control and the time dependent equalization due to the late damping control The vertical axis represents amplitude in dB the axis that runs from left to right represents time in seconds and the rear axis represents frequency in Hz Note that low frequencies are shown on the right in green high frequencies are shown on the left in blue 60 5 MasterVerb From a frequency standpoint this plot is a mirror image of the damping controls which have the frequency plotted left to right as is typical The frequency response is principally affected by the Reverb Decay Time Early Damping and Late Damping parameters Changing the decay time causes the entire surface to pivot about time O as the decay slopes at all frequencies are changed Changing the early damping affects the initial equalization the freque
18. TrackPlug preset into the TrackPlug E7GC configuration only the first seven EQ bands the gate and Comp1 will be set up according to the preset 3 8 User Presets User presets are selected from a rolloff menu just below the Factory presets in the Preset menu When you first run a Wave Arts plug in there will not be any user presets and the menu will be empty When you save a preset using the Save option the preset is added to the User menu All instances of a plug in share the same set of user presets So after you save a preset with one instance of a plug in you can go to another instance and find that the preset can be found in its User preset menu too You can delete an individual user preset by holding down the SHIFT key while selecting the preset The entire set of user presets can be deleted using the Reset option described below User presets are stored in a text file called lt plugin gt Presets txt where lt plugin gt is the name of the plug in you are using If the file is deleted an empty preset file will be created automatically the next time the plug in runs User presets files are stored in the following directory depending on the operating system where lt username gt is your Windows login name Mac OS X Library Application Support Wave Arts lt plugin gt Windows 7 8 C Users lt username gt AppData Local Wave Arts 3 9 Save When you have created an effect you want to save as a preset
19. a single band and the EQ type must be lowpass highpass notch or bandpass Add and Delete The Add and Delete buttons are used to add and delete EQ bands Clicking Add will create a new EQ band and will display a new control handle You can also add a band by double clicking in the frequency response display Up to 10 bands may be created Clicking Delete deletes the current band You can also delete a band by ctrl clicking on its handle Enable The Enable button enables disables the entire EQ section When disabled the Enable button is not lit and the EQ section is bypassed Bypass The Bypass button bypasses the current band only Toggling the bypass useful to hear the effect of a single EQ band Any number of EQ bands can be bypassed As you switch between different bands the bypass button will change to reflect the bypass state of the current band Pre and Post The Pre and Post buttons control whether the EQ operates before the compressors Pre button lit or after the compressors Post button lit See the routing diagram at the start of this chapter for more information EQ Preset The EQ preset selector button allow you to select factory or user presets for just the EQ section of TrackPlug Clicking on the button will step to the next preset Clicking on the preset name will display a popup menu of the current factory and user EQ presets 37 Wave Arts Power Suite EQ Type The EQ Type selector button
20. across all bands simultaneously For example clicking on the threshold global control will move the threshold for all bands at the same time which is readily seen in the frequency response display Like other controls you can get fine resolution by depressing SHIFT or the right mouse button while dragging Global parameter section The global parameters are crossover slope lookahead compression mode and output gain They are described in detail in section 7 4 Dynamic response display The dynamic response display shows how an input level is mapped to a gain An the bottom of the display on the horizontal axis is the input level meter which moves left to right as the peak input level goes from 72 dB to O dB The bright green band shows the range from smallest to largest level processed by the dynamics section since the last meter redraw In the above example the levels range from 25 dB to 30 dB The width of this bright band indicates the dynamic content of the input signal a wide band indicates a rapidly changing input a narrow band or no band indicates a constant unchanging signal 83 Wave Arts Power Suite The thick green line on the graph shows how the input levels map to gains which range from 18 to 18 dB on the vertical axis In the above figure the range of 25 to 30 dB of input levels will map to gains between 3 dB and O dB Note that this mapping happens before the attack and release time filtering is applied to
21. and 800 Hz can improve clarity of lower pitched instruments Too much boost in this whole range can lend to a telephone like quality boost from 500Hz to 1Khz can sound horn like from 1Kz to 2Kz can sound tinny and often frequencies in this area can be reduced on mid range instruments like guitar vocals and keyboards to improve a mix High Midrange 2000 Hz 4000 Hz this range is an important speech recognition area and also determines projection and clarity of mid range instruments Too much boost in this area can be fatiguing on the ear Presence 4000 Hz 6000 Hz controls how close and distinct instruments and vocals sound too much in this range will cause harshness Brilliance 6000 Hz 20 kHz this range is associated with clarity sizzle and air TrackPlug s equalizer provides up to 10 separate EQ bands more than needed for most jobs Each band has a choice of 11 different filter types per band Again we ve erred on the side of providing more control rather than skimping All of the filter types are based on second order filters this limits their rolloff slope to 12 dB per octave The filter types are described below Parametric EQ The parametric EQ type is one you will use often as it allows you to reinforce or attenuate at a specific frequency point The parametric EQ is defined in terms of the 26 4 TrackPlug center frequency the height of the boost cut in dB and the width The width is measured
22. and the presets are written to the file Preset Export is also useful for making backup copies of your user presets If you have a large set of user presets be sure to export them to a backup file 3 12 Reset Reset is used to delete all of your user presets Selecting Reset will first ask you if you really want to do this and if you confirm all the user presets are deleted 3 13 Tools menu The Tools menu contains various important options described below Preset MS FinalPlug About ABLE Open User Manual 16 Unlock Plug In BIT Check For Updates Visit Website El 3 14 About The About option displays important information about your plug in An example is shown below 20 3 Menu Bar and Preset Manager About FinalPlug 5 WNYA wave arts FinalPlug 5 v5 82 YST 32 bit Build date Mar 17 2015 Registration CE8ED 66294 Power Suite 5 Machine ID 62534 User Manual Unlock Plug In Wave Arts Inc Copyright c 2015 99 Massachusetts Ave Wave Arts Inc Arlington MA 02474 8600 All rights reserved USA Tel 781 646 3794 YST plug in technology support wavearts com by Steinberg On the top line the plug in name and version are displayed along with the current plug in format AAX DirectX VST AU RTAS or MAS This is useful if you aren t sure which format of the plug you are running The build date of the plug in is display
23. as a background ambience separate from the foreground sound The loudness of the reverberation relative to the loudness of the foreground sound is an important distance cue The reason for this is due to the acoustics of reverberant spaces The foreground sound consists largely of the sound that propagates directly from the sound source to the listener this so called direct sound decreases in amplitude as the distance to the listener increases For every doubling of distance the amplitude of the direct sound decreases by a factor of one half or 6 dB The amplitude of the reverberation on the hand does not decrease considerably with increasing distance The ratio of the direct to reverberant amplitude is greater with nearby objects than it is with distant objects Thus distant objects sound more reverberant than close objects This relationship is diagrammed in figure 5 In the figure the direct sound amplitude drops 6 dB for each doubling of distance The reverberation amplitude shown in the figure starts at 12 dB relative to the direct gain and drops at 1 5 dB per doubling of distance Hence the reverberation gain is greater than the direct amplitude at distances greater than about 6 units Panorama provides gain and slope parameters for editing the relationship between amplitude and distance The gain parameter defines the starting gain in dB when the source is 1 distance unit feet or meters away from the listener The slope parameter defi
24. by comparing the peak level of a signal to the average level FinalPlug uses a short release time for input signals that are peaky and it uses a long release time for signals that are steady The implementation of the auto release mode is a balance between low distortion operation aggressive compression and reduced pumping effects The auto release mode in FinalPlug achieves a nice balance preventing distortion during steady tones even when compressing heavily while reducing pumping effects and using fast release times when possible Lookahead peak limiters like FinalPlug are often used to increase the volume of a mix In today s loudness wars this technique is often used to excess creating ultra compressed audio that can be quite fatiguing to listen to One need only look at the digital content of a typical pop CD to see how hyper compressed today s music is Readers are encouraged to read the following Current Trends in Mastering The Loudness War by Mark Donahue June 2003 Performer Magazine and What happens to my recording when it s played on the radio By Frank Foti Omnia Audio amp Robert Orban CRL Orban Truncation and Dithering FinalPlug also features bit depth truncation with dithering and noise shaping Bit depth truncation typically occurs as the final step in mastering when the sample resolution must be reduced to fit on the recording medium For example compact discs CDs are recorded with 16 bit samples
25. gain change as a signal crosses threshold is 36 dB To configure MultiDynamics as a compressor set the Lo gain higher than the Hi Gain and the gain will be reduced as the signal increases above threshold To configure MultiDynamics as an expander set the Lo gain below the Hi Gain and the gain will be decreased as the signal decreases below threshold More details on the operation of the Lo Gain Hi Gain Threshold and Ratio controls are given in the Parameter Description section 81 Wave Arts Power Suite 7 3 User Interface The user interface for MultiDynamics is shown at the start of this chapter The large window at the upper left is the frequency response display This shows the dynamic frequency response and the input levels in each band Below the frequency response display are controls to create delete and edit bands At the upper right is the dynamic response display Frequency response display The frequency response display shows most of the information regarding MultiDynamics operation At the top of the display the current equalization response is plotted with a thin green line The shaded green region shows the range of equalization which is determined by the hi and lo gains per band The lo and hi gains can be manipulated by dragging the green and blue triangles respectively up and down Holding down SHIFT or the RIGHT mouse button while dragging the hi or lo gains will move both at once At the bottom of the disp
26. parameter value plug in interface Most of these parameters have a one to one correspondence with controls on the user interface Each of TrackPlug s 10 EQ bands has the following parameters Parameter name Band Enable Type Frequency Height Width Values O Off 1 Bypass 2 On O Parameteric 1 Low shelf 2 High shelf 3 Vintage low shelf 4 Vintage high shelf 5 Lowpass 6 Highpass 7 Bandpass 8 Notch 9 Resonant low shelf 10 Resonant high shelf 20 20000 Hz 24db 24db 0 01 5 0 octaves or Q Each of TrackPlug s three dynamics processors has the following parameters 44 Parameter name Dyn Enable Dyn Knee Dyn Thresh Dyn Ratio Dyn Gain Dyn Attack Dyn Release Dyn Mode Dyn Lookahead Sidechain Mode Sidechain Monitor Sidechain EQ Type Values O Off 1 On O Soft 1 Medium 2 Hard 96 to 0 dB 1 to 50 24 to 24 dB 0 1 to 1000 msec 1 to 5000 msec O Clean peak 1 Clean RMS 2 Vintage peak 3 Vintage RMS 4 Vintage Warm AAX 2 Vintage peak 3 Vintage RMS 4 Vintage Warm O Off 1 1 msec 2 2 msec 3 5 msec O Off 1 Internal EQ 2 Internal EQ Compare 3 Internal EQ Invert 4 External EQ 5 External EQ Compare 6 External EQ Invert O Off 1 On 5 Lowpass 6 Highpass 7 Bandpass 8 Notch 4 TrackPlug Sidechain EQ Freq 20 20000 Hz Sidechain EQ Width 0 01 5 0 octaves The remaining TrackPlug
27. parameters are listed below Parameter name Values EQ Routing O Pre 1 Post EQ Enable O Off 1 On Brickwall Low Enable 0 Off 1 On Brickwall Low Freq 20 20000 Hz Brickwall High Enable 0 Off 1 On Brickwall High Freq 20 20000 Hz Output Gain 18 to 18 dB TrackPlug Enable O Off 1 On Limiter Enable O Off 1 On 4 5 Parameter Descriptions EQ Parameters The TrackPlug equalizer is described in detail in the section on Equalization earlier in this chapter Band Enable This is an internal parameter that is controlled by the Add Delete buttons and the Bypass button Each EQ band is either deleted that is inactive and not displayed in the interface enabled or bypassed Type Parametric Low shelf High shelf Vintage low shelf Vintage high shelf Lowpass Highpass Bandpass Notch Resonant low shelf Resonant high shelf Frequency For parametric notch and bandpass filters this is the center frequency for shelf lowpass and highpass filters this is the cutoff frequency Range is 20 20 000Hz Height Gain attenuation for the currently selected EQ band parameric and shelf filters only Range is 24dB to 24dB Width Bandwidth in octaves The higher the number the wider the filter and vice versa Applies only to parametric bandpass and notch filters Range is 0 01 to 5 0 octaves The vintage and resonant shelf filters use the width parameter to control resonance in units of
28. per octave beyond The lowpass and highpass filters have only one parameter corner frequency The corner frequency is defined at the 3 dB point in the response The choice to use the lowpass and highpass filters or to use the brickwall filters depends on how steep you need to rolloff to be 2 30 50 100 200 500 1k 2k 5k 10k 20k Lowpass filter with freq 2 kHz Real Time Analyzer RTA TrackPlug has a built in real time spectrum analyzer which allows you to see how the energy in the audio signal is distributed across frequencies The RTA is not available when running AAX DSP format The RTA displays the instantaneous energy in each of 31 frequency bands ranging from 20 Hz to 20 kHz The width of each band is 1 3 octave which corresponds roughly to the critical bands in the human auditory system Spectrum analysis is incredibly useful to see at a glance 30 4 TrackPlug which frequencies make up a sound The RTA can be inserted either before or after the EQ processing Placing the RTA before EQ processing is useful to see the unaltered spectrum of the input signal This can then guide subsequent application of EQ perhaps to boost weak frequencies in the input or attenuate strong frequencies Placing the RTA after the EQ can then verify that the EQ has applied the intended adjustments For stereo signals the RTA displays the peak value for left and right channels in each band The RTA will display a flat response with a pink noise inpu
29. section serves as a tab control to select the section for detailed editing below Click on the name of the section to select that section for editing The selected section is shown with a light background The controls below the overview sections are specific to the current selected section and will change based on the section that is currently selected In the figure above Compl is the current selected section Note that all the controls in the overview sections are functional regardless of which section is enabled So for example you can turn on and off Comp2 or change its threshold while Compl is selected for editing The reason to have the overview sections is to make it easy to select some presets set thresholds and get work done without having to bother with the detailed parameters of each section So for example if you were processing a dialog track you could select a noise gate preset for Gate a de esser preset for Compl and a vocal compression preset for Comp2 Then you enable the sections set your 40 4 TrackPlug thresholds and you re done If something requires further tweaking select the section tab and go below into the detailed parameters Enable The Enable button enables disables the corresponding dynamics section When disabled the Enable button is not lit and the dynamics section is bypassed Preset The preset selector button allow you to select factory or user presets for each dynamics section of TrackP
30. selects the EQ type of the current band There are eleven different types described in the Equalization section earlier in this chapter Frequency Height and Width knobs These knobs let you set the frequency height and width of the current band If the selected EQ type does not support all three parameters the unused parameters will display n a not applicable For the vintage and resonant shelf filters the width knob serves as a resonance control by displaying values in units of Q rather than octaves Note that the EQ Type Frequency Height Width and Bypass controls will change as you select different bands to reflect the state of the currently selected band Also note that the EQ preset affects neither the brickwall settings nor the sidechain EQ settings nor the EQ Enable button EQ display range The vertical range of the EQ display can be changed by the user allowing more subtle editing viewing of the EQ response Simply SHIFT click or right click for PC users on the vertical axis at the left of the EQ display to see a popup menu listing the ranges The ranges are 24 to 24 dB 12 to 12 dB and 6 to 6 dB TrackPlug will remember the last used setting When using the zoomed in ranges 12 and 6 dB it is possible that the EQ control handles will be off the display and will not be drawn In this case you can select one of these bands by using the SHIFT click popup menu and then edit the values using th
31. set Nine additional human head models from the UC Davis CIPIC Interface Lab are installed on the file system When Panorama launches it checks a special directory on the file system and will load any HRTFs found in the directory On Windows this directory is usually C Program Files Wave Arts Panorama PanoramaHRTFs on Macintosh OS X this directory is Library Application Support Wave Arts Panorama PanoramaHRTFs When clicking on the HRTF popup menu Panorama will display all built in and installed HRTFs that are available at the current sampling rate Built in HRTFs are available at 22050 32000 44100 48000 112 8 Panorama 88200 and 96000 Hz sampling rates The file system installed HRTFs are available at 44100 48000 and 96000 sampling rate If you are using a different sampling rate you will not see the file system installed HRTFs appear in the HRTF menu When Panorama is created it will internally use the built in sampling rate which is closest to the sampling rate of the host So for example if the host is using a sampling rate of 50 kHz Panorama will internally use 48 kHz and will choose filter coefficients and set delay lines assuming the sampling rate is 48 kHz In practice this means if you are using a non standard sampling rate Panorama will continue to function but it will not be optimal Doppler Enable Enables disables the Doppler pitch effect When Doppler is enabled the sound propagation delays from t
32. the gain modulation The dynamic response curve is determined by the parameters lo gain hi gain threshold ratio and knee The lo gain determines the dynamics gain when the input signal level is below threshold while the hi gain determines the dynamics gain when the input signal level is above threshold If the lo gain is greater than the hi gain then MultiDynamics acts like a compressor decreasing the gain as the input level increases beyond threshold Similarly if the hi gain is greater than the lo gain then MultiDynamics acts like an expander increasing the gain as the input level increases beyond threshold The lo gain is indicated by a green triangle pointing down the high gain by a blue triangle pointing up If the arrows are pointing towards each other MultiDynamics is compressing dynamic range if the arrows are pointing away from one another MultiDynamics is expanding dynamic range In the dynamics response display the green and blue triangles can be dragged up and down to adjust the lo and hi gains respectively The orange ball can be dragged up and down to move both gains simultaneously The horizontal coordinate of the orange ball is the threshold Dragging the orange ball left and right changes the threshold Holding down SHIFT or the RIGHT mouse button while dragging left or right will change the ratio Ratio determines the slope of the dynamics response however the relationship depends on whether MultiDynamics is compr
