Wave Arts Power Suite 5

User Manual
Last updated: March 7, 2007 Copyright 2007, Wave Arts, Inc. All Rights Reserved

Table of Contents

1. Installation and Registration...5 1.1 Mac OS X Installation..5 1.2 Windows Installation...6 1.3 Registration...7 1.4 Registration Troubleshooting.. 10 1.5 Video Tutorials... 11 2. Plug-in Control Operation... 13 2.1 Knobs.... 13 2.2 Text Entry... 13 2.3 Selector button... 14 2.4 Sliders... 15 2.5 Buttons... 15 2.6 Output Meters... 15 3. Menu Bar and Preset Manager... 17 3.1 Bypass... 17 3.2 Undo... 17 3.3 Copy... 17 3.4 A/B buffers... 17 3.5 Preset name and arrow controls... 18 3.6 Preset menu... 18 3.7 Factory Presets... 18 3.8 User Presets... 19 3.9 Save... 19 3.10 Import.... 19 3.11 Export.... 19 3.12 Reset... 20 3.13 Tools menu.... 20 3.14 About... 20 3.15 Open User Manual... 21 3.16 Unlock Plug-in... 21 3.17 Check for Updates... 22 3.18 Visit Website... 22 4. TrackPlug... 23 4.1 Overview.... 23 4.2 About TrackPlug... 25 4.3 User Interface... 35 4.4 Parameters... 43 4.5 Parameter Descriptions... 44 4.6 Specifications... 47 4.7 Presets... 47 5. MasterVerb... 53 5.1 Overview.... 53 5.2 About MasterVerb... 54 5.3 User Interface... 58 5.4 Parameters... 61

Wave Arts PowerSuite

5.5 Parameter Descriptions... 62 5.6 Specifications... 65 5.7 Presets... 65 6. FinalPlug.... 69 6.1 Overview.... 69 6.2 About FinalPlug... 70 6.3 User Interface... 72 6.4 Parameters... 74 6.5 Parameter Descriptions... 75 6.6 Specifications... 77 6.7 Presets... 77 7. MultiDynamics... 79 7.1 Overview.... 79 7.2 About Multiband Dynamics... 80 7.3 User Interface... 81 7.4 Parameters... 85 7.5 Parameter Descriptions... 85 7.6 Specifications... 88 7.7 Presets... 88 8. Panorama... 91 8.1 Overview.... 91 8.2 About 3D Audio and Acoustic Environment Modeling.. 92 8.3 User Interface... 103 8.4 Parameters.... 105 8.5 Parameter Descriptions... 107 8.6 Specifications... 113 8.7 Presets... 113 License Agreement... 115 Support... 117 Index... 119
1. Installation and Registration
1.1 Mac OS X Installation
If you are Installing from the Power Suite CD-ROM, insert the CD-ROM and double click the Power Suite OSX folder. You will see the Power Suite OSX.mpkg file as shown below. If you are installing from a download, the download manager will create a file Power Suite OSX.dmg. Double click on this file to create the file Power Suite OSX.mpkg. If you are downloading and installing an individual plug-in or another bundle, the procedure is similar, but the files will be named after the plug-in or bundle rather than named Power Suite.
Now double click the Power Suite OSX.mpkg icon, and proceed though the installation process:

Wave Arts Power Suite

1.2 Windows Installation
If you are installing from a CD-ROM, insert the CD-ROM and double-click the file PowerSuite5.exe shown below. This will start the installation process as described below. If you are downloading Power Suite, the file PowerSuite5.exe will be created on your computer. Double-click this file to start the installation process. If you are downloading and installing an individual plug-in or another bundle, the installation process is similar, but the files will be named after the individual plug-in or bundle rather than named Power Suite.
The installer will guide you through the following screens:
1. Welcome. Click Next to continue. 2. Read Me File. Displays information about demo and registration procedure. Click Next to continue. 3. License Agreement. Read and click Yes if you agree to the terms of the agreement. The agreement is reprinted at the end of this user guide. 4. Select Components. Here you have the option to select DX, VST, and RTAS formats for installation. Click each box to toggle the checkmark on/off; boxes with checkmarks indicate the corresponding component (i.e., plug-in format) will be installed. By default, all three boxes are checked and thus all three formats will be installed. Click Next to continue. 5. Choose Destination Location. For each of the plug-in formats selected to install, choose the destination location on your file system. The first location is for DX format plugs, and also serves as the destination for the user manual. The second location is the folder where VST plug-ins are installed. The third location is the folder where RTAS plug-ins are installed. Click Next to continue. 6. Select Program Manager Group. Enter the name of the folder in the Windows start All Programs menu where the Wave Arts files can be accessed. 7. Start Installation. Click next to perform installation. 8. Installation complete. Click Finish to exit the installer. If you are attempting to install plug-ins that are currently in use, you may be asked to restart your computer.