33. two compressors a gate a brickwall filter and a lookahead peak limiter in one efficient easy to use and great sounding pro audio plug in ideal for tracking mixing and post production Here are some of TrackPlug s key features e a 10 band EQ section with 11 different filter types e integrated realtime spectrum analyzer e low and high pass brickwall filters e two compressors and a gate each with optional side chain equalizer variable knees and optional lookahead delay e choose between clean or vintage peak or RMS dynamics modes e dual EQ comparison sidechain modes for precise de essing de ploding e external sidechain inputs AU MAS and RTAS only e lookahead peak limiter e EQ routing pre or post dynamics e comprehensive metering e separate presets for each section e comprehensive set of factory presets designed by industry professionals e supports up to 192 kHz sampling rate 23 Wave Arts Power Suite TrackPlug s audio routing and meter placement is shown in the diagram below Gate Comp1 Comp2 Output Meter Meter Meter Meter In Brickwall EQ Egi filter aal pre TrackPlug audio routing diagram Peak Out Limiter The input signal is processed first by the brickwall filter This allows the user to eliminate noise signals that are outside the frequency range of the recorded instrument Thumps rumble and hum can be eliminated using a highpass brickwall while high frequency hiss or tone
34. 80 degrees 40 to 90 degrees O 20 feet or meters 20 to 20 feet or meters 20 to 20 feet or meters 20 to 20 feet or meters O Polar 1 Cartesian O Off 1 On O Off 1 On O Off 1 On 18 to 12 dB 18 to 12 dB 18 to 12 dB 12 to O dB per distance doubling 12 to O dB per distance doubling 12 to O dB per distance doubling 1 128 1 128 O Off 1 On O Off 1 On O Off 1 On O Off 1 On O Off 1 On O Off 1 On 1 20 feet or meters 1 20 feet or meters 1 20 feet or meters 1 20 feet or meters 1 20 feet or meters 1 20 feet or meters O 17 see table O 17 see table O 17 see table O 17 see table O 17 see table 105 Wave Arts Power Suite Bottom Material Reverb Decay Time Reverb Room Size Reverb Early Damping Reverb Late Damping Reverb Delay Output Configuration Units Output Gain HRTF Doppler Enable Panorama Enable Stereo Width O 17 see table 0 5 60 sec O 100 20 20 000 Hz 20 20 000 Hz O 250 milliseconds O headphones 1 speakers 10 degrees 2 20 degrees 3 30 degrees 4 40 degrees O meters 1 feet 18 to 18 dB O Human 1 MIT KEMAR CIPIC HRTFs are 1000 subject number O Off 1 On 0 Off 1 On O Mono 10 90 degrees The reflection materials are listed in the table below The materials range from most reflective to least reflective The Per
35. AU AAX AAX DSP RTAS DirectX MAS Sampling Rates up to 192 kHz 1 O Formats mono mono stereo stereo 6 7 Presets Following is the list of presets that ship with FinalPlug Name Description Default Peak limiter with thresh at 0 1 dB ceiling at 0 1 dB CD Mastering Default for CD mastering peak limiter with thresh at 6 dB ceiling at 0 1 dB 16 bit truncation and 44 1 kHz Shape 2 DVD Mastering Default for DVD audio mastering peak limiter with thresh at 6 dB ceiling at 0 1 dB 24 bit truncation and 96 kHz shape Track Peak Limit Soft Soft track compression Track Peak Limit Medium Medium track compression actually pretty hard Track Peak Limit Hard Knock the stink out of it 77 7 MultiDynamics 7 MultiDynamics 0 Bypass Undo Copy MPB lt gt Preset Tools A UN 100 200 30 dB 30 0 dB 90 dB LOGAIN THRESH HIGAIN RATIO eae 7 1 Overview MultiDynamics is a powerful multi band dynamics processor useful for mastering track processing sound design and noise reduction MultiDynamics provides up to 6 bands with independent compression or expansion gating per band Up to 6 independent bands Full featured compressor or expander gate per band Clean and vintage compression modes Proprietary crossover filter network eliminates amplitude distortion between bands 6 dB oct 18 dB oct and 30 dB oct crossovers Per band bypass solo and mute controls
36. Bit Depth The resolution of the audio output from the plug in will be quantized to this value It should be set to the final desired bit depth of the target medium i e 16 bit for CD 24 bit for DVD You can select a value from the selector control or type a value between 4 and 24 into the text field 75 Wave Arts Power Suite Dither Shape Shape of the dither noise injected into the signal The frequency plot shows how the noise energy is distributed throughout the frequency range The idea of shaping is to concentrate the noise in areas that the human ear is less sensitive making it less perceptible The 44 1K shapes are designed for CD 16 bit 44 1K and the 96K shapes for DVD audio The 96K shapes concentrate the dither noise to frequencies above 20kHz Below 20kHz the dither noise is flat The table below lists the shapes gives descriptions and references other commercial products that use similar shapes Shape name Similar to Description Flat No noise shaping 44 1 kHz Shapel Waves L1 Normal Gentle reduction of noise below 10 kHz noise is concentrated between 15 kHz and 20 kHz 44 1 kHz Shapel Ultra Waves L1 Ultra Exaggerated version of Shapel 44 1 kHz Shape2 Waves L2 Normal Perceptually based noise shaping reduces noise below 5 kHz and concentrates above 15 kHz 44 1 kHz Shape2 Ultra Waves L2 Ultra Exaggerated version of Shape2 44 1 kHz Shape3 ExtraBit Medium Evenly reduces noise between 2 13
37. Comprehensive visualization of input levels and dynamic EQ response Adjustable knees and lookahead Mono or stereo 79 Wave Arts Power Suite Following is a diagram of MultiDynamics audio routing and meter placement Input Dynamics Output Meter Meter Meter Out Bandpass Peak Dynamics Filter Detector Processor up to 6 individual bands The input signal passes through a set of bandpass filters which separates the signal into individual frequency bands Each band is processed by a separate peak detector and dynamics processor The resulting signals are summed to form the final output As the above diagram shows a multi band dynamics processor implements a set of dynamics processors that respond to different frequency ranges However it is equally valid to think of a multi band dynamics processor as implementing a set of equalizers that respond to different dynamic ranges MultiDynamics 4 band and 3 band For AAX DSP MultiDynamics provides 4 band and 3 band reduced configurations to increase instance counts 7 2 About Multiband Dynamics Multiband dynamics processing combines the techniques of equalization and single band dynamics processing A multiband dynamics processor can do things that neither an EQ nor a single band dynamics processor can do However it is more complicated to use and harder to master A simple application of multiband dynamics is to apply compression or expansion to a specific frequency range
38. FT click or right click for PC users on the axis to the right of the attenuation meter to see a popup menu listing the ranges The ranges are 36 to 0 dB 24 to 0 dB 12 to 0 dB and 6 to O GB A different range can be selected for each dynamics meter TrackPlug will remember the last used setting for each meter Threshold control The input meter has a triangular control that lets you set the input threshold level Drag the control up and down to change the threshold 41 Wave Arts Power Suite Dynamics response display The detailed dynamics interface is highlighted by the dynamics response display on the right hand side The dynamics response graph shows how input levels map to output levels The default mapping obtained with a ratio of 1 1 is shown as a straight diagonal line A compressor with ratio greater than 1 will have a knee point above which the response bends to become more horizontal The x coordinate of the knee point is the threshold level Hence input values above the threshold will be attenuated Similarly a gate with a ratio greater than 1 will have a knee point below which the response bends to become more vertical In this case input levels below the threshold are attenuated Below the dynamics response is a horizontal input meter This shows the same information as the vertical input meter in the overview section The dynamics response contains several control handles that allow you to change the respons
39. License Agreement END USER LICENSE AGREEMENT FOR WAVE ARTS SOFTWARE IMPORTANT READ CAREFULLY This Wave Arts End User License Agreement EULA is a legal agreement between you either an individual or a single entity and Wave Arts Inc Wave Arts for the Wave Arts software accompanying this EULA which includes computer software and may include associated media printed materials and on line or electronic documentation SOFTWARE PRODUCT or SOFTWARE By exercising your rights to make and use copies of the SOFTWARE PRODUCT you agree to be bound by the terms of this EULA If you do not agree to the terms of this EULA you may not use the SOFTWARE PRODUCT SOFTWARE PRODUCT LICENSE The SOFTWARE PRODUCT is protected by copyright laws and international copyright treaties as well as other intellectual property laws and treaties The SOFTWARE PRODUCT is licensed not sold GRANT OF LICENSE This EULA grants you the following rights Installation and Use When using Wave Arts licensing you may install and use the SOFTWARE PRODUCT on up to three computers provided you are the only user A license must be acquired for each individual user of the SOFTWARE PRODUCT When using Pace iLok licensing you are granted a single user license which you can transfer to an iLok dongle or to a machine Reproduction and Distribution You may not reproduce or distribute the SOFTWARE PRODUCT except to make backup copies or to install as provided for
40. TVICW classi dente hee tie Ae te ee ahs See Se A St ee Sere a 79 7 2 About Multiband Dynamics i220 rro to rar aos o a 80 Te User Interface ts eG rave eel yard ect atone ii 82 TA Parameter S a AA a fleas ADA daa a 85 7 5 Parameter DescriptionS ci ic lies eee 86 7 0 Specifications a A a tetanic teas 88 LAS A And ba 88 8 Pa oraMa ii A A anodes 91 8 I OVERVIGW ci AAA A AAA id 91 8 2 About 3D Audio and Acoustic Environment Modeling oooocococcccco o 92 8 3 User Interface ti A da A a a 103 8 4 Parameters odios e rna TE E EER A CEE catia E R 105 8 5 Parameter Descriptions su ni i 107 8 0 SPECIAL a cee eines 113 8 7 Presets aras A ria adrede 113 License AGTrESMENE LLasaisitirla tii ill 115 SUPPO arasan AARAA tl trat iia 117 o A O 119 1 Introduction 1 Introduction 1 1 Power Suite Overview Power Suite is a set of plug ins for mixing and mastering consisting of the following individual plug ins TrackPlug channel strip with EQ and dynamics MasterVerb stereo reverb FinalPlug peak limiting and dither MultiDynamics multi band dynamics Panorama spatial audio processor 1 2 Installation Installers for Power Suite are found on the downloads page of the Wave Arts website There are separate installers for Mac and Windows You can download the full suite installer or individual plug in installers Mac installers are dmg files which after downloading will expand into a pkg installer file double clic
41. Time parameter which is defined at 500 Hz For example a scaling of x2 means the reverberation will be twice as long as the Reverb Time parameter while a scaling of x0 5 means the reverberation will be half as long The late damping EQ is a three band shelf equalizer with lo mid and high gains and adjustable band edge frequencies Late Damping Lo Freq Sets the transition frequency between the low band and mid band of the late damping EQ Late Damping Lo Scale Sets the reverb time scaling of the low band of the late damping EQ 63 Wave Arts Power Suite Late Damping Mid Gain Sets the reverb time scaling of the mid band of the late damping EQ Late Damping Hi Freq Sets the transition frequency between the mid band and high band of the late damping EQ Late Damping Hi Scale Sets the reverb time scaling of the high band of the late damping EQ Envelope Enable Enables or disables the envelope When the envelope is enabled the wet reverb sound is modulated by the envelope When the envelope is disabled it is bypassed Envelope Threshold Set the trigger threshold of the envelope from 36 to 0 dB Envelope Attack Time Sets the attack time of the envelope from 1 to 2000 msec Envelope Hold Time Sets the hold time of the envelope from 1 to 2000 msec Envelope Release Time Sets the release time of the envelope from 1 to 2000 msec Dry Delay Sets the delay of the dry signal ranging from O to 500 millis
42. Wave Arts Power Suite 5 MAN User Manual Last updated May 12 2015 Copyright 2015 Wave Arts Inc All Rights Reserved Table of Contents Table of Contents Ls IEP OGUICE OD naa ita ida is ile canada 5 1 1 Power Suite OVErviSW costilla rd bane tense 5 OZ MASA es sd aa 5 3 ASIS EOI a bed cae CDE 5 1 4 Wave Arts licensing using key codes 0oooocconccncconccnnncnncrnncnnnrnnnnnnrnos 6 LS ME CAINE Dis a A ae 6 1 6 PACES ILOK NEONSING ia calls ev beds 7 1 7 Power Suite AAX A a deadeeeidevdededara Peenban 8 1 8 Modifications for AAX DSP az feces oa ceaddciader atari es opesianeay dvaansid vou heas 8 1 9 Reduced configurations for AAX DSP ccee eens e eee eee eee teeta ene 9 1 10 Plug in Instance Counts for AAX DSP cccecceeee eee eee eee eee ees 10 2 Plug in Control Operation pescar ee ie ee toe ae chee aes 13 DAL KNOBS wastes O as 13 22 VOX EMUCY oc eaa ion save santa gente ccakdunsatoraetaeness diakioean EE EA A A E eaten 13 2S Selector DUO sr isa las 14 2 A SIAC S ova wee seed a a a eirekcaasbaea a LAEE 15 2 DvBUClLO MNS A A A ect A Lotro 15 2 6 Qutp t MeteES ta A A tins 15 3 Menu Bar and Preset Manager cocccccconccncconccnnnnnccnnnnnnnnnnnnnnrnnrnrnrnnnnns 17 A O 17 3 2 RO wade aaa delat Gidget alee etaaed Gittins REE 17 JI COPY berth id co cae 17 3 4 A B A he ake ewe tata Ol eae he Goren Rat avila ith geo 17 3 5 Preset name and arrow Controls ccc ceeee cece eee eee eee tee eeeeeeee
43. above DESCRIPTION OF OTHER RIGHTS AND LIMITATIONS Limitations on Reverse Engineering Decompilation and Disassembly You may not reverse engineer decompile or disassemble the SOFTWARE PRODUCT except and only to the extent that such activity is expressly permitted by applicable law notwithstanding this limtation Software Transfer 115 Wave Arts Power Suite You may permanently transfer all of your rights under this EULA provided you retain no copies you transfer all of the SOFTWARE PRODUCT and the recipient agrees to the terms of this EULA Termination Without prejudice to any other rights Wave Arts may terminate this EULA if you fail to comply with the terms and conditions of this EULA In such event you must destroy all copies of the SOFTWARE PRODUCT and all of its component parts COPYRIGHT All title and copyrights in and to the SOFTWARE PRODUCT including but not limited to any images text and applets incorporated into the SOFTWARE PRODUCT the accompanying printed materials and any copies of the SOFTWARE PRODUCT are owned by Wave Arts or its suppliers The SOFTWARE PRODUCT is protected by copyright laws and international treaty provisions Therefore you must treat the SOFTWARE PRODUCT like any other copyrighted material LIMITED WARRANTY NO WARRANTIES Wave Arts expressly disclaims any warranty for the SOFTWARE PRODUCT The SOFTWARE PRODUCT and any related documentation is provided as is without warra
44. back and vice versa An elevation error refers to a misperceived elevation angle for example an overhead source may be perceived as being in from of or behind the listener Both front back and elevation performance is much better when using headphones than when using loudspeakers because the high frequency cues are more faithfully reproduced 94 8 Panorama 8 2 5 Panorama HRTFs The HRTFs used by Panorama were measured from human subjects and a dummy head microphone The human measurements were made at the CIPIC Interface Laboratory at the University of California at Davis Panorama s built in Human set is derived from CIPIC Subject 003 The dummy head measured was a Knowles Electronic Manikin for Acoustic Research KEMAR The KEMAR was measured at the MIT Media Laboratory A variety of measurements are supplied with Panorama to allow users to audition different head models It s possible that users of Panorama will find a particular HRTF set works better for them than the others Which HRTF set should be used in production There s no easy answer to this question We have found the Human set works well for headphone reproduction and the KEMAR set works well for loudspeaker reproduction 8 2 6 Reproducing 3D Audio Using Loudspeakers When reproducing localization cues to a listener it is important that the left and right audio channels remain separated that is the left ear signal should go to the listener s left ea
45. calization performance As with binaural synthesis crosstalk cancellation performance is ultimately limited by the variation in the size and shape of human heads Because of the limitations of crosstalk cancellers 3D audio over loudspeakers is prone to both front back confusions and elevation errors In particular back to front confusions tend to be very common which means that many listeners may not be able to perceive sounds as being in the rear In practice this means that when panning a sound from the front around to the side and to the rear the result will be perceived as a sound panning to the side and then back to the front Also elevation performance is generally not good over loudspeakers For example when a sound is moved from directly to the right to directly overhead this may be perceived as though the sound is moving from the right to directly in front 96 8 Panorama 8 2 8 Acoustic Environment Modeling Acoustic environment modeling refers to combining 3D spatial location cues with distance motion and ambience cues to create a complete simulation of an acoustic scene By simulating the acoustical interactions that occur in the natural world we can achieve stunningly realistic recreations above and beyond that possible with just 3D positional control Panorama combines 3 D positioning with accurate simulations of room surface reflections late diffuse reverberation distance cues and the Doppler effect These phenome
46. ceiling defines the maximum level the limiter will output The threshold defines the input level at which limiting takes place The easiest way to use the limiter is to first set the ceiling control to the desired peak output level then lower the threshold to achieve the desired amount of compression Adjust the ceiling and threshold by clicking on the triangular indicators next to the meters and dragging them up and down If you continue to lower the threshold you will get increasingly more signal compression At any time you can compare the limited signal with the unlimited signal by toggling the limiter enable control or the FinalPlug bypass control Note that FinalPlug applies a constant gain equal to the different between the ceiling and threshold controls so that even if limiting is not taking place FinalPlug is applying a constant gain to the signal Three meters show the stereo input levels stereo output levels and amount of limiter attenuation At the top of the meters are peak indicators that show the maximum value achieved Click on the indicators to reset them When the input exceeds the threshold and limiting takes place the attenuation meter simply shows the difference between the input and the threshold Like other Wave Arts compression meters the attenuation meter has a two color scheme to show the minimum and maximum attenuations the dark red color extends from 0 to the minimum attenuation while the bright red extends from the m
47. crosstalk cancellation is degraded when the listener is off center or not facing forward Loudspeaker 3D audio systems are extremely effective in desktop computing environments This is because there is usually only a single listener the computer user who is almost always centered between the speakers and facing forward towards the monitor Thus the primary user gets the full 3D effect because the crosstalk is properly cancelled In typical 3D audio applications like video gaming friends may gather around to watch In this case the best 3D audio effects are heard by others when they are also centered with respect to the loudspeakers Off center listeners may not get the full effect but they still hear a high quality stereo program with some spatial enhancements Many crosstalk cancellers are based on a highly simplified model of crosstalk for example modeling crosstalk as a simple delay and attenuation process or a delay and a lowpass filter Other crosstalk cancellers have been based on a spherical head model The crosstalk canceller used by Panorama is based on actual HRTF measurements and thus accurately models the crosstalk that occurs with human listeners For typical human listeners the Panorama crosstalk canceller improves channel separation by about 20 dB in the 100 Hz to 6 kHz range This may seem like a modest improvement but in fact it is quite good Even a small improvement in channel separation leads to a large improvement in lo
48. delay amplitude and tonal transformation for the particular sound source location to the left or right ear of the subject The measurement procedure is repeated for many locations of the sound source relative to the head resulting in a database of hundreds of HRTFs that describe the sound transformation characteristics of a particular head The term Head Related Impulse Response HRIR is often used to identify the time domain impulse response corresponding to an HRTF which can refer specifically to the frequency domain representation of the measurement In this tutorial we use the term HRTF to describe either the time domain or frequency domain representation of the measurement In fact Panorama uses time domain representations 8 2 3 Binaural Synthesis A 3D audio system works by mimicking the process of natural hearing essentially reproducing the sound localization cues at the ears of the listener This is most easily done by using a pair of measured head related transfer functions HRTFs as a specification for a pair of digital audio filters equalizers When a sound signal is processed by the digital filters and listened to over headphones the sound localization cues for each ear are reproduced and the listener should perceive the sound at the location specified by the HRTFs This process is called binaural synthesis binaural signals are defined as the signals at the ears of a listener 93 Wave Arts Power Suite listener weari
49. e 25 30 Hold CIC as ade 57 HR TE det eee dooce O A 92 95 112 l oka e AEAT 7 Import prese Earr ra aaa AEA 20 lnstallatior eeka ana anana Aa 5 K Key COJE coin iiae Aa N aE 6 Na 46 85 87 NOS aio 13 L Late early MIX oooooccccccncoccnccnconccnncnnns 64 HIGEASING A aa oa 6 LO gan 84 86 Lookahead 0 0occccoccccccccnno 46 70 83 85 Lowpass filter 25 30 M Machine Dicc 6 Makeup Dai ooococcccccoccnncncnnccnccnnnnncns 31 Master Verb oooocccccccccnncnnnnnnnnnnn cnn 53 MaterlalS coc ia 106 Menu Dai rc 17 MeterS covers ini 15 41 A A R i a 33 34 Multiband dynamics 80 MultiDynamics ooococccccconnncnnccnncnncnnnos 79 N Noise ShapinQ ococcccccccconcncnnccnccnnnnnons 71 NOtcCh Filter bocina a ae 28 O Output configuration s es 111 Overhead VICW cccccceee ee eeeeeee ee eeues 104 P PAGE nse lea oe ig Se nas 7 Parametric EQ unas 26 Peake limite a eee 35 43 70 Peak mode tii ev cuaniee ves tates 32 Plater a ia 63 Wave Arts Power Suite Plate reverb acido 56 Pre POS Eater casinos 24 Pre delay vc c Avereeccaercdasaesieetevn heeds 64 Preset menpian raa ra na 18 Preset NAME cccccceccceceeeeeeeeeeeeuneaaes 18 Presets Atusi na AAA 18 R Ea LELO ind E E 31 84 86 Real Time Analyzer RTA 30 38 Reflection type oocococcccoccnccnnconcnnnnnannos 62 Reflections c ccceeaes 55 56 98 103 Registration cccceeeee cece eect eae ea eee 5 Release time
50. e EQ parameter knobs Brickwall filter The brickwall filter has an enable button and a knob for both the lowpass and highpass brickwall filter When one of the bands is enabled a corresponding brickwall control handle appears in the frequency response display You can edit the brickwall frequency by dragging the control handle or dragging the knob Real Time Analyzer RTA To enable the RTA click on the RTA button below the EQ display note that the RTA function is not enabled when running AAX DSP format The RTA display appears overlaid on the EQ display Further customization of the RTA modes is accomplished by SHIFT clicking or right clicking for PC users on the vertical dB axis at the left of the EQ display This creates a popup menu showing the various RTA options below the options for selecting the EQ axis range The RTA options include RTA range RTA pre post and show hide RTA axis 38 4 TrackPlug RTA range Select the dB range you wish to view The choices are 144 to O dB 96 to O dB and 48 to O dB Note that when the 48 dB range is selected and the EQ is set to 24 to 24 dB there is a one to one correspondence between the RTA levels and the EQ display Hence if you move the EQ up by 6 dB you should see the same 6 dB rise in RTA levels assuming the RTA is running post EQ RTA pre post Select RTA Pre EQ to run the RTA before the EQ hence the RTA levels you see are before the EQ processing is applied Select
51. e directly The controls differ for the gate and the two compressors The table below describes how the handles can be used to change the parameters Compressor Orange Handles Control Mac or Windows Adjust threshold and gain Click knee handle drag Adjust threshold gain and ratio so right Shift click Mac or right click Win knee handle stays at same point handle drag Adjust ratio Click right handle drag up down Click orange triangle above input meter Adjust threshold drag left right The compressor knee dragging is cleverly designed to let you adjust threshold ratio and gain with a single mouse operation Normally dragging left right changes the threshold and dragging up down changes the gain However if you hold down the Shift key or the right mouse button on Windows the graph will pivot on the right handle so dragging up down will change the ratio as well Gate Blue Handles Control Mac or Windows Adjust threshold and gain Click knee handle drag Click blue triangle above input meter drag left right Adjust ratio Click bottom handle drag left right Adjust threshold Detailed dynamics controls Knobs are provided for the Threshold Ratio Attack Release and Gain controls In addition there are selector controls to set the dynamics mode knee curvature and 42 4 TrackPlug lookahead delay These parameters are described elsewhere in this chapter The sidechain is setup using two controls a selec
52. e in a small room the soundwaves collide with walls much more frequently and thus are absorbed more quickly than in a large room Reverberation is a really important acoustic phenomenon There is at most one direct path from the source to the listener whereas there may be millions of indirect paths particularly in a room where a sound can bounce around hundreds of times before being absorbed Thus in typical listening situations most of the energy we hear from a sound source is actually reverberation The perception of reverberation depends on the type of reverberation and the type of sound In small room with fast decaying reverberation the reverberation imparts a tonal quality to the sound that is readily identified as a small room signature In a larger room the reverberation can create a background ambience that is easily distinguished from the foreground sound and this is readily identified as a characteristic of large spaces In this manner reverberation imparts useful spatial information about the size of the surrounding space Early reflections arriving within the first 80 msec after the direct sound tend to become perceptually fused with the direct sound Hence the early reflections modify the character of the sound itself whereas much later reverberation becomes associated with the background ambience A prominent reflection that occurs soon after the direct sound will create a comb filtering effect as frequencies are either ampl
53. e licenses are grouped together 1 7 Power Suite AAX The AAX format Power Suite plug ins have some differences with their non AAX counterparts The next few sections will summarize the differences Also AAX specific differences are indicated in the detailed descriptions for each plug in later in this manual Previous Power Suite plug ins have been available in various native formats including Avid s RTAS and AS native formats but also in AU VST DirectX and MAS formats Avid s TDM format for operation on DSP cards was not supported The AAX Avid Audio eXtensions format replaces RTAS AS and TDM and provides a single format for both native and DSP operation Three of the Power Suite plug ins support AAX DSP format for operation on DSP accelerated HDX hardware TrackPlug FinalPlug and MultiDynamics MasterVerb and Panorama have not been ported to AAX DSP and remain native only 1 8 Modifications for AAX DSP In order to port to AAX DSP some modifications were made to the underlying algorithms e TrackPlug EQs now use new 32 bit filter architecture which performs nearly as well at the 64 bit implementation previously used This is done for efficient operation on 32 bit TI DSPs used in HDX hardware e The above mentioned 32 bit filter architecture is now used for TrackPlug brickwall filters and MultiDynamics bandpass filters The TrackPlug Real Time Analyzer is operational only in AAX native e The Clean dynamics process
54. e parameters after the edit Both A and B buffers described below have their own undo buffers 3 3 Copy Clicking the Copy button copies the current set of effect parameters to the unused A B buffer Hence if the A buffer is currently selected the parameters are copied to B and if the B buffer is selected the parameters are copied to A After clicking Copy you can continue to make changes and then revert to the original copied settings by clicking either the A or B buttons to switch buffers 3 4 A B buffers The A B edit buffers allow you to compare two different sets of parameters or presets One of the A or B buttons is always lit the button that is lit shows the current buffer Clicking either the A or B button will switch to using the other buffer thus changing the effect settings assuming different settings are stored in A and B Here s how to use the A B buffers to compare two different presets Select a preset from the Preset menu then switch to the other buffer and select a different preset Now switch between the two buffers to alternate between the two different presets 17 Wave Arts Power Suite 3 5 Preset name and arrow controls The currently selected preset name is displayed in the text field in the menu bar Changing any parameters causes an asterisk to be displayed at the end of the name This indicates that changes have been made to the preset In order to save the changes to a user preset you must selec
55. eaes 18 3 6 Preset TCI o o eta tes tos A AA et beatae 18 Did Fatry Presets icons iranien at aen rail ia bi AASE llas 18 3 0 USEF Presets sar lA Ao oli 19 BD SAVE oa II ca co tere 19 DELO WIM DOM Esti iii dias adds 19 EROS 20 A A E A 20 SL TOS AMEN a ii ceci 20 IA ADO eiior erai ada aa E AARE a A A AA a a 20 3 15 Op n User Manli iii 21 3 16 Unlock Pl Sisa 21 3 17 Check for Updates meiinoar aiai eian dia 21 3 18 Visit Websites raae A A EAE RAEE EE T 22 Ae TraekPl g a e E id 28 23 ALI as 23 4 2 About TRACK PIUGi sc 422252tecs ds Cues li 25 4 3 User Interface sii a see a Tata ia 36 4 4 Parameters cuore diia 44 4 5 Parameter DeSCriptiOnS eeceeee eee eect eee tence eee ee eee ne eeeeeaee ees 45 AOS DOCINGANONS es A O a ae a a EEN 48 AT A n a a a aa a E a E 48 Wave Arts PowerSuite MAS ED sia 53 LV OI SA a 53 5 2 About Master V SPD a A AR geese 54 E ll AA itr NAAA EEE NA PEE EN ETE EARRA Aa aes 58 DA Parameters oso di a dada TAA 61 5 5 Parameter DescriptlOlS espiar Siri 62 e AAA untetsdeehuaweeteas cues PATATA tes DA iaai 65 DPTO io A shQieeett AA aa 65 Gs PIN GMP AP EE A E EEN 69 LOVE oil 69 6 2 ADOUEFiNnalPI Gia adi E A e AAA RE 70 6 3 User Interface oia dd dd 72 6 4 Parameters viii daa 74 6 5 Parameter DescriptiONs 1 comisionados ie cea los dl oda 75 6 6 SPECINICALIONS iis eA asec eer das drt tare AA 77 6 7 Presets arnis Au sentient elt a ett oe 77 Te M ltiDYyNa MICS A ete ea eae hes et ads ad 79 Tet OVE
56. econds By delaying the dry signal the wet reverberation can precede the dry signal One use of this parameter is to create a simulated reverse reverb effect where the reverb builds up and then ends with the dry sound Wet Delay Sets the delay of the wet output ranging from 0 to 500 milliseconds Wet Delay is often called Pre Delay Wet Delay is used to achieve an additional echo effect in the reverberation by delaying the reverberation relative to the dry sound Wet Delays larger than 100 msec can be used to simulate large spaces such as stadiums and canyons The Wet Delay will be inaudible unless the Wet Dry Mix is set to something greater than 0 so that both the wet and dry outputs are audible Wet Dry Mix Sets the amount of wet processed sound and dry unprocessed sound that is mixed together to create the output 100 is all wet 0 is all dry and 50 is an equal mix of wet and dry sound Effectively the Wet Dry parameter sets the perceived distance of the input sound 100 sounds very distant and 0 sounds very near Late Early Mix Sets the amount of late and early reverberation that is mixed together to create the wet reverb signal The late reverberation comes from the late reverb algorithm The early reverberation comes from the early reflection generator The Late Early Mix is used to set the definition of the early portion of the reverb response by mixing in early reflection with the late reverb Note that if t
57. ed on the next line If the plug in is using Wave Arts licensing the registration status is displayed on the next line If the plug in is operating in demo mode the time remaining if any is displayed If the plug in has been successfully registered unlocked the key code and bundle name is displayed The Machine ID of the computer is displayed on the next line Finally buttons are provided for opening the registration dialog and the user manual If the plug in is using Pace iLok licensing there is no registration status or Machine ID displayed and the Unlock button will be disabled 3 15 Open User Manual Select this option to open the user manual in a browser If the manual isn t found you will be asked to navigate to it Once the manual is opened successfully the plug in remembers the location 3 16 Unlock Plug in This option is described in the Installation and Registration chapter of this manual 3 17 Check for Updates 21 Wave Arts Power Suite If you are connected to the internet selecting this option will launch a browser and will navigate to the Wave Arts Downloads page 3 18 Visit Website If you are connected to the internet selecting this option will launch a browser and will navigate to the Wave Arts home page 22 4 TrackPlug 4 TrackPlug FEypass Undo Copy PE lt Piano Pop gt Preset Tools AN IN OUTPUT 4 1 Overview TrackPlug combines a 10 band EQ a realtime analyzer
58. effective length of the reverberation tail in seconds The decay time parameter is independent of the room size Reverb Room Size Scales the size of the simulated room ranging from 0 to 100 A room size of 0 sounds like a very small space such as a bathroom or closet 25 sounds like a small studio 50 sounds like a jazz club 75 sounds like a concert hall and 100 sounds like a very large room such as an aircraft hangar or stadium The room size parameter is independent of the decay time In general small rooms sound best with short decay times and large rooms sound best with longer decay times Reverb Pre Delay Delays the reverb output Depending on the sound source small delays less than 50 msec will generally increase the audibility of the reverberation Larger delays greater than 80 msec will usually be perceived as a 109 Wave Arts Power Suite separate echo Hence the pre delay parameter can be used to create an echo effect for simulating really big spaces Reverb Early Damping Adjusts the cutoff frequency of a lowpass filter at the input of the reverb thus the Early Damping parameter sets the upper limit of frequencies admitted into the reverb The range of the Early Damping parameter is from 20 Hz to 20 kHz Setting this parameter to the maximum value opens up the filter completely for maximum brightness A low Early Damping setting results in a muffled reverberation whereas a high Early Damping setting r
59. eption of reverberation depends on the type of reverberation and the type of sound In small room with fast decaying reverberation the reverberation imparts a tonal quality to the sound that is readily identified as a small room signature In a larger room the reverberation can create a background ambience that is easily distinguished from the foreground sound and this is readily identified as a Characteristic of large spaces In this manner reverberation imparts useful spatial information about the size of the surrounding space Reverberation that contains a lot of high frequency energy in the decay is associated with rooms that have hard reflective walls which do not readily absorb high frequencies Similarly reverberation that is dull sounding is associated with rooms that contain soft materials such as plush carpets and drapes which readily absorb high frequencies In this manner reverberation imparts useful information about the composition of the surrounding space 97 Wave Arts Power Suite Reverberation is also important for establishing distance cues In a reverberant space when the distance between the source and the listener is increased the level of the direct sound decreases considerably but the level of reverberation does not decrease much Thus the level of direct to reverberant sound can be used as a distance cue with dry non reverberant sounds perceived as being close and reverberant sounds perceived as being distan
60. ers or headphones Crosstalk canceling based on real head models Ships with 10 human and 1 dummy head HRTF set The following sections provide a tutorial on 3 D audio and acoustic environment modeling including descriptions of Panorama s implementation 91 Wave Arts Power Suite 8 2 About 3D Audio and Acoustic Environment Modeling 8 2 1 What is 3D Audio A 3D audio system has the ability to position sounds all around a listener The sounds are actually created by the loudspeakers or headphones but the listener s perception is that the sounds come from arbitrary points in space This is similar to stereo panning in conventional stereo systems sounds can be panned to locations between the two loudspeakers creating virtual or phantom images of the sound where there is no loudspeaker However conventional stereo systems generally cannot position sounds to the sides or rear of the listener nor above or below the listener A 3D audio system attempts to do just that To understand how 3D audio systems work it is useful to start by considering how humans can localize sounds using only two ears A sound generated in space creates a sound wave that propagates to the ears of the listener When the sound is to the left of the listener the sound reaches the left ear before the right ear and thus the right ear signal is delayed with respect to the left ear signal In addition the right ear signal will be attenuated because of shadowing
61. essing or expanding The MultiDynamics ratio parameter ranges from 1 to 50 When compressing the ratio is interpreted as ratio 1 when expanding the ratio is interpreted as 1 ratio The figure below shows how various expansion and compression ratios would appear in the MultiDynamics dynamics response display Multidynamics ratio goes to 50 which functions as an infinite ratio thresh thresh t t 1 inf inf 1 expansion compression The knee parameter sets the curvature of the dynamics response when passing through the threshold which is usually called the knee point A soft knee has a rounded shape a hard knee has a sharp edge Each dynamics response in 84 7 MultiDynamics MultiDynamics has two knees the first knee occurs at the threshold and the second knee occurs when the gain is limited by the hi or lo gain setting With small ratios near 1 the second knee may not be visible on the dynamics response graph because it is out of the range of 72 to O dB 7 4 Parameters This section describes all the internal parameters of MultiDynamics as would be displayed by a generic parameter value plug in interface Most of these have a one to one correspondence with controls on the user interface Band parameters Parameter name Values Band Enable O Off 1 Bypass 2 On 3 Solo Band Lo 20 20000 Hz Band Hi 20 20000 Hz Band DynEnable O Off 1 On Band Thresh 72 to 0 dB Band LoGain 18 to 18 dB
62. esults in a bright crisp reverberation Reverb Late Damping Sets the cutoff frequency of the damping filter in the reverb A lower setting causes high frequencies to decay faster resulting in a warmer sounding late reverberation a higher setting results in a brighter sounding late reverberation The range of the Late Damping is from 20 Hz to 20 kHz Setting this parameter to the maximum value opens up the filter completely for maximum brightness A value of around 4 kHz results in a natural sounding reverberation for typical rooms Rooms with hard reflective walls consisting of concrete or tile such as a bathroom would be simulated using a higher Late Damping Rooms containing much absorbent material would be simulated using a lower Late Damping Spatialization Parameters DIRECT REFLECTION REVERB SPATIAL MIX GAIN SLOPE LENGTH Direct Enable Enables disables the direct source e the propagation from the source directly to the listener When disabled the direct source is muted allowing the user to audition just the early reflections or reverberation Reflection Enable Enables disables the set of six early room reflections Reverb Enable Enables disables the late diffuse reverberation Direct Gain Sets the gain of the direct source Reflection Gain Sets the gain of the reflected sources Reverb Gain Sets the gain of the reverberation 110 8 Panorama Direct Slope Sets the attenuation of the direc
63. fect reflector material is artificially created to have a perfectly flat frequency response Material Value 0 COND UB WIN 106 Material Perfect reflector Poured concrete Plaster on brick Concrete block painted Wooden floor Damped plaster on brick Plaster on lath Glass window Wooden platform Concrete block unpainted Leather seating Heavy carpet on concrete Curtain Heavy carpet on padding Acoustic tile Thick padding Thick padding extended Plush seating Audience 8 Panorama 8 5 Parameter Descriptions Direct Parameters DIRECT O PRESET AZIMUTH E ELEVATION E DISTANCE COORDINATE SYSTEM Mono Azimuth The azimuth angle of the source in degrees Elevation The elevation angle of the source in degrees Distance The distance of the source in distance units feet or meters depending on the setting of the distance units parameter X The X coordinate of the source Y The Y coordinate of the source Z The Z coordinate of the source Coordinate System Sets polar or Cartesian coordinates When polar coordinates is selected the position of the source is set by the azimuth elevation and distance parameters When Cartesian coordinates is selected the position of the source is set by the X Y and Z parameters When the coordinate system is changed by clicking on the Polar Cartesian button the Panorama user interface will automatically convert the current source position to the