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 <plugin> Presets.txt, where <plugin> is the name of the plug-in you are using. The file is located in the same directory where the plug-in was installed. If the file is deleted it will be created automatically by the plug-in.

3.9 Save

When you have created an effect you want to save as a preset, 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 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 plugin 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 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.

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 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 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 0dB. 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. 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 sidechain 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. 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.

4.6 Specifications

Description Platforms Operating Systems Plug-in Formats Sampling Rates Precision I/O Formats Channel strip with 10-band EQ, RTA, 2 compressors, gate, brickwall filter, and peak limiter. Pentium & compatible; PowerPC Windows 2000/XP; Mac OS X 10.4 or higher AU, DirectX, MAS, RTAS, VST up to 192 kHz 64-bit EQ, 32-bit dynamics mono-mono, mono-stereo, stereo-stereo

4.7 Presets

TrackPlug global presets are listed below. The presets are descriptively named.
Category General Drums Preset name 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 Channel Walkin' Jazz Bass Piano Pop Jazz Piano Ballad Sax Solo Vocal Showstopper Warm Rap Control Angry Rapper

Guitar

Bass Keyboard Horns Vocals
Wave Arts Power Suite Songstress in my Ear Sexy Pop Diva Female R+B Ballad Effected Vox 1 Very Nice 2-Mix Compress Aggressive Rock 2-Mix
TrackPlug also ships with TrackPlug v4 presets, listed below:
Category Drums Preset name Drums v4 Drums - Squisher v4 Drums - Squasher v4 Drums - Tighter v4 Kick v4 Snare 2 v4 Snare 3 v4 Toms v4 Hi-hats - Cymbals v4 Acoustic Guitar v4 Acoustic Gtr 2 Bass v4 Bass - Acoustic v4 Bass - Attack v4 Bass - Attackier v4 Piano v4 Horns v4 Sax v4 Vocal - Female v4 Vocal - Male v4 Voiceover - Female v4 Voiceover - Male v4 Mix - Dance v4 Mix - Rock v4 Master Plug vBand EQ v4 The Works v4 Xtreme Beat v4 Xtreme Vocals v4 What Dynamics? v4 Noise Filter v4 Telephone v4 Telephone - Screaming v4

Guitar Bass

Keyboard Horns Vocals

Special

EQ presets Trackplug EQ presets are listed below.
Category General Preset name 4-band Default
4. TrackPlug Brighter Warmth Biting Mid-range Fat-Bottomed Pipeline Xylophonated Comb the Area!! Telephone Transmission Old Time Radio Show Hum 60Hz Hum 50Hz NTSC PAL-SECAM 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 Acoustic Gtr 1 Acoustic Gtr 2 Nu-Metal Guitar Typical Death Metal Filtered Funk Guitar Ukelelephone Modern Rock Bass Gave my Bass Wood Deep Bass Bass DI Helper Snappy Upright BAss Male Vox 1 Female Pop Vox

A diagram of MasterVerbs audio routing and meter placement is shown below.

Env Meter Output Meter

Wet Delay
Early Early EQ Reflections Late Envelope

Output Out

Late Reverb Dry Delay
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 processed 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.
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 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 amplified 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, but the level of reverberation does not decrease much. Thus, the level of direct to reverberant sound can be used as a

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.

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-right 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
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.

Time response display

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. 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
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.