64. g the attack and release times correspond to the attack and release times of the instrument you are processing Peak and RMS modes Dynamics processors often have peak and RMS modes In peak mode the processor is tracking the peak levels in the signal that is looking at the peak absolute values of the signal In RMS mode the processor is tracking the RMS levels of the signal RMS stands for root mean square it essentially means you square the signal take the average value and take the square root of this value This is a measure of the average power level of the signal which correlates roughly with the perceived loudness of the signal In practice computing a running estimate of the RMS level requires doing a short term average of the recent input hence the RMS levels change much slower than the peak levels Furthermore peak levels will tend to be much higher than RMS levels The relationship between peak and RMS levels depends on the signal A square wave or a constant DC signal will have identical peak and RMS values A sinusoid pure tone has a peak that is 3 dB higher than its RMS value For many music and speech signals the peak values may be 10 dB higher than the RMS levels One would typically use peak mode compression when processing an instrument sound or aggressively compressing a mix whereas RMS mode would typically be used to even out the loudness of mix 32 4 TrackPlug TrackPlug dynamics modes TrackPlug ha
65. ggered by input sound The envelope makes it easy to set up gated reverb effects where the wet reverb is turned on when there is an input and then abruptly shuts off after a certain amount of time 57 Wave Arts Power Suite Advanced users may want to know that the attack and release envelope segments are generated using exponential shapes Hence the attack and release segment times are actually exponential time constants For attack segments the segment will have grown to 63 of full scale 4 dB after the attack time It will continue to grow to O dB during the hold segment Release segments will decay 8 7 dB every release time The exponential shapes sound very natural Use your ears to adjust the time constants to suit Mixing with MasterVerb For music production it is common to place a reverb plug in on an aux bus in your host application program and use the aux send on the tracks you want to be processed with reverb In this case the reverb wet dry mix should be set to 100 therefore the aux return gain will set the wet level and the track gain will set the dry level Using reverb in this way not only saves CPU resources but lends to a more natural sound since each track is in a sense sharing the same space When using reverb as an insert effect you will want to use much drier mix levels to prevent the sound from becoming too muddy with reverb The presets are designed assuming the reverb will be used as an insert effec
66. ght changes both the early reflection size and the late reverb size Moving the control up down changes the reverberation decay time Small sounding spaces are selected at the lower left large spaces at the upper right Damping controls EARLY DAMFING Zk 20k LATE DAMPING Each of the damping controls shows the frequency response of the three band shelving filter that implements the damping For the early damping control the vertical axis is labeled in dB while the late damping control vertical axis is labeled in terms of reverb time scale factor The colored control handles allow the adjustment of the band levels and frequency crossover points The midrange band controls light blue are fixed at 640 Hz and can only be moved up and down to adjust the level The user should note that the early damping control affects all the reverberation both early reflections and late reverb It is called early to distinguish it from the late damping that only affects the response of the late reverb decay 59 Wave Arts Power Suite Time response display E xl Peal The time response displays shows a caricature of the actual time response of the reverb The direct response is shown as a vertical white bar in the above diagram it is on the far left at time 0 The early reflection response is shown as a series of green vertical lines that represent the individual reflections The late reverb response is shown as a green shaded region
67. hain EQ applicable only when bandpass and notch types are selected Global Parameters EQ Routing Sets the position of the EQ module in TrackPlug s signal processing chain See the routing diagram at the start of this chapter for details When routing is set to Pre the EQ is processed before the compressors When set to Post the EQ is processed after the compressors EQ Enable Enables or bypasses the EQ section 46 4 TrackPlug Brickwall Low Enable Enables or bypasses the brickwall lowpass filter Brickwall Low Freq Sets the brickwall lowpass filter cutoff Brickwall High Enable Enables or bypasses the highpass brickwall filter Brickwall High Freq Sets the brickwall highpass filter cutoff Output Gain Sets the output gain TrackPlug Enable Enables or bypasses TrackPlug This parameter is controlled by the bypass button in the TrackPlug preset and menu bar Limiter Enable Enables or bypasses the peak limiter 47 Wave Arts Power Suite 4 6 Specifications Description Channel strip with 10 band EQ RTA 2 compressors gate brickwall filter and peak limiter Operating Systems Windows 7 8 Mac OS X 10 6 or higher Plug in Formats VST AU AAX AAX DSP RTAS DirectX MAS Sampling Rates up to 192 kHz I O Formats mono mono mono stereo stereo stereo 4 7 Presets TrackPlug global presets are listed below The presets are descriptively named Category Preset name
68. harmonic overtones Vintage Warm mode is like Vintage Peak mode but only the positive peaks of the input 33 Wave Arts Power Suite signal are detected using a half wave rectifier Thus the positive and negative swings of the signal are processed differently and this causes the production of both even and odd harmonic overtones With aggressive compression settings the difference in tonal character between the three Vintage modes is striking TrackPlug sidechain modes All dynamics processors track the level of the input signal using a peak or RMS level detector The signal path of the input signal leading to the detector is called the sidechain By inserting an EQ in the sidechain one can build a compressor that responds to particular frequencies This architecture is typically used to create a de esser which is a compressor that reduces the sound of sibilants ess sounds in dialog or vocals Sibilant energy is concentrated around 5 kHz so one can insert a 5 kHz bandpass filter in the sidechain to create a compressor that will reduce gain when sibilants are present This is a standard de esser circuit It works better than applying a high frequency equalizer to reduce sibilants because the equalizer will reduce high frequencies during voiced sound as well as sibilants Trackplug provides four internal sidechain EQ modes Off Internal EQ Internal EQ Compare and Internal EQ Invert The Off mode means the sidechain EQ is no
69. he 64 5 MasterVerb early reflection type is set to Bypass then the early signal is the dry signal Typically if the early reflection type is set to Bypass you will want to set the Late Early mix to 100 all late and then use the Wet Dry mix to mix in dry signal Diffusion Sets the extent to which individual echoes propagating in the reverb are spread in time Low values of diffusion produce reverberation that will have audibly discrete early echoes almost granular sounding whereas high values of diffusion tend to produce a softer fuzzier sounding response The Diffusion parameter affects the diffusion of both the early reflections and the late reverberation algorithm Output Gain Sets the overall output gain in decibels This gain is applied to the output of the wet dry mix therefore the gain does not affect the relative levels of wet and dry sound 5 6 Specifications Multi algorithm reverberator with hall and plate algorithms comprehensive early reflection patterns and envelope control Operating Systems Windows 7 8 Mac OS X 10 6 or higher Plug in Formats VST AU AAX RTAS DirectX MAS Description Sampling Rates up to 192 kHz I O Formats mono mono mono stereo stereo stereo 5 7 Presets Preset Name Default Warm Booth Bright Slapback Vocal Doubler Room Lead Vocal Room Small Bright Room Brassy Bright Room Huge Padded Room Chamber Bright Guitar Chamber Big Drum Chamber Vocal S
70. he decay time of the late reverberation in seconds from 0 5 sec to 60 sec The reverberation time is the time it takes for frequencies near 500 Hz to decay by 60 dB For typical input sounds the reverberation will be essentially inaudible after decaying 60 dB so the Reverb Time specifies the effective length of the reverberation tail in seconds The Reverb Time parameter is independent of the Reverb Size parameter Reverb Size Scales the size of all delays within the late reverberation algorithm ranging from x0 5 50 to x1 5 150 A Reverb Size of x0 5 sounds like a very small space such as a bathroom or closet x1 0 sounds like a concert hall and x1 5 sounds like a very large room such as an aircraft hangar or stadium The Reverb Size parameter is independent of the Reverb Time In general small rooms sound best with short decay times and large rooms sound best with longer decay times Late Damping Parameters The late damping EQ is a three band shelf equalizer that is embedded in the late reverb algorithm The late damping EQ affects the sound that is recirculating in the late reverb algorithm Hence if the late EQ is set to attenuate high frequencies this means the high frequencies will decay faster than the low frequencies Thus the late damping EQ sets the frequency dependent decay time of the reverb The equalization gains are specified as reverberation time scaling factors relative to the nominal reverb time i e the Reverb
71. he direct source two sources if stereo and reflection virtual sources are simulated using variable delay lines this naturally reproduces the Doppler pitch effect When Doppler is disabled sound propagation delay is simulated as if the source is at the listening position origin while the sound propagation gain is simulated normally Hence with Doppler disabled the direct source has a delay of 0 and the room reflections have delays corresponding to twice the wall distances regardless of the position of the source Panorama Enable Enables disables the Panorama plug in When Panorama is disabled it is bypassed the audio input is routed to the output without processing This button is on the title bar at the top left 8 6 Specifications Description Virtual acoustics processor Operating Systems Windows 7 8 Mac OS X 10 6 or higher Plug in Formats VST AU AAX RTAS DirectX MAS Sampling Rates up to 192kHz 1 O Formats mono stereo stereo stereo 8 7 Presets Following is the list of presets that ship with Panorama 1 Vox room 9 Small space 2 Vox room headphones 10 Small space headphones 3 Live room 11 Big space 4 Live room headphones 12 Big space headphones 5 Gymnasium 13 Doppler space 6 Gymnasium headphones 14 Doppler space headphones 7 Medium room 15 Stereo widen 8 Medium room headphones 16 Extreme widen 113 Wave Arts Power Suite 114 License Agreement
72. he menu bar The plug in is enabled when the button is lit and the parameter is On and the plug in is bypassed when the button is extinguished and the parameter is Off Output Level This parameter determines the output gain of the MultiDynamics plug in It ranges from 18 to 18 dB Crossover Slope This parameter determines the crossover slope in dB oct of the bandpass filters The crossover network in MultiDynamics uses either 1st order 3d order or 5th order Butterworth bandpass filters The crossover slope choices are 6 dB oct for the 1st order filters 18 dB oct for the 3d order filters or 30 dB oct for the 5th order filters 87 Wave Arts Power Suite Compression Mode MultiDynamics has two compression modes clean and vintage The clean compression mode is specially designed to minimize harmonic distortion when compressing tonal input signals Consider how a traditional compressor processes a low frequency sinusoid As the amplitude of the sinusoid decreases and passes through 0 the compressor will apply a boost making the sinusoid look more like a square wave and thus adding odd harmonic distortion The Wave Arts clean compression mode avoids this situation by updating its estimate of the signal level after every period of the input signal For the case of a sinusoidal input the clean compression mode will see a purely constant input level and will apply a constant gain in response Hence harmonic distortion i
73. hreshold by 10 dB the expander will attenuate the signal by 20 dB a ratio of 1 4 will attenuate the signal by 40 dB and a ratio of 1 inf will attenuate the signal by infinite dB total mute Note that the MultiDynamics processor limits the gain reduction to 18 dB Band Knee The band knee parameter sets the smoothness of the dynamics response through the knee points Like ratio this parameter can be set individually for each band A Hard knee uses straight lines in the dynamics response a Medium knee uses some smoothing and a Soft knee applies the maximum smoothing Band Attack and Release Time These parameters set the attack time and release times of the dynamics processor The attack time ranges from 1 to 1000 msec the release time ranges from 1 to 2000 msec When compressing the attack time determines how quickly the gain is decreased in response to the signal exceeding the threshold The release time determines how quickly the gain can increase back to its original value The role of attack and release times is reversed when expanding When expanding the release time determines how quickly the gain is decreased when the signal level falls below the threshold and the attack time determines how quickly the gain can increase to its original value Enable The MultiDynamics Enable parameter defines the enable bypass state of the entire MultiDynamics plug in The user interface handles this parameter through the Enable button in t
74. ified or attenuated due to the reinforcement or cancellation caused by adding the delayed reflection to the direct sound This comb filtering effect happens with any reflection but is most noticeable with very early reflections which cause broad frequency effects The result is a timbre modification of the sound possibly hollow sounding which can readily be associated with an enclosed space or a small room Early reflections can also modify the apparent size of the sound source Reflections that arrive in staggered order from the left and right directions confuse our sound localization perception Instead of perceiving a narrowly localized sound source the result will be a fuzzier wider perception Hence the early reflection pattern can affect both the timbre and apparent size of the source Reverberation that contains a lot of high frequency energy in the decay is associated with rooms that have hard reflective walls which do not readily absorb high frequencies Similarly reverberation that is dull sounding is associated with rooms that contain soft materials such as plush carpets and drapes which readily absorb high frequencies In this manner reverberation imparts useful information about the composition of the surrounding space Reverberation is also important for establishing distance cues In a reverberant space when the distance between the source and the listener is increased the level of the direct sound decreases considerably bu
75. imiting Hence the output level as shown by the output meters may slightly exceed the limiter ceiling level when dithering is active This is because the dithering and noise shaping occurs after peak limiting The effect is negligible for typical settings We recommend that a ceiling of 0 1 dB be used for CD and DVD audio mastering to allow a tiny bit of headroom for dither noise Additional headroom may be needed when dithering signals at small bit depths such as 8 bits 6 3 User Interface FinalPlug comes with a number of presets for both track compression and for final processing prior to CD or DVD audio burning If you want to dig deeper than the presets read the following sections on how to use FinalPlug The user interface for FinalPlug is shown at the start of this chapter FinalPlug is separated into a peak limiter section on the left and a truncate dither section on the right each section has its own enable button at the top left If you are using 72 6 FinalPlug FinalPlug as a track compressor just enable the limiter section and leave the truncation section disabled Truncation should only be enabled if you are using FinalPlug as the last step prior to burning a CD or DVD The one exception to this is if you are using the truncation section specifically to obtain bit truncation distortion effects Using the Limiter section The peak limiter section has only three controls threshold ceiling and release time The output
76. in octaves between the half height points in the response For example if the center frequency is 2 kHz the height is 12 dB and the width is 2 octaves then the width is defined at the 6 dB points in the response which will be at 1 kHz and 4 kHz one octave below and one octave above the center frequency respectively 24 280 50 100 200 500 1k 2k 5k 10k 20k Parametric EQ with freq 2000 Hz height 12 dB and width 2 octaves Shelf filter The shelf filter boost or cuts by a fixed amount above or below a corner frequency The shelf filter has parameters of corner frequency and shelf height in dB The figure below shows a low shelf filter with a boost of 12 dB and a corner frequency of 200 Hz The corner frequency is defined at the half height point in the response so for example in the figure below the response is 6 dB at 200 Hz TrackPlug provides three variations of the shelf filter which differ in the shape of the shelf transition standard resonant and vintage The standard shelf filter as shown in the figure has the steepest possible transition without having any overshoot The resonant shelf filter has a variable transition slope defined by a resonance parameter Filter resonances are usually defined using a Q parameter called the quality factor higher values of Q are more resonant and hence sharply tuned TrackPlug uses the width knob to set the resonance in units of Q A shelf with a high resonance has a steep transit
77. inimum to the maximum attenuation Thus the bright red section shows how much the attenuation is changing If the attenuation meter is solid dark red then this indicated the attenuation is constant A large bright section indicates that the gain is rapidly changing across a large range which would be the case for aggressive compression using a fast release time We recommend you enable the auto release option which causes FinalPlug to dynamically choose the optimal release time based upon the program content If the auto release option is disabled you can manually select a release time from 1 msec to 1 sec using the slider Using the Truncate Dither section The main controls in the truncation dithering section are bit depth and noise shape The principal reason for having the truncation dithering feature is to do proper truncation dithering for CD mastering Hence most users will only want to select a bit depth of 16 bits Clicking on the selector button will cycle through the values 8 16 18 20 22 and 24 Clicking on the value itself initiates text editing mode 73 Wave Arts Power Suite allowing the user to type in any value from 4 to 24 Small bit depths can be used for bit truncation effects and are also very useful for demonstrating dithering Assuming you are doing neither of these then select 16 bits for CD mastering or 24 bits for DVD audio mastering Now select a dither shape in the shape selector control One can audi
78. ion but also overshoots symmetrically on each side as shown in the figure below The standard shelf filter is identical to the resonant shelf filter with a resonance Q of 0 707 Finally the vintage shelf filter has a particular sort of asymmetrical overshoot which is quite gentle and pleasing sounding The vintage shelf is modeled after the response shapes of certain analog equalizers 27 Wave Arts Power Suite 230 50 100 200 500 1k 2k 5k 10k 20k Low shelf with freq 1000 height 12 dB 24 2 30 50 100 200 500 1k 2k 5k 10k 20k Resonant low shelf with freq 200 height 12 dB Q 2 24 2 30 50 100 200 500 1k 2k 5k 10k 20k Vintage low shelf with freq 200 height 12 dB Q 1 414 Notch filter The notch filter cuts a specific frequency It has parameters of center frequency and width The width of the notch is measured in octaves between the 3 dB points of 28 4 TrackPlug the filter response TrackPlug s notch filter eliminates the center frequency completely this would correspond to a height of minus infinity dB The notch is commonly used to eliminate hum caused by power line interference In the US power lines oscillate at 60 Hz in Europe and other parts of the world power lines oscillate at 50 Hz Overtones of the hum are common and require an additional notch filter to cancel each overtone Another commonly seen interference tone is caused by the horizontal flyback oscillation of televisions I
79. ion doesn t sound like a room s reverberation the sound is pleasing and often used on vocals and drums Digital reverbs model acoustical and mechanical reverbs using a large number of delay lines through which the signal is recirculated Each reverb algorithm has its own characteristic sound and idiosyncrasies MasterVerb contains two reverb algorithms the Hall and Plate algorithms The Hall algorithm inherited from MasterVerb v4 sounds like a concert hall The Plate algorithm sounds like a plate reverb characterized by a very rapid buildup of densely spaced reflections The Hall algorithm can be scaled to sound like small rooms or very large spaces Scaling a reverb algorithm is done by scaling the delays within the algorithm The Plate algorithm can also be scaled but because the Plate response is so dense the effect of scaling the Plate is less pronounced than scaling the Hall Early reflections MasterVerb models the early reflection pattern using a stereo tapped delay line Each individual reflection is implemented with a single tap in the delay line Typical reflection patterns may have 25 to 60 reflections per channel The output of the tapped delay line is further processed by a diffuser circuit that smears out the response in time causing the reflections to sound less discrete The reflection patterns in MasterVerb are based on real and synthetic reverberation responses Patterns are included for real acoustic spaces including hall
80. k on the pkg file to launch the installer Windows installers are exe files double click on the exe file to launch the installer The installers provide various options for selecting which plug in formats to install and whether to use Pace iLok or Wave Arts licensing for non AAX formats There are also installers for AAX DSP formats These should only be used if you have Pro Tools HDX hardware acceleration 1 3 Registration Wave Arts Power Suite We support two licensing methods Wave Arts licensing and PACE iLok When installing the plug ins you must select which version of the plug ins you wish to use AAX format plug ins always use PACE iLok licensing When you purchase a plug in you will be e mailed a serial number looks like WA PPP XXXX XXXX where PPP is a product code and X is a hex digit Use the serial number to unlock the plug in as described below 1 4 Wave Arts licensing using key codes Prior to registration the plug ins operate in demonstration mode they are fully functional but stop operating after 30 days To unlock the plug in after purchasing go to our Product Registration page select Wave Arts licensing in the Licensing Method menu and enter your serial number and your computer s Machine ID see Machine ID below A key code looks like XXXX XXXX will be displayed and also e mailed to you Then open the plug in select Unlock Plug In from the Tools menu and enter your key code You sh
81. kHz and concentrates above 15 kHz Some boost at low frequencies 44 1 kHz Shape3 Ultra ExtraBit Ultra Exaggerated version of Shape3 44 1 kHz Shape4 Perceptually motivated noise shaping based on the sensitivity of the ear at the threshold of hearing according to classic paper by Robert Wannamaker Journal of the Audio Engineering Society Volume 40 Number 7 8 1992 44 1 kHz Shape4 Ultra Exaggerated version of Shape4 44 1 kHz Shape5 Apogee UV22 High frequency shelf featuring flat noise reduction to 13 kHz with concentration of noise above 17 kHz 96 kHz High frequency shelf with flat noise reduction to 20 kHz and concentration above 20 kHz Intended for 96 kHz operation but can also be used successfully at 88 2 176 4 and 192 kHz 96 kHz Ultra Exaggerated version of 96 kHz shape Mute Dither The Mute Dither parameter disables just the Dithering functionality of the Truncate Dither section When muted the audio is still quantized but no dither noise is added This is useful for auditioning the beneficial effect of the dither For example set the bit depth to 6 enable truncation and toggle the Mute Dither to 76 6 FinalPlug hear the benefit of dither Mute Dither can also be used to obtain bit quantization i e bit crushing effects 6 6 Specifications Description Peak limiter with noise shaped dithering Operating Systems Windows 7 8 Mac OS X 10 6 or higher Plug in Formats VST
82. king on the TOP tab located at the bottom right of the display area The overhead view shows the position of the source with respect to the listener The sound source is drawn as a blue sphere and the head of the listener is drawn at the center of the view The source can be moved in the X and Y dimensions by clicking and dragging in the overhead view If Panorama is processing stereo inputs and the width parameter is not set to Mono then a pair of semi transparent virtual speakers will be drawn to the left and right of the source position These represent the virtual source positions of the left and right input channels The overhead view also shows the current output configuration If the output configuration is set to headphones then the listener is drawn wearing headphones If the output configuration is set to speakers then a pair of opaque speakers is drawn in front of the listener at the angles corresponding to the configuration Note that the overhead view is not drawn to scale in terms of the size of the head and the speakers Finally the background of the overhead view changes depending on the coordinate system in use When Polar coordinates are selected the background shows a circle when Cartesian coordinates are selected the background shows a grid 3D Room View The 3D room view is selected by clicking on the 3D tab located at the bottom right of the display The 3D room view shows a perspective 3D wire frame drawing of the
83. l Control Combination plosive remover de esser and overall dynamics Guitar Acoustic Guitar Tones down mid range mud and adds air Shimmer Warm Acoustic Guitar Brings the sound out of the speakers without losing depth or warmth Noise gate 4 band Noise Gate 4 band noise gating Multidynamics excels at noise reduction without the burbling effects of frequency domain de noisers 6 band Noise Gate 6 band noise gating Adjust all bands threshold and behold MultiDynamics also ships with MultiDynamics v4 presets listed below Basic v4 Hard v4 Bass Comp v4 Vocals v4 Dance Mastering v4 Acoustic Guitar v4 Female De ess v4 Male De ess v4 Medium Enhance v4 Mid Punch v4 Bass Punch v4 Noise Reduce v4 90 8 Panorama 8 Panorama Bypass Undo Copy MPB lt Defauit gt Preset Tools AN IN DIRECT REVERB OUTPUT AZIMUTH ELEVATION DISTANCE 8 1 Overview Panorama is a virtual acoustics processor that can produce stunningly realistic auditory scenes for playback over headphones or loudspeakers Panorama uses 3 D audio and acoustic environment modeling technology to simulate the acoustics of sound propagation in real spaces Panorama features the following e HRTF based binaural synthesis Early room reflection modeling Late diffuse reverberation Reflection surface modeling Doppler pitch effect Distance modeling Processing for playback over loudspeak
84. lair The low band expands to gate out a leaky kick drum while the high band does a profound compression on the snare 5ms of lookahead helps to_ assure that we don t lose a nice tight attack Angry Rock Kick Drum Go ahead Try it Drum mix Drum Craze Extreme expansion is the key to this preset which excels at taking an acoustic drum performance and creating something very processed with resonant artifacts The more room sound on the source the more interesting the preset becomes Try switching into vintage mode for a totally different sound Dries Up Drum Reverb A good dose of expansion on the lo mid band where most reverb lives makes this preset good at tightening up a too wet recording Drums Acoust gt Elec Acoustic drums are again led down the path to Tweakville again by contrasting bands of expansion and compression Excited Live Drums MultiDynamics allows amazing control the over ambience in a track With this preset the room sound of acoustic drums can be enhanced and exaggerated to achieve big rock Vocals Poppa Stoppa Obnoxious vocal p sounds check in but they don t check out 89 Wave Arts Power Suite Broadband De ess A simple de esser which should reel in most over zealous performances Multi Band De Ess When simple de essing doesn t solve the problem or causes unwanted artifacts try this preset Solo each band as you search for your trouble spots Voca
85. lay on a separate axis is shown the input and threshold levels The threshold levels per band are plotted with a thick red line the red ball can be dragged up and down to change the threshold level The peak input levels in each band are plotted in the background using pale green yello Band edges are plotted using vertical white lines The edges can be moved by clicking and dragging a line An entire band can be moved by clicking and dragging in the center area between the top and bottom plots One band is always selected for editing the active band is drawn with a blue background The active band s parameters can be edited using the knobs below the band display The active band s dynamic response can be edited in the dynamics response display described later Band control buttons Clicking the Add button will create a new band by splitting the current band in two The new band will inherit all the parameters of the current band Up to 6 bands can be created at once Clicking the Delete button will delete the currently selected band Clicking the Bypass button will enable disable bypassing of the current band Bypassing a band means that the dynamics processing is bypassed hence the audio is passed to the output but the gain of the band is fixed at 0 dB Bypassing a band is useful for hearing the action of the dynamics processor on a band Clicking the Solo button will enable disable soloing of the current band Soloing a band causes the o
86. leasing increasing the gain back to the nominal level as set by the makeup gain A short release will restore gain immediately in which case the drum sound will have essentially the same decay it had originally A longer release time will restore gain more slowly thus increasing the decay time of the original drum sound So the original BUMPH of the drum sound becomes BOOOOMMMMPH after compressing That s punch The ability to change the decay time of acoustic instruments after they have been recorded is a really powerful capability of compressors When gating the attack and release time parameters are reversed The release time controls low fast the gain is turned down and the attack time controls how fast the gain is restored Consider the example of a drum recording containing background noise assuming the threshold has been set just above the background noise and the ratio is large When the signal goes below threshold the gate kicks in and begins reducing gain A short release time will decrease gain rapidly abruptly cutting off the decay of the drum Longer release times cause the gain to decrease more slowly which may sound more natural but also allow the noise to be audible at the end of the decay On the next drum hit the gate will restore gain according to the attack time Using a short attack time is prudent in this case otherwise the attack of the drum will be lost due to the slow attack fade in of the gate So when gatin
87. lection Front Enable Enables disables the front wall reflection Back Enable Enables disables the back wall reflection Top Enable Enables disables the top wall ceiling reflection Bottom Enable Enables disables the bottom wall floor reflection Left Distance Sets the distance of the left wall from the listener Right Distance Sets the distance of the right wall from the listener Front Distance Sets the distance of the front wall from the listener Back Distance Sets the distance of the back wall from the listener Top Distance Sets the distance of the top wall ceiling from the listener Bottom Distance Sets the distance of the bottom wall floor from the listener 108 8 Panorama Left Material Sets the material of the left wall Right Material Sets the material of the right wall Front Material Sets the material of the front wall Back Material Sets the material of the back wall Top Material Sets the material of the top wall ceiling Bottom Material Sets the material of the bottom wall floor Reverb Parameters REVERB CE PRESET Reverb Decay Time Sets the decay time of the reverberation in seconds from 0 5 sec to 60 sec Also known as the reverberation time the decay time is the time it takes for frequencies near 500 Hz to decay by 60 dB For typical input sounds the reverberation will be essentially inaudible after decaying 60 dB so the Decay Time specifies the
88. lly generated using recursive filters filters that have feedback elements such as comb and allpass filters Panorama uses a high quality diffuse reverberator based on an allpass feedback loop topology The character of the reverberation is controlled by the following parameters reverberation time room size pre delay and early and late damping frequencies The reverberation time is the 60 dB decay time of the reverb The room size parameter alters the pattern of reflections and the character of the late reverberation to simulate various room sizes The early damping parameter controls the amount of high frequencies admitted into the reverberator The late damping frequency parameter controls the absorption of high frequencies in the late reverberation High damping frequencies result in a bright sounding room low damping frequencies result in a warm sounding room 8 2 12 Distance Cues The principal cue for distance is the loudness of the sound A sound source will be louder when it is closer to the listener than when it is farther away However this cue is often ambiguous because the listener doesn t know a priori how loud the source is Thus a moderately loud crashing sound could be perceived as a quiet close crash or a distant loud crash Another important cue for distance is the loudness of reverberation When sound is produced in a reverberant space the associated reverberation may often be 99 Wave Arts Power Suite perceived
89. lug Clicking on the button will step to the next preset Clicking on the preset name will display a popup menu of the current factory and user EQ presets Input and Gain meters Each section meter shows an input meter on the left and a gain reduction meter on the right The input meter shows the signal level coming into the dynamics section and the gain reduction meter shows the amount of gain reduction being applied The input meter display depends on the dynamics mode and sidechain mode If the dynamics mode is peak the meter shows peak values if the mode is RMS the input meter shows RMS values RMS values are typically 3 to 10 dB lower than peak values depending on the material If the sidechain is in use then the input meter shows the signal level resulting from sidechain processing The meters show both the minimum and maximum values since the last meter redraw The meter is drawn with a dark color up to the minimum value and drawn with a lighter color from minimum to maximum value These types of meters are a Wave Arts innovation The min max meters let you see at a glance how much a signal is modulating If the signal level is constant the meter bar will be a solid dark color If the signal level is modulating rapidly this is shown by a large light section The size of the light section indicates how much a signal is changing dynamically The vertical range of the dynamics attenuation meters can be selected by the user Simply SHI
90. n be shared amongst all configurations of one plug in with the obvious caveat that not all features will be enabled or editable in the reduced feature configurations Also changing from one configuration to another may cause the settings to change For example when changing from the full 6 band MultiDynamics to 3 band and back to 6 band only 3 bands will be enabled even if all 6 were originally enabled 1 10 Plug in Instance Counts for AAX DSP The table below gives the maximum number of instances that will run on a single DSP chip for each plug in configuration at 48 kHz 96 kHz and 192 kHz sampling rates for mono and stereo formats These instance counts are valid as of the writing of this manual future updates will likely increase instance counts as the plug ins are further optimized Configuration Format 48 kHz 96 kHz 192 kHz TrackPlug momo 3 3 2 A stes al BA Al TrackPlug EZGC_____ mono 6 6 y stero 6 5 2 TrackPlugE7C_____ mono 8 8 4 MA 2 E A A Multibynamics mono 3 2 1 oo Qu istere 3 1 0 EA E teo ee ee MultiDynamics 3 band mono 6 3 1 AS eros a fee Alter A A a Po stereo 17 17 8 e i sh p i Un he Raja N 10 1 Introduction 11 2 Plug in Control Operation 2 Plug in Control Operation 2 1 Knobs FREQUENCY Please refer to the following guide for information about the various ways you can use knobs Function
91. n the US NTSC TV sets oscillate at 15 734 Hz In Europe PAL and SECAM TV sets oscillate at 15 625 Hz Often you will find these tones in acoustic recordings made anywhere near TV sets 24 2 30 50 100 200 500 1k 2k 5k 10k 20k Notch filter with freq 60 width 0 1 octaves Bandpass filter The bandpass filter is the opposite of the notch it passes only the frequencies in a band around the center frequency The bandpass filter has parameters of center frequency and width The width of the bandpass is measured in octaves between the 3 dB points of the filter response The bandpass filter would typically only be used in sound design to simulate the sound of a reduced frequency response such as a telephone One could also use the brickwall filters for this The bandpass filter rolls off at 12 dB per octave on each side of the center frequency much gentler than the brickwall which rolls off at 60 dB octave The bandpass filter can also be used in analysis to isolate a particular frequency For example if there is a contaminating tone in a signal one can sweep a narrow bandpass filter to isolate the tone then switch to a notch filter to eliminate the tone 29 Wave Arts Power Suite 2 30 50 100 200 500 ik 2k 5k 10k 20k Bandpass filter with freq 2 kHz width 1 octave Lowpass and highpass filters TrackPlug also provides lowpass and highpass filters These pass frequencies up to a corner frequency then roll off at 12 dB
92. na are described in the following sections 8 2 9 Room Reflections and Reverberation When an object in a room produces a sound a soundwave expands outward from the source reaching walls and other objects where sound energy is both absorbed and reflected Technically speaking all reflected energy is called reverberation Assuming a direct path exists between the source and the listener the listener will first hear the direct sound followed by reflections off nearby surfaces called early reflections After a few tenths of a second the number of reflected waves becomes very large and the resulting reverberation is characterized by a dense collection of soundwaves traveling in all directions called diffuse reverberation The time required for the reverberation to decay 60 dB below the initial level is defined as the reverberation time Generally reverberation in a small room decays much faster than reverberation in a large room because in a small room the soundwaves collide with walls much more frequently and thus are absorbed more quickly than in a large room Reverberation is an important acoustic phenomena There is at most one direct path from the source to the listener whereas there may be millions of indirect paths particularly in a room where a sound can bounce around hundreds of times before being absorbed Thus in typical listening situations most of the energy we hear from a sound source is actually reflected energy The perc
93. ncy response at time 0 Changing the late damping affects the decay slope The characteristic plot always shows the 100 wet response assuming wet delay is O and is not affected by the Wet Dry Wet Delay or Output parameters 5 4 Parameters This section describes all the internal parameters of MasterVerb as would be displayed by a generic parameter value plug in interface Most of these have a one to one correspondence with controls on the user interface Parameter name Enable Reflection Type Reflection Size Early Damping Lo Freq Early Damping Lo Gain Early Damping Mid Gain Early Damping Hi Freq Early Damping Hi Gain Late Reverb Type Late Reverb Decay Time Late Reverb Size Late Damping Lo Freq Late Damping Lo Scale Late Damping Mid Scale Late Damping Hi Freq Late Damping Hi Scale Envelope Enable Envelope Thresh Envelope Attack Time Envelope Hold Time Envelope Release Time Dry Delay Wet Delay Values O Off 1 On O Bypass 1 Halll 2 Hall2 3 Cathedral 4 Rooml 5 Room2 6 Chamberl 7 Chamber2 8 Platel 9 Plate2 10 Echo 11 Ping pong x0 5 to x1 5 20 to 640 Hz 24 to O dB 24 to 0 dB 640 to 20 000 Hz 24 to 0 dB O Hall 1 Plate 0 5 to 60 sec x0 5 to x1 5 20 to 640 Hz x0 25 to x4 x0 25 to x4 640 to 20 000 Hz x0 25 to x4 O Off 1 On 36 to 0 dB 1 to 2000 msec 1 to 2000 msec 1 to 2000 msec 0 to 500 msec 0 to 500 msec 61 Wave Arts Power Suite
94. nd EQ v4 The Works v4 Special Xtreme Beat v4 Xtreme Vocals v4 What Dynamics v4 Noise Filter v4 Telephone v4 Telephone Screaming v4 EQ presets Trackplug EQ presets are listed below Category Preset name General 4 band Default Brighter Warmth Biting Mid range 49 Wave Arts Power Suite Fat Bottomed Pipeline Xylophonated Comb the Area Telephone Transmission Old Time Radio Show Noise reduction Hum 60Hz Hum 50Hz NTSC PAL SECAM Drums Simple Kick Solution Another Kick Drum Saved D112 Kick Industry Kick Snare 1 Old Snares Never Die Typical Snare Better Snare Classic Snare Snare De ringer Alien Snare Drum Narrow Drum Band Tingly Sidestick Synthetic Sidestick Tom Control Vintage Toms Make Room for Tom When Toms Ruled the Earth Punchy Tom Cymbal Sizzle Subtle Cymbal Shine Jazzy Ride Cymbal Ride Cymbal Recipe Hi Hats Guitar Acoustic Gtr 1 Acoustic Gtr 2 Nu Metal Guitar Typical Death Metal Filtered Funk Guitar Ukelelephone Bass Modern Rock Bass Gave my Bass Wood Deep Bass Bass DI Helper Snappy Upright BAss Vocals Male Vox 1 Female Pop Vox Airy Back Vox Big Ballad Vox Stimulated BVox 50 4 TrackPlug Keyboard Strings Brigh