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, 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 limiting. 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 DVDaudio 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 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

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 display 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 bands parameters can be edited using the knobs below the band display. The active bands 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 other 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 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 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 0 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. 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

dB 0 -6 -12 -18 -24 -30 1

direct gain reverb gain

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.
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 waveform 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 Panoramas audio routing:

Slope, Distance

Material

Distance

Position, HRTF, Filter Length

Spkr/Hdph Configuration

Output Meter

Direct

Material Filters Virtual Source

Doppler Delay

3-D Filters

Crosstalk Canceller

Out Output Gain 2

R 1 Mono Mix

Left Right Top Reflections Bottom Front Back

Virtual Source 2

Virtual Source Virtual Source Virtual Source Virtual Source Virtual Source Virtual Source

Reverb

Reverb R
Reverb Slope, Distance Reverb Parameters
Figure 8.6. Panorama audio routing block diagram.
Panorama uses the concept of a virtual source, a monophonic 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.

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 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

Panoramas user interface is shown below.
Overhead View The overhead view is selected by clicking 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 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
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 +180 degrees -40 to +90 degrees 0 - 20, feet or meters -20 to +20, feet or meters -20 to +20, feet or meters -20 to +20, feet or meters 0 = Polar, 1 = Cartesian 0 = Off, 1 = On 0 = Off, 1 = On 0 = Off, 1 = On -18 to +12 dB -18 to +12 dB -18 to +12 dB -12 to 0 dB, per distance doubling -12 to 0 dB, per distance doubling -12 to 0 dB, per distance doubling 1 - - = Off, 1 = On 0 = Off, 1 = On 0 = Off, 1 = On 0 = Off, 1 = On 0 = Off, 1 = On 0 = 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 0 - 17, see table 0 - 17, see table 0 - 17, see table 0 - 17, see table 0 - 17, see table

monophonic virtual source. When processing mono sources, the width parameter is fixed at Mono. Reflection Parameters
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 reflection. 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.
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 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 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

A A/B buffers..17 About..20 Attack time.. 31, 57, 70, 87 Auto-release..71 B Bandpass filter..29 Binaural synthesis..93 Brickwall filter..25 Buttons..15 Bypass..17 C Ceiling...73 Character...59 Clean mode..33 Compression.31, 80, 84 Coordinates. 107 Copy..17 Crosstalk..95 D Damping..59, 62, 63 Demo mode.. 7 Diffusion..65 Distance..99 Dither shapes.76 Dithering..71 Doppler..101, 102, 113 Dry delay..64 Dynamics...31 Dynamics response..42 E Envelope. 57, 64 EQ...25 Equalization..59 Expansion.. 80, 84 Export preset.20 F Factory presets..18 FinalPlug..69 Frequency response..60 G Gate..32 H Hi gain.. 84, 86 Highpass filter.. 25, 30 Hold time... 57 HRTF.. 92, 95, 112 I Import preset.. 19 Installation.. 5, 6 K Key code.. 9 Knee..46, 85, 87 Knobs... 13 L Late/early mix.. Lo gain.. 84, Lookahead.. 46, 70, 83, Lowpass filter.. 25, M Mac OS-X.. 5 Machine ID... 7 Makeup gain.. 31 MasterVerb.. 53 Materials...106 Menu bar... 17 Meters.. 15, 41 Mode.. 33, 34 Multiband dynamics.. 80 MultiDynamics.. 79 N Noise shaping.. 71 Notch filter.. 28 O Output configuration..111 Overhead view..104 P Parametric EQ.. Peak limiter..35, 43, Peak mode.. Plate.. Plate reverb.. Pre/Post.. 85 30
Wave Arts Power Suite Pre-delay..64 Preset menu..18 Preset name..18 Presets..18 R Ratio..31, 84, 86 Real Time Analyzer (RTA). 30, 38 Reflection type..62 Reflections..55, 56, 98, 103 Registration.. 7 Release time.. 31, 57, 70, 87 Reverb algorithm.. 56, 63 Reverb decay time.. 55, 63 Reverberation. 54, 97, 103, 109 RMS mode..32 S Sampling rate.. 113 Save preset..19 Selector button..14 Serial number. 7 Shelf filter...27 Sidechain. 34, 43 Size.. 62, 63 Sliders...15 Slope.. 100 T Tabs.. 37, Text entry.. Threshold..31, 57, 73, 84, Time response.. Tools menu.. TrackPlug... Truncation.. U Undo.. 17 Unlock.. 7 User manual.. 21 User presets.. 19 V Video tutorials.. 11 Vintage mode.. 33 W Wet delay.. 64 Wet/dry mix.. 64 Width..107 Windows.. 23 71