95. ned to share adjacent edges e g Band1 Hi is always the same as Band2 Lo etc Band Dyn Enable This parameter defines the enable bypass state of the dynamics processor in each band The user interface handles this parameter through the Bypass button When a band is Bypassed its DynEnable is Off when not bypassed its DynEnable is On Band Thresh This is the input threshold in dB for the band which ranges from 72 to O dB Band Lo Gain and Band Hi Gain When the signal is below the input threshold the dynamics processor will apply the Lo Gain to the signal when above the threshold the processor will apply Hi Gain to the signal See the description of the dynamics response display for more information Both Lo Gain and Hi Gain range from 18 dB to 18 dB Band Ratio The ratio parameter determines how quickly the dynamics processor will change gain as the signal level passes the threshold The ratio ranges from 1 to 50 When compressing the ratio is interpreted as ratio 1 using the classic definition of compressor ratio A ratio of 2 1 means that if the signal exceeds the threshold by 10 dB the compressor will attenuate the signal by 5 dB a ratio of 4 1 will attenuate the signal by 7 5 dB and a ratio of inf 1 will attenuate the signal by 10 dB 86 7 MultiDynamics When expanding the ratio is interpreted as 1 ratio using the classic definition of expander ratio A ratio of 1 2 means that if the signal is below the t
96. nes the attenuation per doubling of distance Gain and slope parameters are provided for the direct source the early reflections and the reverberation For the purposes of creating an effective sounding scene it is often necessary to tweak the parameters to get the desired distance effect For example when synthesizing virtual acoustic scenes it can sound unnatural if the reverberation doesn t attenuate sufficiently with increasing distance It also becomes difficult to localize distant sound sources if there is too much reverberation Also attenuating the direct source by 6 dB per doubling of distance usually sounds excessive although it is physically accurate Natural acoustics has a very large dynamic range when creating virtual acoustic scenes it is often necessary to compress the dynamic range by decreasing distance effects i e using lower slopes than physical acoustics would dictate direct gain reverb gain 1 2 4 8 16 distance Figure 8 5 A typical distance model used by Panorama The direct source has a gain of 0 dB and a slope of 6 dB The reverb has a gain of 12 dB and a slope of 1 5 dB 100 8 Panorama 8 2 13 Doppler motion effect The Doppler motion effect is commonly heard in nature as a pitch change when a speeding object passes a listener When the object is approaching the listener the pitch is higher than the resting pitch of the object This is because in the time it takes the object to emit one wa
97. new coordinate system This can change the source position if switching from Cartesian to polar coordinates and the distance is greater than 20 units in this case the distance is limited to 20 When automating the source position you must be careful to automate the proper coordinate parameters For example if Cartesian coordinates is selected then automating any of the polar coordinates will have no effect Stereo Width This sets the width in degrees of stereo inputs measured from the center position The virtual left speaker is positioned width degrees to the left of the source and the right speaker is positioned width degrees to the right of the source If the width parameter is set to Mono the stereo input is mixed to a single 107 Wave Arts Power Suite monophonic virtual source When processing mono sources the width parameter is fixed at Mono Reflection Parameters REFLECTION O rn gt PRESET CJ AA MATERIAL DISTANCE MATERIAL 2 DISTANCE MATERIAL DISTANCE gt DE _ y 408 ft 16 98 ft MATERIAL DISTANCE MATERIAL DISTANCE MATERIAL DISTANCE The full set of reflection parameters are displayed as shown above by clicking on the REFLECTION tab Click either the DIRECT or REVERB tab to show the reduced set of reflection parameters which are the wall distances without enables or materials Left Enable Enables disables the left wall reflection Right Enable Enables disables the right wall ref
98. ng headphones virtual sound source perceived here sound source digital filters Figure 8 2 Binaural synthesis of a single source Binaural synthesis works extremely well when the listener s own HRTFs are used to synthesize the localization cues However measuring HRTFs is a complicated procedure so 3D audio systems typically use a single set of HRTFs previously measured from a particular human or manikin subject 8 2 4 Non I ndividualized HRTFs Localization performance generally suffers when a listener listens to directional cues synthesized from HRTFs measured from a different head called non individualized HRTFs Human heads are all different sizes and shapes and there is also great variation in the size and shape of individual pinnae This means that every individual has a different set of directional cues The greatest differences are in the tonal transformations at high frequencies caused by the pinnae It is clear we become accustomed to localizing with our own ears and thus our localization abilities are diminished when listening through another person s ears Our uniqueness as individuals is a significant limitation of practical 3D technology The use of non individualized HRTFs results in two particular kinds of localization errors commonly seen with 3D audio systems front back confusions and elevation error A front back confusion results when the listener perceives the sound to be in the front when it should be in
99. nter Select next parameter to edit Select previous parameter to edit Shift Tab Shift Tab Typing ESC causes the text to revert its original value before editing You ll find that many parameters such as frequency will recognize units typed into the text field The following values when typed into a frequency value box are equivalent 2k 2 kHz 2000 2000 Hz 2 3 Selector button The selector button cycles through a number of fixed values Click on the button to go to the next value Click on the text to display a pop up menu of the available values The table below describes the functionality of the selector button Function Go to next value Click on the knob Click on the knob Go to previous value l Shift click on knob Shift click on knob Display pop up menu of all choices Click on text Click on text 14 2 4 Sliders Function RELEASE Increase Decrease a parameter value Fine adjustment increase decrease Reset slider to default value 2 Plug in Control Operation Mac Click on the slider handle drag Windows Click on the slider handle drag up down up down Right click drag Shift click drag i ai up down Shift click drag up down Command click Or Double click Control click Or Double click Lighted buttons show a toggle state A green orange or yellow light indicates on and a black extinguished light indicates off
100. nty of any kind either express or implied including without limitation the implied warranties or merchantability fitness for a particular purpose or noninfringement The entire risk arising out of use or performance of the SOFTWARE PRODUCT remains with you NO LIABILITY FOR CONSEQUENTIAL DAMAGES In no event shall Wave Arts or its suppliers be liable for any damages whatsoever including without limitation damages for loss of business profits business interruption loss of business information or any other pecuniary loss arising out of the use of or inability to use this Wave Arts product even if Wave Arts has been advised of the possibility of such damages Because some states jurisdictions do not allow the exclusion or limitation of liability for consequential or incidental damages the above limitation may not apply to you MISCELLANEOUS If you acquired this product in the United States this EULA is governed by the laws of the State of Massachusetts If this product was acquired outside the United States then local laws may apply 116 Support Support For assistance please send email to support wavearts com If you are having problems with a plug in please include the following information plug in name operating system and host software you are using The version numbers are also helpful For software updates revision history frequently asked questions FAQ and more please visit our website at Ww
101. onophonic sound that is reproduced in the virtual space at a particular location The stereo input is split into left and right channels and each channel is processed as a virtual source Virtual source processing starts with a gain stage to simulate the attenuation of sound as a function of distance from the sound to the listener The signal is then processed by filters that simulate the reflections of sound off reflecting materials for direct sources those that are not reflected this filtering is bypassed The signal is then processed through a variable delay line that simulates the propagation of sound through air The variable delay naturally creates the Doppler pitch effect Finally the signal is processed to simulate the diffraction of sound by the torso head and ears of the listener labeled 3 D filters in the figure The resulting left and right ear signals are simulations of the signals that would reach a listeners ears in a real listening environment These signals are summed to a stereo output bus 102 8 Panorama Simulating the acoustical reflections off the walls floor and ceiling is accomplished by creating additional virtual sources at mirror image locations behind each wall surface The reflected virtual sources are filtered according to the sound absorption properties of the reflecting surface material Note that the reflections are monophonic even with stereo sources The outputs of the reflected virtual sources are
102. or in TrackPlug and MultiDynamics has been jettisoned in favor of the superior Vintage processor e The Vintage dynamics processor has been slightly modified for increased efficiency on the TI DSPs e MultiDynamics dynamics modes are now Peak and RMS RMS mode has been added 1 Introduction e Dynamics lookahead is maximum of 1 msec for both TrackPlug and MultiDynamics due to limited memory on DSP e The limiting algorithm in FinalPlug has been rewritten for greater efficiency on TI DSPs and has a new auto release algorithm e The dither algorithm in FinalPlug is now 32 bits rather than 64 bits We believe the above changes were necessary to provide reasonable instance counts when running on DSP without diminishing sound quality 1 9 Reduced configurations for AAX DSP Even with the above DSP optimizations the full versions of TrackPlug MultiDynamics and FinalPlug use a considerable amount of CPU and memory on the HDX DSPs This is because they were originally designed for native operation and have a large feature set In typical native operation many of these features may be disabled and will not consume any CPU But on HDX DSP we must allocate enough CPU for the worst case situation of all features enabled For example TrackPlug needs to allocate enough CPU to run all 10 bands of EQ plus both brickwall filters with all of the Gate Compl and Comp2 dynamics enabled all three running dual sidechain EQs pl
103. ould see a message saying your registration was successful If you have purchased a plug in suite when you unlock any one of the plug ins within the suite the entire suite will be unlocked 1 5 Machine ID Your Machine ID is a unique 5 digit identification number our plug ins generate based on properties of your computer Our key code system uses the Machine ID and the product serial number to generate a key code that will unlock the plug ins on your specific computer The Machine ID is NOT used for PACE licenses you will redeem a PACE code instead To find your Machine ID Open the plug in within a host application click on the Tools button inside the plug in window and select the About item Open User Manual Unlock Plug In PIT Check for Updates Visit Website 1 Introduction This will show the About box displaying information about your plug in including the Machine ID for example 0 About WNVA wave arts FinalPlug 5 v5 82 Audio Unit 64 bit Build Date Mar 18 2015 Machine Demo mode Time expired User Manual Unlock OK On Mac OS X and Windows systems if your computer has Ethernet the Machine ID is based on your hardware Ethernet MAC address Therefore you should not have to re register your plug ins if you update your operating system or reformat your hard drive On Windows systems changing the network configuration can sometimes result in your Machine ID changing If this ha
104. ound for a standard stereo production we recommend using the speakers setting 20 or 30 degrees because the result will probably be listened to over speakers The flying track will stand out from the rest of the stereo mix created using standard stereo panning If you want to create a special effect for headphone listeners e g iPod MP3 listeners use the headphone mode This mode gives the most realistic 3D experience Listening to headphone mode audio over loudspeakers is not bad but you won t get the really wide sounds that are possible using crosstalk cancellers In contrast listening to crosstalk cancelled audio over headphones can sound pretty unnatural Crosstalk cancellation can create out of phase low frequencies which can impart an unusual pressure sensation to the ears when listened to over headphones This effect is more noticeable if the entire mix has been processed with a crosstalk canceller Units Sets the distance units to either feet or meters Changing the units will scale the dimensions of the space but all parameter values remain the same Use the meters units to get larger spaces Output Gain Sets the output gain of Panorama in dB This control is located under the output meters HRTF Sets the head model used to synthesize binaural cues See the tutorial on 3D audio for more information about HRTFs and binaural synthesis Two head models are compiled into Panorama the Human set and the MIT KEMAR
105. ppens you can get a keycode for the new Machine ID on our Register page and the plug in should stay registered even if your system alternates between Machine IDs Contact us if you need further assistance such as additional keycodes 1 6 PACE iLok licensing All our plug ins support PACE iLok Prior to activation the plug ins will allow you to start a 14 day trial by creating an iLok account To unlock the plug in after purchasing go to our Product Registration page select PACE iLok and enter your serial number A PACE redeem code looks like XXXX XXXX XXXX XXXX XXXX XXXX XXXX XX will be displayed and also emailed to you There are two ways to redeem the code and generate a license When opening the plug in a dialog window will appear and give you the option to Activate the plug in you can paste the PACE redeem code there and proceed to create or login to an iLok account and then transfer the license to an iLok or your machine Otherwise go to http www ilok com create an iLok account and download and install the iLok License Manager Within the Wave Arts Power Suite manager under the Licenses menu select Redeem Activation Code and paste your redeem code Then transfer the license to either an Lok dongle or your machine The plug in will run only if it can find a license on an iLok or the machine When purchasing a plug in suite the redeem code will generate multiple licenses one per plug in in the suite but th
106. r only and the right ear signal should go to the listener s right ear only This is easy to achieve when the listener is using headphones When using loudspeakers however there is significant crosstalk between each speaker and the opposite ear of the listener A large portion of the left speaker signal will go to the right ear of the listener and similarly a large portion of the right speaker signal will go to the left ear of the listener Crosstalk is depicted in the figure below as the acoustic paths from each speaker to the opposite ear The crosstalk severely degrades localization performance and must be eliminated Figure 8 3 Crosstalk when listening to stereo loudspeakers 95 Wave Arts Power Suite 8 2 7 Crosstalk Cancellation Fortunately it is possible to build an elaborate digital filter called a crosstalk canceller that eliminates crosstalk The crosstalk canceller adds a cancellation signal to each of the two channels of audio such that when the listener is properly positioned between the loudspeakers the crosstalk is acoustically cancelled at the listener s ears The listener must be centered between the two loudspeakers in order for the crosstalk to be cancelled In 3D audio parlance the listener must be in the sweet spot to get the full 3D effect Provided the listener is centered between the loudspeakers crosstalk cancellation is relatively insensitive to front back motions of the listener however
107. rument sounds can be used for sound design It is possible to split up the sound into different frequency ranges apply compression to some and expansion to others and dramatically change the character of the original sound Dynamics controls in MultiDynamics The dynamics controls in MultiDynamics are a bit different than in traditional compressors that have a threshold ratio and makeup gain MultiDynamics has Threshold and Ratio but the makeup gain is replaced with two gain controls called Lo Gain and Hi Gain Simply put Lo Gain is the gain applied to the signal when the signal is below threshold and Hi Gain is the gain applied when the input signal is above threshold The steepness of the transition from Lo Gain to Hi Gain as the signal passes through the threshold is determined by the ratio control much like a traditional compressor or expander With a high ratio the transition is very fast and with a small ratio the transition is slow Unlike a traditional compressor which continues to apply more gain as the signal goes further over threshold MultiDynamics limits the maximum amount of gain to Hi Gain Similarly a traditional expander will continue to apply more attenuation as a signal falls below threshold in MultiDynamics the maximum attenuation is limited to Lo Gain Thus the Lo Gain and Hi Gain controls define the maximum range of gains that can be applied to the signal Both Lo Gain and Hi Gain range from 18 to 18 dB so the maximum
108. s a 56 5 MasterVerb cathedral and some rooms and for synthetic reverb responses including chambers and plates The Anechoic reflection type has a single unit gain reflection at time O in both left and right channels so it simply passes the input signal unchanged to the diffuser circuit The user should note that the CPU processing required for the early reflection generator is about the same as for the late reverb algorithm hence if you need a really CPU efficient reverb try bypassing the early reflection generator by selecting the Anechoic pattern then set the late early mix to 100 late Envelope MasterVerb s envelope control is used to alter the amplitude response of the reverb When the envelope control is enabled the output of the reverberator is amplitude modulated by the envelope The envelope itself has three segments attack hold and release and is triggered when the input signal exceeds the threshold This is shown in the diagram below The envelope is nominally at gain 0 When the input signal exceeds the threshold the envelope is triggered This causes the envelope to increase to unit gain with rate specified by the attack time parameter The envelope then holds for the hold time and finally releases according to the release time parameter The envelope will retrigger return to attack state whenever the input exceeds the threshold threshold envelope attack hold release Diagram of envelope tri
109. s avoided Clean mode should not be used for transient shaping or low end fattening because it is specifically designed to retain transients and to avoid distortion of low frequency tones Clean mode is excellent for gentle compression of material or for harder compression where you want to retain transients and clean low frequency tones The vintage compression mode works just like a traditional analog compressor If you set it up to compress a sinusoid into a square wave it will do that Hence with aggressive settings of ratio and attack and release time you can cause the vintage compressor to create rich overtones With less aggressive settings it sounds like a very nice traditional compressor For AAX formats the modes are limited to Peak and RMS using the Vintage compressor See the description of Peak and RMS modes in the TrackPlug chapter 7 6 Specifications Description Multi band dynamics processor Operating Systems Windows 7 8 Mac OS X 10 6 or higher Plug in Formats VST AU AAX AAX DSP RTAS DirectX MAS Sampling Rates up to 192 kHz 1 O Formats mono mono mono stereo stereo stereo 7 7 Presets Multidynamics is shipped with a variety of presets described in the table below After loading a factory preset the first thing to do is to adjust the All bands Threshold to suit your input signal After that tweak away Category Preset Name Description Templates Default A basic three band compres
110. s can be eliminated using a lowpass brickwall The signal is processed next by the gate The gate is commonly used to attenuate background noises when the main signal is not present The gate can be used to reduce amplifier noise microphone leakage etc If the EQ routing is set to pre the EQ runs after the gate and before the compressors if set to post it runs after the compressors EQ is the principal tool to shape the tonal character of the sound The realtime spectrum analyzer runs either just before or after the EQ section depending on whether the RTA is set up to be pre or post EQ The two compressors run next They can be configured in a variety of ways from slow auto gain control to faster compression to aggressive distorting compression The compressors can also be configured as de essers or de ploders by using the side chain EQs If the EQ routing is set to post the EQ runs after the compressors Some engineers prefer this configuration because they find that compression dulls the sound requiring some EQ after compression The output gain and peak limiter run last The peak limiter can be used as a hard compressor loudness maximizer by cranking up the output gain or used simply to prevent peaks from exceeding 0 1 dB This latter function is important for Pro Tools RTAS format because signals that exceed 0 dB are hard clipped and sound very distorted TrackPlug E7GC Configuration For AAX DSP the E7GC config