1.4 Registration

After your plug-ins are installed, they will function in demo mode for 30 days. While in demo mode they will be fully functional, however you must register (i.e., unlock) your plug-ins or they will stop working after 30 days. You will be warned when you have 7 days remaining in your demo period. To unlock the plugs after installation, go to http://wavearts.com/register.html and enter the required information. Youll need your Serial Number (either emailed to you or printed on a page in the product box), and your Machine ID, which can be determined by selecting the About command in the File menu of any Wave Arts plug-in.
Youll see an About dialog box like the one below. The About box shows the version number of the plug-in, the host plug-in format (e.g., DirectX, VST), the registration status of the plug-in, and the machine ID of your computer. Make note of the Machine ID:
At our registration site (http://wavearts.com/register.html) you will enter your PowerSuite Serial Number (the number that begins with WA-PS) and your Machine ID. If you purchased an electronic download, your Serial Number was emailed to you when you purchased the product. If you purchased the software in a shrink-wrapped box, the Serial Number can be found in the box contents. The registration page has the following fields to fill out:
The registration page needs the following information: your name, your email address, your Serial Number, and your Machine ID. Its very important to provide a working email address because your Key Code will be emailed to this address. Also be sure to enter the Serial Number and Machine ID correctly.
Wave Arts Power Suite Once the required information is entered at the Register web page you will be emailed a reply containing your 10-digit Key Code. An example reply is shown below:
Here is the key code for your Wave Arts product with Machine ID 18869: PowerSuite: 111-222-3333 To activate your software, go to the Register menu item of a Wave Arts plug-in and enter this number. Please contact us if you have any problems. Thank you! Wave Arts, Inc. You then unlock the plug-ins be selecting the Register command in the File menu of any Wave Arts plug-in:

Then enter the 10-digit key code in the dialog box:
You should see a dialog indicating that the registration was successful. If you need to install the plug-ins on another machine, such as a laptop, you may repeat the procedure above. Our license policy permits up to three installations provided you are the only user. The registration web page keeps track of the number of machines on which you have installed your plugs.
1.5 Registration Troubleshooting
Invalid key code message make sure you are entering the 10-digit key code from the registration email, NOT the 14-digit serial number. Website says my number of registrations is used up - contact Wave Arts support (support@wavearts.com) and let us know the number of computer(s) the plug-ins are installed on. As a single user you can install the plug-ins on up to three computers, but additional users must purchase their own copies of the plug-ins. Key code email not received - make sure the mail is not being filtered by a junk or SPAM mail program.
2. Plug-in Control Operation

2.1 Knobs

Please refer to the following guide for information about the various ways you can use knobs: Function Increase/Decrease a parameter value (rotate clockwise/counterclockwise) Mac Click on the knob + drag up/down Shift + click + drag up/down Fine adjustment increase/decrease RTAS: Command + click Immediately jump to a value Click on the ticks around the knob Command + click -orDouble-click RTAS: Option + click Windows Click on the knob + drag up/down Right click + drag up/down -orShift + click + drag up/down RTAS: Ctrl + click Click on the ticks around the knob Control + click -orDouble-click RTAS: Alt + click
Reset knob to default value

2.2 Text Entry

All value displays are editable text. Following is a table that fully describes its functionality: Function Enter text entry mode Select text Select entire text Delete character to left of cursor Delete character to right of cursor Move the cursor left/right Extend the current selection Mac Click in the display Click + drag Double-click Delete Del Shift + click + drag Windows Click in the display Click + drag Double-click Backspace Delete Shift + click + drag

Graph handles are the small colored balls that sit on the EQ and dynamics curves in the graphs. They are the quickest, most direct, and most versatile controls in the TrackPlug interface. Simply click-and-drag a handle around its graph to change corresponding parameter value(s) Following are tables of behaviors for the various TrackPlug graph handles, for both Mac and Windows platforms. EQ Section Control Adjust an EQ band's frequency and height Adjust an EQ band's width Mac Click + drag Shift + click + drag Windows Click + drag Right click + drag -orShift + click + drag
Add a new EQ band Delete an EQ band Compressor (Orange Handles) - AMG Enabled Control Adjust threshold and ratio Adjust output level Adjust threshold
Double-click Ctrl + click
Mac or Windows Click leftmost handle + drag Click rightmost handle + drag up/down Click orange triangle above input meter + drag left/right
Compressor (Orange Handles) - AMG Disabled Control Adjust threshold and output level Adjust ratio Adjust threshold Mac or Windows Click leftmost handle + drag Click rightmost handle + drag up/down Click orange triangle above input meter + drag left/right
Gate (Blue Handles) Control Adjust threshold Adjust ratio Mac or Windows Click topmost handle + drag left/right -orClick blue triangle above input meter + drag left/right Click bottommost handle + drag left/right

3.4 Parameters

Band parameters
Each of TrackPlug's 10 bands have the following parameters: Parameter name Band Enable Type Values 0 = Off, 1 = Bypass, 2 = On 0 = Parameteric, 1 = low shelf, 2 = high shelf, 3 = lowpass, 4 = bandpass, 5 = highpass, 6 = notch Hz -24db - +24db 0.01 - 4.0 octaves

Frequency Height Width

Global parameters
Parameter name Compressor Enable Compressor AMG Compressor Thresh Compressor Ratio Compressor Knee Compressor Att Compressor Rel Compressor Gain Gate Enable Gate Thresh Gate Ratio Gate Knee Gate Att Gate Rel Peak Detect Routing Lookahead Delay Input Gain Values 0 = Off, 1 = On 0 = Off, 1 = On -96 to 0 dB 1:1 to 20:= Soft, 1 = Medium, 2 = Hard 1 to 1000 msec 1 to 5000 msec -24 to +24 dB 0 = Off, 1 = On -96 to 0 dB 1:1 to 20:= Soft, 1 = Medium, 2 = Hard 1 to 1000 msec 1 to 2000 msec 0 = RMS, 1 = Peak 0 = Pre, 1 = Post 0 = Off, 1 = 1 msec, 2 = 2 msec, 3 = 5 msec -96 to 96 dB
3.5 Parameter Descriptions
The EQ allows up to 10 bands, each of which may be parametric, low shelf, high shelf, lowpass, bandpass, highpass, or notch filter. This section will describe all of the EQ control parameters. The following EQ control buttons are located directly below the EQ graph. From left to right, they are: EQ Enable, Current Band, Add, Delete and Band Enable.

The following controls are used to set the gate's parameters. The two buttons on top are: Gate Enable and Knee. The knobs below are: Threshold, Ratio, Attack, and Release.
Gate Enable Turns the gate section on and off. Knee Sets the gate's shaping characteristic; options are soft, medium and hard. Threshold Level below which the gate is active. Range is -96dB o 0dB. Ratio Amount of gate applied. Range is 1:1 to 1:20. Attack Time it takes the gate to deactivate once the signal crosses above the threshold. Range is 1ms to 1000ms. Release Time it takes the gate to activate once the signal falls below the threshold. Range 1ms to 2000ms.
3. TrackPlug Dynamics Shared The following controls are shared by both the compressor and the gate: Dynamics Input Gain, Detector Type, and Lookahead Delay. Detector Type The gate and compressor share the same detector. You can toggle between RMS and peak detection. Lookahead Delay TrackPlug's dynamics section has on optional lookahead delay that will enable it to respond quicker to attacks. Possible values are: Off, 1 msec, 2 msec, and 5 msec. Dynamics Input Gain An input gain for the signal coming into the Compressor/Gate. Range is -96dB to 24dB.