111. s three dynamics processors a gate and two compressors Each compressor has 5 modes Clean Peak Clean RMS Vintage Peak Vintage RMS and Vintage Warm while the gate is limited to using one of the two Clean modes AAX format has a different selection for dynamics modes Each processor has 3 modes Peak RMS and Warm while the gate is limited to using either Peak or RMS mode These modes correspond the to the Vintage modes in non AAX versions of TrackPlug for AAX the so called Clean modes have been omitted The clean modes are inherited from earlier versions of TrackPlug they prevent harmonic distortion when processing tonal sounds even when using fast attack release times Consider how a traditional compressor responds to a sinusoidal waveform it will attack on the peaks and release during the troughs essentially shaping the sinusoid into a square wave and adding odd harmonic distortion In contrast the TrackPlug clean processor estimates the signal level once after each complete waveform hence the clean mode sees a sinusoidal input as having a purely constant amplitude Thus the clean compressor and clean gate will not shape each waveform The clean modes are indispensable for gating They prevent any waveform distortion as the gate is turning on or off The gate is limited to only using the two clean modes The clean modes are also useful when you are compressing aggressively but you want to preserve attack transients sa
112. sed by the early damping equalizer The wet signal is then split into early and late paths the early path is processed with the early reflection generator while the late path is processed by the late reverb Note that the late damping equalizer is embedded in the late reverb algorithm The early and late reverb components are mixed and fed into the envelope processor The envelope processor is triggered by the non delayed dry input signal although this path is not shown in the diagram The final wet signal is mixed with the delayed dry signal and the result is processed by the final output gain 5 2 About MasterVerb About Reverberation When an object in a room produces a sound a soundwave expands outward from the source reaching walls and other objects where sound energy is both absorbed and reflected Assuming a direct path exists between the source and the listener the listener will first hear the direct sound followed by reflections off nearby surfaces called early reflections After a few tenths of a second the number of reflected waves becomes very large and the resulting reverberation is characterized by a dense collection of soundwaves traveling in all directions called diffuse reverberation 54 5 MasterVerb The time required for the reverberation to decay 60 dB below the initial level is defined as the reverberation time Generally reverberation in a small room decays much faster than reverberation in a large room becaus
113. select the Save option You will be asked to name the preset and the preset will be saved in the set of User presets If you supply the same name as an existing user preset the preset will be overwritten with the new preset without any warning notice 3 10 Import User presets can be written to files using the Export function and read from files using the Import function Selecting the Import option will first ask if you want to replace or merge the imported presets Replacing 19 Wave Arts Power Suite causes your current set of user presets to be deleted and replaced with the presets read from the file merging will add the presets read from the file to your set of User presets Then you will be asked to choose a preset file for importing and the presets are read from the file Import can also be used to convert presets from an older version of the plug in to the current version If the plug in detects presets from an older version and it knows how to convert them to the current version it will ask you if you want to convert the older presets to the current format 3 11 Export Selecting the Export option will first ask if you want to replace or merge the exported presets Replacing causes the presets in the file to be deleted and replaced with the exported user presets merging will add the user presets to the presets in the file Then you will be asked to choose a preset file for exporting
114. sor responds to out of band minus in band energy The Invert mode is useful to hear the opposite effect of the Compare mode so if the Compare mode is used to de ess the Invert mode will actually isolate the ess sounds Trackplug also supports external sidechain input in certain host formats AU MAS and RTAS formats only When using TrackPlug AU or RTAS with Logic or Pro Tools 34 4 TrackPlug select the source of the sidechain within Logic or ProTools When using TrackPlug MAS with DP select the bus you wish to use in the Tools menu of TrackPlug Then in TrackPlug select one of the External options for the sidechain type to use the external sidechain source rather than the internal sidechain All the same EQ options are supported for both internal and external sidechain Peak limiting The TrackPlug peak limiter when enabled prevents any peak from exceeding 0 1 dB It uses a 2 msec lookahead buffer to detect and respond to peaks before they occur and hence incurs a 2 msec latency when enabled When it detects a peak that will clip it turns down the gain very quickly then slowly restores the gain after the peak has passed The TrackPlug limiter is very transparent sounding when used for light limiting and it can be used to compress heavily However the TrackPlug limiter was designed to be very CPU efficient and it is not as good as the FinalPlug limiter for heavy limiting and volume maximization 35 Wave Arts Power Sui
115. sor with mild compression settings 88 7 MultiDynamics Default 1 band Template for a single band processor that does nothing ratio is 1 lo gain and hi gain are both 0 dB and threshold is O dB To make a compressor lower the hi gain increase the ratio and lower your threshold to suit To make an expander lower the lo gain or raise the hi gain Default 3 band Template with 3 bands Default 6 band Template with 6 bands Basic Expander Basic 1 band expander preset set up as a noise gate Basic Compressor Basic 1 band compressor vintage mode with 4 1 ratio and 1ms of lookahead Mix Bright PopRock Mix This 5 band compressor really does the trick of pulling the music up around the lead vocal without sacrificing its clarity You get a lot more perceived volume and thickness without giving up every last shred of dynamics Wide Punchy Mix A 4 band compressor with a punchy mid range and beefy bottom end Drums Taming the Snare This single band expander gates out the noise while boosting the snare hits and adding attack Increase attack time to make your drum sound last longer Snare Ringer Sometimes a ringy snare sound is just what you need This preset makes some sonic room for the bright part of the snare drum attack and then lets the low mids fill in the sound 20ms later Note that the frequency band from 600 1 4 kHz is compressing while other bands are expanding Snare Repair with F
116. summed to the output bus The late diffuse reverberation in the space is simulated using a high quality reverberator A monophonic mix of the input is sent to reverberator and the stereo reverb output is summed to the output bus The stereo output bus carries the signals that should be reproduced at the ears of the listener If loudspeaker playback is selected the signals are processed by a crosstalk canceller If headphone playback is selected the crosstalk canceller is bypassed Finally the signal is processed by the output gain stage and level metering 8 2 15 Virtual Speakers for Stereo Inputs Panorama uses virtual speakers to simplify the placement of stereo inputs When processing stereo inputs a pair of virtual speakers is created at locations to the left and right of the source position on the same horizontal plane The spacing of the virtual speakers is determined by the width parameter The virtual left speaker is positioned width degrees to the left of the source and the right speaker is positioned width degrees to the right of the source If the width parameter is set to Mono the stereo input is mixed to a single monophonic virtual source 8 3 User Interface Panorama s user interface is shown below peee Bypass Undo Copy W B Default gt Preset Tools Panorama WANA WA DIRECT REVERB OUTPUT PRESET PRESET GAIN SLOPE 103 Wave Arts Power Suite Overhead View The overhead view is selected by clic
117. t Dynamics Dynamics processors are extremely useful and versatile tools All dynamics processors work by tracking the level of the input signal and applying a gain in response to the input level Compressors turn down the gain when the input exceeds a threshold level whereas gates expanders turn down the gain when the input goes below a threshold Compressors are often used to even out dynamics in a performance making loud portions softer and making soft portions louder When used more aggressively they can also add punch to a performance or to individual sounds and when used really aggressively can add overtones and hence change the timbre of a sound By combining a compressor with a side chain equalizer functions such as de essing reducing sibilant speech sounds and de ploding reducing plosive speech sounds can be done Gates are typically used to squelch background noises in noisy recordings Threshold and Ratio A compressor has five principal controls threshold ratio attack time release time and makeup gain The threshold is the input level at which compression will kick in The ratio parameter determines how much gain reduction will be applied as the input exceeds the threshold The definition is somewhat archaic and confusing The range of ratios is 1 to infinity A ratio of 1 means no gain reduction while a ratio of infinity means the gain reduction is equal to the amount the input exceeds the threshold hence the inp
118. t Simulating reverberation is essential for establishing the spatial context of an auditory scene Reverberation gives information about the size and character of the surrounding space it is very useful for correctly perceiving distances and it adds greatly to the realism of the simulation 8 2 10 Early Reflection Model The early reflections in a room can be modeled by considering the walls to be acoustic mirrors The soundfield produced by a source and a single wall reflection is equivalent to the soundfield created by the source and an image source located at the mirror image location behind the wall with no wall present Determining the early echo response for a room requires reflecting the source position across each wall boundary to obtain a set of first order image sources those that correspond to a single wall reflection The free path propagation from these image sources to the listener then determines the echo response Second order reflections can be modeled by reflecting the first order image sources across the wall surfaces to obtain second order image sources The figure below shows a rectangular room containing a source X and a listener O Some nearby first and second order image sources are also indicated i 1 i source i I i i listener Figure 8 4 Image sources in a rectangular room The dotted line from the source to the listener represents a reflected sound path which is equivalent to the free field
119. t active this is the normal dynamics mode Internal EQ means the EQ is inserted in the sidechain as described above This is the usual form of sidechain EQ found in studio gear and software The EQ can be one of the following types lowpass highpass bandpass and notch The remaining sidechain modes are Wave Arts innovations that further refine the sidechain equalizer capabilities Internal EQ Compare runs two sidechains the main sidechain uses the EQ type set by the user while the alternate sidechain uses the opposite EQ type So if the main sidechain is using a lowpass the alternate sidechain uses a highpass if main is using bandpass the alternate uses notch etc This way the main sidechain is detecting the in band frequencies while the alternate sidechain is detecting the out of band frequencies The result of the main sidechain detector is subtracted from the alternate sidechain detector so the dynamics processor responds to in band minus out of band levels The EQ Compare mode makes an exacting de esser When the main sidechain is set to a 5 kHz bandpass the alternate sidechain uses a 5 kHz notch The compressor will only kick in when the main sidechain has energy in the 5 kHz band and there is no energy outside of this band This prevents the compressor from triggering on voiced sounds that have energy in the 5 kHz band but also have energy elsewhere The EQ Invert mode simply negates the Compare mode the dynamics proces
120. t so the preset mix levels are a good place to start Beginning users of MasterVerb may be confused by the Wet Dry and Late Early controls The Wet Dry mix sets the wet mix level 100 is all wet 0 is all dry Similarly the Late Early mix sets the late mix level 100 is all late and 0 is all early Hence the controls give you the most wet effect when pushed to the maximum 100 value The confusion arises from the placement of the labels on the Ul moving the knob to 0 points the knob indicator to the Wet or Late label and moving the knob to 100 points the knob indicator to the Dry or Early label which is counter to the meaning Think of the controls as simply Late and Wet controls and remember that larger parameter values produce a wetter sound 5 3 User Interface In addition to the standard controls and displays found in other Wave Arts plug ins MasterVerb contains several custom displays which facilitate operation and allow you to visualize the inner workings of the plug in The displays are grouped together in the upper left of the MasterVerb window They include the character control the early and late damping controls the time response display and the frequency response display 58 5 MasterVerb Character control The character control allows you to simultaneously adjust the reverb time and the room size by clicking on the red ball and dragging it within the control area Moving the control left ri
121. t Pop Piano Warm Piano Disco String Section Horns Smoky Jazz Sax Sax with Attitude Gate presets Trackplug Gate presets are listed below Category Preset name General Default Typical Fast Gate Drums Tight Kick Looser Kick Snare 1 Snare 2 Tom Quick Decay Tom Long Decay Lo Tom Long Decay Room Mic triggered by Snare Compressor presets Trackplug Compressor presets are listed below Category Preset name General Default Transparent Gentle Compress Gentler Attack Brings Out Attacks Drums Kick Tightener Much Roomier Kick Snare After Ring Snappy Snare Softer Snare Attack Thunderous Tom Drum Room Mics Guitar A Gtr 1 Funk Guitar Hard Attack Slight Crunch Rhythm Guitar Harsh Guitar Tamer Bass Bass Evener Bass Add Punch Mid Range Bass Attack Forgot to change Bass Strings Clacky Bass 51 Wave Arts Power Suite 52 Percussive Upright Bass Vocals Out Front vocals Classic Warm Vocal Subtle Vocal Control Female Pop Vox Angry Rock Vocal Punishes Pesky Plosives De esser 1 De esser 2 Keyboard 80 s Pop Piano 5 MasterVerb 5 MasterVerb rm a Bypass Undo topy W E 4 P Preset Tools w NN CD wa0 at OUTPUT ms C 981 kHz 122 dB au 6 SIZE lt
122. t source per doubling of distance A value of 6 dB simulates natural acoustics but may seem exaggerated A value of 0 dB will cause the direct source to have the same gain regardless of distance Reflection Slope Sets the attenuation of the reflected sources per doubling of distance The gain of each reflected source is set depending on the distance of the reflected source from the listener i e the distance is the total air propagation distance from the source to the reflecting surface to the listener Reverb Slope Sets the attenuation of the reverb per doubling of distance The gain of the reverb is set based on the distance from the source to the listener Direct Filter Length Sets the filter length of the HRTF filters used to spatialize the direct source Using longer filters reproduces more fine spectral detail in the HRTFs using shorter filters smooths out the spectral details Longer filters are more accurate but more expensive computationally We recommend using at least length 16 filters for the direct source at 48 kHz At 96 kHz sampling rate you need length 32 filters to achieve the same sharpness Reflection Filter Length Sets the filter length of the HRTF filters used to spatialize all early reflections We recommend using shorter filters for the reflections than for the direct source for two reasons 1 the reflections are not as important to reproduce accurately and 2 there are up to six reflections so the reflec
123. t the Save item in the Preset menu described below The arrow controls to the left and right of the preset name cycle through the set of factory and user presets Clicking the right arrow goes to the next preset clicking the left arrow goes to the previous preset 3 6 Preset menu The Preset menu contains lists of factory and user presets for easy selection and options for managing presets The functions are described in the following sections res Tools MultiDynamics Defaut Save Hard Import Bass Comp Export vocals Reset Dance Mastering Acoustic Guitar Female De ess Male De ess Medium Enhance Mid Punch Bass Punch 2 Band Template 3 Band Template 4 Band Template Noise Gate Noise Reduce 3 7 Factory Presets Factory presets are selected from a rolloff menu at the top of the Preset menu Factory presets cannot be modified or deleted The Default preset is always first in the list it defines all default parameter settings For AAX DSP plug ins which have reduced configurations the same set of factory presets is presented for each configuration even though the 18 3 Menu Bar and Preset Manager configuration may not have all the features selected by the preset For example when loading a 6 band MultiDynamics preset into the 3 band configuration the first three bands will be set up while the top 3 bands from the preset are ignored Similarly when loading a
124. t the level of reverberation does not decrease much Thus the level of direct to reverberant sound can be used as a 55 Wave Arts Power Suite distance cue with dry non reverberant sounds perceived as being close and reverberant sounds perceived as being distant Reverberation gives information about the size and character of the surrounding space and the distance of the sound source For recordings that are made with close microphone techniques artificial reverberation must be added to restore spatial context Reverb is thus an essential tool for imparting a sense of space to your recordings Reverb algorithms Before digital reverbs were available recording studios had to use other means to generate reverberation One method required building a very reverberant room into which speakers and microphones were placed Sounds to be reverberated were piped into the room and the resulting reverb was picked up by the microphones These rooms were known as reverb chambers An alternative method of generating artificial reverb that requires much less space than a chamber is the plate reverb Essentially the reverb chamber is replaced by a suspended steel plate Signals are injected into the plate using transducers mounted on the plate causing the plate to vibrate The resulting vibrations are picked up by additional transducers mounted on the plate The reverberation created by a plate is quite dense and diffuse Although a plate s reverberat
125. t the peak limiter with no dither section This is because the dither section is computationally expensive so you can get more instances of just the peak limiter 6 2 About FinalPlug Peak Limiting FinalPlug s peak limiter can be used to do tasks ranging from light peak limiting to volume maximization to very heavy compression The FinalPlug limiter detects peaks in the signal and very quickly reduces the input gain prior to the peak preventing clipping After the peak has passed FinalPlug restores the input gain to the nominal level The peak limiter will limit any peak that exceeds the threshold level set by the user The peak limiter uses a built in lookahead delay to respond to peaks just before they occur Both the lookahead delay and attack time are fixed at 1 5 milliseconds This is designed according to the pre masking phenomenon of the human auditory system whereby loud sounds effectively mask sounds that occurred just prior Hence any distortion caused by the rapid gain ducking will be effectively masked by the onset of the peak The shape of the peak itself is unaffected by the gain ducking this is in contrast to peak limiters that use wave shaping techniques After the peak has passed the gain is restored to nominal level according to the release time set by the user The release time is an important parameter Short release times cause the gain to be increased rapidly after peaks This is important when you want to aggressively
126. tage Choir Loft Space for Metal Guitar Small Dark Church Concert Hall 65 Wave Arts Power Suite Small Hall Bloom Hall Opera Hall Drum plate Tame Plate Vocal Plate 80 s Big Hair Snare Tight Gated Toms Shotgun Snare Gated plate Reverse Plate In the Cave Ghostly Ending Gigantic Trailing into the Mist Pad Generation Echo verb Trippy Echoes Vox Splatter Delay Filtered SlapDelay MasterVerb also ships with MasterVerb v4 preset listed below Tiny Room Bright v4 Tiny Room Muffled v4 Studio v4 Studio bright v4 Studio warm v4 Nightclub v4 Nightclub bright v4 Nightclub warm v4 Concert hall v4 Concert hall bright v4 Concert hall warm v4 Cathedral v4 Cathedral bright v4 Outdoor ambience v4 Small resonant space v4 PA System v4 Scatter Verb v4 Thud Verb v4 Menace Verb v4 Large Concert v4 Live Hall v4 Grand Hall v4 Gymnasium v4 Small room v4 Bathroom v4 Oil Drum v4 Delay Verb v4 Dark Room v4 66 5 MasterVerb Church v4 Medium Room v4 Richochet v4 Slapback Echo v4 67 6 FinalPlug 6 FinalPlug AAA Bypass Undo Copy MPB lt 4 CD Mastering gt Preset Tools AN IN TRUNCATE DITHER 6 1 Overview FinalPlug is a lookahead peak limiter with bit depth trunca
127. te 4 3 User Interface 100 200 EQ The TrackPlug EQ interface is shown above At the top is the frequency response display which shows the frequency response of the current EQ The frequency response display contains a number of control handles drawn as colored balls which correspond to EQ bands clicking a handle will select that EQ band for editing dragging a handle will change the parameters for the EQ The band that is selected shows a halo around its control handle The table below summarizes the operations that can be done inside the frequency response display Adjust an EQ band s frequency and height Click drag Click drag Right click drag Adjust an EQ band s width Shift click drag or Shift click drag Add a new EQ band Double click Double click 36 4 TrackPlug Delete an EQ band Ctrl click Ctrl click Right click on display or Shift click on display Shift click on display Show popup menu of all EQ bands listing all parameters EQ Tabs Just below the frequency response display are tabs for selecting which EQ is to be displayed and edited Usually you will have the EQ tab selected which means the main TrackPlug EQ The other tabs are labeled G SC C1 SC and C2 SC for Gate sidechain Comp1 sidechain and Comp2 sidechain respectively Click on these tabs when you want to edit one of the dynamics sidechain EQs Note that the sidechain EQs are restricted to only
128. ther bands to be muted unless they are already soloed hence only soloed bands will pass audio Soloing a band is useful to hear the frequency range of the band and also the action of the dynamics in that one band Clicking the Mute 82 7 MultiDynamics button will enable disable muting of the current band Muting a band causes the band to become silent To clear the actions of the Bypass Mute and Solo buttons SHIFT click or on Windows right click the corresponding button SHIFT clicking Bypass causes all bypassed bands to return to un bypassed state SHIFT clicking Mute or Solo will undo all soloing and muting To quickly solo a single band SHIFT click or on Windows right click the band This will solo just this one band To clear the SOLO SHIFT click on the SOLO button Band parameter controls The six knobs below the frequency response display allow the user to change the parameters for the currently selected band These parameters are lo gain threshold hi gain ratio attack time and release time When the current band is selected the band parameter controls will update to reflect the values of the current band In addition to the above band parameters the knee control can be set on a per band basis All Bands controls Below the band parameter knobs are wheel controls that permit changing the band parameters for all bands simultaneously Click on the wheel and drag up and down This causes the parameter to change
129. tion and noise shaped dithering Here are some of FinalPlug s key features e Sonically transparent peak limiting for use as peak limiter or volume maximizer Auto release mode Bit depth truncation from 4 to 24 bits TPDF dithering Comprehensive noise shaping options Operation up to 192 kHz sampling rate Mono to mono or stereo to stereo operation Presets for CD and DVD audio mastering and track compression A diagram of FinalPlug s audio routing and meter placement is shown below Input Attenuation Output Meter Meter Meter Truncation Dither Out 69 Wave Arts Power Suite FinalPlug is most commonly used as the final plug in to process a mix on the master output It allows you to maximize the volume of a mix while preventing the sound from exceeding a certain level so that digital clipping doesn t occur It is more flexible than normalizing since you aren t simply increasing the volume of the entire track by a constant gain FinalPlug dynamically reduces peaks to be more in line with the rest of the signal and with careful use you can increase the perceived loudness of a mix by 6 to 12 dB without any audible artifacts If you are preparing a CD master FinalPlug s truncation and dithering features can be used to reduce the bit depth of your recording to 16 bits while minimizing the amount of quantization noise that can be perceived FinalPlug Limit For AAX DSP the FinalPlug Limit configuration runs jus
130. tion the shapes by selecting a bit depth of 8 bits which allows the quantization to be readily audible The 96 kHz shapes are intended for 96 kHz sampling rate and are flat below 20 kHz Hence if these shapes are used at 44 1 kHz sampling rate the result will be the same as the Flat shape The Mute control mutes the dither noise while continuing to truncate the bit depth of the audio This is useful for hearing the beneficial effects of adding dither noise Also if you want to use the quantization section as a low bit distortion effect then you need to enable the dither Mute Important for mastering if you are using the truncation dither section of FinalPlug make sure that any truncation dithering in your host application is disabled Similarly if you are using your host application to do truncation dithering and only using FinalPlug for peak limiting then disable the truncation dithering section of FinalPlug Very importantly make sure that FinalPlug is the last plug in that processes audio and that no processing of any sort happens after FinalPlug processing 6 4 Parameters The table below lists the parameters of FinalPlug Parameter name Values Limiter Enable O Off 1 On Threshold 36 0 dB Ceiling 36 0 dB Release 1 1000 ms Auto Release 0 Off 1 On Truncate Dither O Off 1 On Bit Depth 4 24 bits Dither Shape O Flat 1 44 1 kHz Shapel 2 44 1 kHz Shapel Ultra 3 44 1 kHz Shape2 4 44 1 kHz
131. tions can consume six times the CPU as the direct source when the filter lengths are the same Global Parameters TOP 3D E O c Mode Sets the output configuration mode Panorama can be optimized for playback over loudspeakers or headphones For playback over loudspeakers the output signal is processed with a crosstalk canceller circuit The crosstalk canceller has settings to optimize playback for different speaker angles where the angle is measured from the center point between the speakers to each speaker The possible output configurations are Headphones use for headphone playback regardless of headphone type Speakers 10 degrees use for playback over closely spaced speakers such as found on laptop computers This setting has the most exaggerated effect of the speaker settings 111 Wave Arts Power Suite Speakers 20 degrees use for playback over desktop computer speakers Speakers 30 degrees use for playback over standard stereo setups Speakers 40 degrees use for playback over stereo setups with widely spaced speakers This setting has the least exaggerated effect of the speaker settings Unfortunately the output modes are not entirely compatible with both headphones and loudspeakers Panorama is ideally applied when you know what the playback configuration will be What configuration should be used if you are not sure If Panorama is being used to create a special track effect e g a flying s
132. tor control for sidechain type and a monitor button If the monitor button is lit then the dynamics section outputs the sidechain signal This will usually be the input signal processed through the sidechain EQ however if the sidechain type is off the monitor will simply pass the input signal effectively bypassing the dynamics section If an external sidechain source is selected the monitor will pass the external source This is a good way to verify the proper external source is being used Output section TrackPlug s output section user interface is shown below OUTPUT The stereo output meters have peak indicators above the meters If the value exceeds O dB the display font turns red Clicking on either indicator resets both channels to 96 dB Right clicking Windows or shift clicking Mac on either value will automatically normalize the output gain to give a peak of 0 1 dB Below the meters is the output gain control and below this is the peak limiter control The peak limiter is enabled when the Limiter button is lit When enabled peaks that exceed 0 1 dB are automatically limited The display below the limiter 43 Wave Arts Power Suite shows the peak signal level before limiting this will be displayed in red text if it is above O dB Click on the display to reset it to 96 dB 4 4 Parameters This section lists all the internal parameters of TrackPlug and shows the range of values as would be displayed by a generic
133. u GAIN N A FREQ GAIN a x152 J x8 68 5 1 Overview MasterVerb is a classic digital reverb with a spacious natural sound and a silky smooth decay Featuring multiple reverb algorithms with comprehensive early reflection equalization and envelope control MasterVerb is capable of reproducing a vast palette of reverberation effects MasterVerb features the following e Choose between hall and plate late reverb algorithms e Comprehensive early reflection selections including halls rooms chambers plates or special effect reflections e Independent room size and decay time parameters allow simulation of a variety of spaces from bathrooms to cathedrals e Three band early and late equalizers allow detailed tone shaping e Envelope generator for creating gate and reverse effects e Character control lets you find the room sound you want in just seconds 53 Wave Arts Power Suite e Time response display shows relative levels of direct sound early reflections and late reverb e Frequency response display shows the reverb decay in three dimensions e Extremely CPU efficient algorithms A diagram of MasterVerb s audio routing and meter placement is shown below Env Output Meter Meter Out Early EQ Reflections gt 4 Late a gt MasterVerb audio routing diagram The input signal is split into wet and dry paths The wet path is delayed by the wet delay also known as pre delay and is then proces
134. uration provides a 7 band EQ gate and compressor The audio routing is shown in the figure below 24 4 TrackPlug Gate Comp1 Output Meter Meter Meter TrackPlug E7GC audio routing diagram TrackPlug E7C Configuration For AAX DSP the E7C configuration provides a 7 band EQ and compressor The audio routing is shown in the figure below Comp1 Output Meter Meter Compi pre TrackPlug E7C audio routing diagram 4 2 About TrackPlug Brickwall filters Brickwall filters are lowpass or highpass filters with very steep cutoffs used to pass all frequencies up to the cutoff frequency and eliminate all frequencies beyond the cutoff The frequency range that is passed unaltered is called the passband the frequency range that is attenuated is called the stopband TrackPlug s brickwall filters are implemented using 10th order elliptical filters with at least 90 dB of stopband attenuation and less than 0 1 dB of passband ripple Brickwall filters are used to eliminate unwanted frequency ranges Typically a brickwall filter would be used when processing a noisy recording of an instrument sound that does not use the entire frequency range The brickwall filter would be positioned at the edge of the instrument s frequency range to eliminate out of band noise So for example when processing voice one could use the brickwall filters to eliminate all frequencies below 100 Hz and above 8 kHz The brickwall filters can also be
135. us the Limiter Because most users do not use all these features it makes sense to provide feature reduced configurations that require less CPU and memory and thus can run more instances on HDX DSP The various configurations of the three plug ins are summarized in the table below Plug in Configuration Notes TrackPlug MultiDynamics 4bands 4 bands FinalPlug TrackPlug comes with two feature reduced configurations TrackPlug E7C provides 7 bands of EQ plus a single compressor with optional sidechain EQ The TrackPlug E7GC configuration provides 7 bands of EQ a gate expander and a single compressor Choosing 7 bands of EQ is motivated by the Avid D control and D command interfaces that have dedicated control layouts for 7 bands of EQ Wave Arts Power Suite MultiDynamics comes with two feature reduced configurations consisting of 3 band and 4 band variations as compared to the full implementation of 6 bands FinalPlug comes with a limiter only configuration This is because the dither section consumes considerably more CPU than the limiter but most users want only the limiter When running the reduced configurations the same graphical user interface is shown but certain user interface controls will be disabled if they control features not supported in the current configuration The factory preset list is tailored to each specific configuration User presets via Wave Arts preset manager or Pro Tools preset manager ca
136. used to zero in on a particular frequency range just for analysis purposes For example one could use the brickwall filters to listen to selected overtones in an organ or to isolate the click of a kick drum 25 Wave Arts Power Suite Equalization The equalizer is used to boost certain frequencies and reduce others thus changing the tonality of a sound When used in a mixing application equalization is applied to each track so that it sits better in the mix Generally you want each instrument sound to be distinctly audible and sound natural while maintaining a balanced mix A good approach is to use as little EQ as possible and when doing so try to reduce frequencies rather than boost them to give each track enough space in the mix When equalization is used as a sound design tool it can be applied aggressively to completely change the character of a sound It is helpful to have a general understanding of the frequency ranges in a mix Low Bass 20 60 Hz controls rumble this range is best reproduced on a subwoofer equipped system Bass 60 to 250 Hz the low end of the mix where the fundamentals of bass and other rhythm oriented instruments like the kick drum reside Controls the overall fullness roundness of a track or mix Reducing around the 100 to 200 Hz range can help reduce boominess Low Midrange 250 to 2000 Hz many of the low harmonics of most instruments are in this range and some boost here between 250
137. ut will be pinned at the threshold value In general for a ratio R the gain reduction is R 1 R of the amount the input exceeds the threshold So to take a typical example if the input is 12 dB over threshold with a ratio of 3 the gain reduction will be 8 dB But compressors are not really about decreasing gain they are really about increasing gain What they do is decrease the peaks in a signal so the rest of the signal can be boosted This is what the makeup gain control does it applies a constant gain boost to increase the soft parts of a signal while the compressor dynamically pushes down the peaks Attack and Release Time The attack time and release time parameters are really important The attack time controls how fast the gain is turned down when gain reduction is to be applied and 31 Wave Arts Power Suite the release time controls how fast the gain is turned back up Consider the example of compressing a drum hit With a small fast attack time the initial transient of the drum hit will be compressed because as soon as the transient exceeds the threshold the gain reduction kicks in immediately As you increase the attack time it takes longer for the gain reduction to fade in and the initial transient of the drum hit passes through uncompressed So in this case when you want to hear more drum attack you increase the attack time After the transient has passed and the drum sound begins its decay the compressor will begin re
138. veform the object has moved closer to the listener and thus the emitted wavelength is shorter than normal Similarly when the object is retreating from the listener the pitch is lower than the resting pitch because the emitted wavelengths are longer than normal Simulating the Doppler effect is important for generating realistic motion effects Panorama simulates the Doppler motion effect using a variable delay line The amount of delay is proportional to the distance between the listener and the sound object Thus the delay line effectively simulates the propagation of sound through the air When the distance changes so does the length of the delay and the pitch also changes as it would in nature Unlike nature Panorama provides the feature to turn off the Doppler effect When the Doppler effect is turned off the amplitude changes as a function of distance but the pitch stays constant 8 2 14 Panorama Audio Routing Following is a diagram of Panorama s audio routing 101 Wave Arts Power Suite Output Slope Material Distance Position HRTF Spkr Hdph Meter Distance Filter Length Configuration Out Crosstalk Doppler Canceller Delay Output 2 Gain A A A AA A Direct Virtual Source Virtual Source Virtual Source Reverb Reverb Slope Reverb Distance Parameters Figure 8 6 Panorama audio routing block diagram Panorama uses the concept of a virtual source a m
139. w wavearts com Contact information Wave Arts Inc 99 Massachusetts Ave Arlington MA 02474 tel 1 781 646 3794 866 925 5070 fax 1 781 646 7190 117 Index A ALB DUES odian diran amen 17 DAX DSP anar aN earthen 8 ADO a ede shee 20 Attack time scce 31 57 70 87 Auto release cococcccccccnconccnncnnnnancnnnnnnn 71 B Bandpass filter l a 29 Binaural synthesis 93 Brickwall filter a se 25 BUTEONS criai ia a RA 15 BYPASS iio 17 C e aTe E E A TE 73 Character iria 59 Clean Mode oocccccccconconcncanccnncannnn nano 33 Compression coocccccncccccnnnnnnnannos 31 80 84 Coordinates ococccccccncnncncnncconnnnnnnannos 107 COPY ratita E Ga 17 Crosstalk cial al 95 D DAMPING riea ea ra cheers sacs 59 62 63 DIFUSION aaa 65 Distance iaa arpa ar an s 99 Dither Shapes oooccccccccoccnccnnnncnnnancnnnos 76 DItheriN ri kes cea 71 Doppler coooccccccnccnccnncnnnnnnos 101 102 113 Dry delay coccion a 64 DYNAMICS sia a 31 Dynamics response c coccoccnccncconcnnnnncnnos 42 E Envelope ccceceeeeeeeeeeeeeeeeaees 57 64 EQ oeei vie ncesan ana E iY 26 Equalization oooooccccccccnccncconcnncncnncnnnos 59 EXPANSION cerei p E iA 80 84 Export preset ceeeeeeeee ee eeeeeee eee 20 F Factory preses estian a 18 FinalPlU0d aia aian Eia EEE 69 Frequency reSponse coocccccccconnncccannnnnos 60 G E e EE APEE ated EEE EE EEE 32 Index H AROA ctrl teh ed 84 86 Highpass filter cece ee
140. y for a kick drum for example or if you want to prevent any harmonic distortion during constant level tonal sections When we refer to aggressive compression settings we primarily mean using fast attack and release times say 0 1 to 5 msec which are fast enough to cause the compressor to reattack with every period of the incoming signal This in turn causes waveform shaping and the production of harmonic overtones When using long attack and release times the gain changes are spread over many periods of the signal and distortion is substantially reduced Using large ratios and hard knees also contributes to aggressive compression The vintage modes work like traditional compressors If you select aggressive compression settings the vintage modes will distort The vintage modes include Vintage Peak Vintage RMS and Vintage Warm The Vintage Peak mode detects the peak absolute value of the input signal the resulting compression will affect the positive and negative swings of the signal equally and will thus create odd harmonic overtones The Vintage RMS mode computes the squared input signal and runs this through an averaging envelope follower Because of the relatively slow averaging used to obtain the RMS level compared to the very fast peak level detection using fast attack and release times will produce less distortion with Vintage RMS mode than with Vintage Peak mode However like Vintage Peak mode Vintage RMS mode also produces only odd
141. zation Typically the EQ is set up to attenuate high frequencies because reverberation is usually not very bright due to the absorption of high frequencies by materials in the room The damping EQ is a three band shelf equalizer with lo mid and high gains and adjustable band edge frequencies Early Damping Lo Freq Sets the transition frequency between the low band and mid band of the early damping EQ Early Damping Lo Gain Sets the gain in dB of the low band of the early damping EQ Early Damping Mid Gain Sets the gain in dB of the mid band of the early damping EQ 62 5 MasterVerb Early Damping Hi Freq Sets the transition frequency between the mid band and high band of the early damping EQ Early Damping Hi Gain Sets the gain in dB of the high band of the early damping EQ Late Reverb Type Selects the late reverb algorithm There are currently two completely independent reverberation algorithms the Hall algorithm and the Plate algorithm The hall algorithm sounds like a concert hall which can be scaled to sound like small rooms or very large spaces The plate algorithm sounds like a plate reverberator An actual plate reverb is created by attaching transducers to a large steel plate suspended from wires the transducers send the dry sound into the plate which vibrates and the resulting reverb is picked up by other transducers Plate reverbs have extremely dense and diffuse reverberation Reverb Time Sets t

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