3.6 Specifications

Description Platforms Operating Systems Plug-in Formats Sampling Rates Precision I/O Formats All-in-one 10-band EQ, Compressor and Gate Pentium & compatible; PowerPC Windows 2000/XP; Mac OS 8.5 or higher; Mac OS X 10.2 or higher AU, DirectX, MAS, RTAS, VST up to 96 kHz 64-bit EQ, 32-bit Compressor/Gate mono-mono, mono-stereo, stereo-stereo

3.5 Presets

Following is the list of presets that ship with TrackPlug: 1. Default 2. Bass 3. Bass - Acoustic 4. Bass - Attack 5. Bass - Attackier 6. Drums 7. Drums - Squisher 8. Drums - Squasher 9. Drums - Tighter 10. Hi-hats/Cymbals 11. Kick 12. Snare 1 13. Snare 2 14. Toms 15. Vocal - Female 16. Vocal - Male 25. 26. 27. 28. 29. 30. 31. 32. 33. 34. 35. 36. 37. 38. 39. 40. MasterPlug 10 Band EQ The Works Xtreme Beat Xtreme Vocals What Dynamics? C C#/Db D D#/Eb E F F#/Gb G G#/Ab A

Wave Arts Power Suite 17. 18. 19. 20. 21. 22. 23. 24. Voiceover - Female Voiceover - Male Acoustic Guitar Piano Horns Sax Mix - Dance Mix - Rock 41. 42. 43. 44. 45. 46. 47. A#/Bb B Telephone Telephone - Screaming Noise Filter 50 Hz Hum Remover 60 Hz Hum Remover

4. MasterVerb

4.1 Overview
Spacious and lush, MasterVerb is a true stereo reverb with a natural sound and a silkysmooth decay. Independent room size and decay time parameters allow simulation of a variety of spaces from studio rooms to warm concert halls to bright cathedrals. MasterVerb features a responsive and informative 3-D plot of your reverb's characteristics, updated in real-time as you tweak control knobs for decay time, room size, diffusion, early damping, late damping, and low frequency scale. An added 2-D control helps you set your desired room response in just seconds. MasterVerbs routing diagram and meter placement is shown below:

4.2 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. 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 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 reflected energy. 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

Wave Arts Power Suite 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. 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 (nonreverberant) 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. 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 youd want the reverb mix to be 100%, since you will be hearing the track dry already, and the reverb signal on the aux return. 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. If you use reverb as an insert effect, you may want to decrease the wet/dry mix to taste so that the reverb effect is not too muddy.

5. 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 threshold 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 the 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 3d-order or 5th-order Butterworth bandpass filters. The crossover slope choices are -18 dB/oct (for the 3d-order filters) or -30 dB/oct (for the 5th-order filters).

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 so that 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 slowly restored to nominal level according to the release time set by the user. The nominal gain is set so that the maximum output level is determined by the ceiling parameter set by the user. In other words, any input signal that exceeds the threshold will be output at the ceiling level.

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, 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" 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.
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, 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.

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 limiting. 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

In addition to the standard buttons, sliders, and meters, whose operation is described in the section Plug-in Control Operation, the following controls are specific to FinalPlug.

Slider Meters

There are two sliders which allow you to set signal threshold and maximum ceiling. Click on one of the arrows (or anywhere in between) to move the slider up and down. You can also enter text directly into the bottom text field. The meters show the peak signal power in short time updates from -36dB to 0dB. The meter also stores the overall peak value for each channel, and displays these values in the peak indicator boxes to the right of the meter. If the detector finds a peak value above 0dB, the indicator will turn red as a warning. Click on either indicator box to reset them.

Dither Controls

Two menus allow you to set the bit depth and dither shape; select values by clicking on the drop down menu button and selecting the desired value from the popup menu. Clicking on the bit depth field allows you to edit the text and enter any value between 4 and 24. Clicking on the shape text field will cycle to the next shape, holding SHIFT will cycle to the previous shape. The plot shows the response of the noise shape filter. The horizontal axis is frequency in Hz, the vertical axis is amplitude in dB. Noise is reduced where the plot is below 0 dB and increased where the plot is above 0 dB.

6.4 Parameters

Parameter name Limiter Enable Threshold Ceiling Release Truncate/Dither Bit Depth Values 0 = Off, 1 = On -36 - 0 dB -36 - 0 dB 1 - 1000 ms 0 = Off, 1 = On 0 = 8 bits, 1 = 16 bits, 2 = 18 bits, 3 = 20 bits, 4 = 22 bits, 5 = 24 bits, custom

Dither Shape

Mute Dithering
0 = Flat, 1 = 44.1 kHz Shape1, 2 = 44.1 kHz Shape1 Ultra, 3 = 44.1 kHz Shape2, 4 = 44.1 kHz Shape2 Ultra, 5 = 44.1 kHz Shape3, 6 = 44.1 kHz Shape3 Ultra, 7 = 44.1 kHz Shape4, 8 = 44.1 kHz Shape4 Ultra, 9 = 44.1 kHz Shape5, 10 = 96 kHz, 11= 96 kHz Ultra 0 = Off, 1 = On
6.4 Parameter descriptions

Limiter

Following is an explanation of FinalPlug's control parameters: Limiter Turns limiter on and off. Threshold Sets the level at which the limiter starts to take effect on the signal. Ceiling Sets the maximum output level of the signal. It's typically useful to set this slightly below 0.0dB (-0.1dB to -0.3dB 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 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. Truncate/Dither Enable/disables the truncate/dither section of the plug-in. 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 menu or type a value between 4 and 24 in the text field. As an experiment, you can hear the various noise shapes by changing the bit depth to a low value (like 6 or 8) and cycling through the shapes. For this to work it is helpful to have a host application that has a continually running audio engine (like Digital Performer). 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
Wave Arts Power Suite for DVD audio. The 96K shapes concentrate the dither noise to frequencies above 20kHz. Below 20kHz, the dither noise is flat. Flat - No noise shaping. 44.1 kHz Shape1 - Gentle reduction of noise below 10 kHz; noise is concentrated between 15 kHz and 20 kHz. 44.1 kHz Shape1 Ultra - exaggerated version of Shape1. 44.1 kHz Shape2 - Perceptually based noise shaping reduces noise below 5 kHz and concentrates above 15 kHz. 44.1 kHz Shape2 Ultra - exaggerated version of Shape2. 44.1 kHz Shape3 - Evenly reduces noise between 2-13 kHz and concentrates above 15 kHz. Some boost at low frequencies. 44.1 kHz Shape3 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 - 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 Dithering The Mute Dither parameter disables just the Dithering functionality of the Truncate/Dither section. This is useful for auditioning the effect of the dither. For example, set the bit depth to 6, enable truncation, and toggle the Mute Dither to hear the benefit of dither. This can also be used to obtain bit quantization (i.e. "bit crushing") effects.

Binaural Synthesis

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.
Each HRTF, typically consisting of several hundred numbers, describes the time 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. Processing HRTFs into 3D Audio 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).
3-D Audio Using Loudspeakers
Crosstalk 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 listeners left ear only, and the right ear signal should go to the listeners 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.

7. WaveSurround head recording or some other binaural format which already contains 3-D binaural cues. Flat: The 3-D binaural filters have flat frequency response. The filters are implemented with a single delay and gain. More accurately, the filters render an interaural time delay (ITD) and a (frequency-independent) interaural intensity difference (IID). These filters are the most transparent sounding, but have the least spatial effect. Smooth: The 3-D binaural filters have smooth frequency dependent features. These filters have some tonal variation, but better spatial effect than the Flat filters. Medium: The 3-D binaural filters have frequency dependent features of medium complexity. More tonal coloration than the Smooth filters, and better spatial effect. Sharp: The 3-D binaural filters are fully detailed and contain sharp frequencydependent features. These filters have the most tonal coloration and have the sharpest spatial effect Angle Sets the angle of the virtual speakers. This is the angle at which the sound will appear to emanate from with respect to the listener, if the listener is positioned in between the two real loudspeakers (or is wearing headphones). The virtual speaker angle can range from 10 degrees to 120 degrees. At 10 degrees, the effect will narrow the stereo field; 90 degrees is the widest setting; and at 120 degrees, the sound may appear to emanate from behind the listener. Real Angle Sets the position of the real loudspeakers (or headphones) with respect to the listener. Set this parameter to the most likely environment the listener will be in when listening to the music. Possible configurations are: headphones, 10, 20, 30, and 40 degrees.

Reverb

Reverb Turns reverb module on and off. 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 mix parameter sets the perceived distance of the input sound: 100% sounds very distant, and 0% sounds very near. Time Sets the decay time of the reverberation in seconds, from 0.5 sec to 20 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 effective length of the reverberation tail in seconds. The decay time parameter is independent of the room size.