Quick Start

You can quickly burn a CD using existing audio files and default settings for pauses and crossfades. To quickly burn a CD: 1 Open WaveBurner Pro. 2 Choose File > New to create a new project. 3 Drag the audio files you want to add to the Region list (or the Wave View area) in the order you want them to appear on the CD. 4 Click the Burn button. The Burn dialog appears at the top of the window. The name of the CD burner is shown in the Burn dialog. If there is no blank disc in the tray of the CD burner, the Burn dialog prompts you to insert a blank disc. 5 Insert a blank CD-R disc, if you have not already done so. 6 In the Burn dialog, click the Burn button. When you click the Burn button, WaveBurner Pro begins burning your project to the blank CD. The Burn dialog shows the progress of the burn process. When the burn process is complete, the CD is ejected from the CD burner and is ready to play.
The WaveBurner Pro Interface
You create your projects in the main WaveBurner Pro window. You can add and organize the regions in your project, graphically edit regions, pauses, and crossfades, play the project to preview your changes, and add effects plug-ins in the WaveBurner Pro window.
The WaveBurner Pro interface features the following main areas:
Counter: Displays the current track number and the position of the Position Line. Overview: Displays a timeline of the entire project, including all regions. Wave View area: Displays the regions in time order. You can graphically edit regions,
pauses, and crossfades in the Wave View area.
Region list: Lists the regions in time order. You add and organize regions in the

Region list.

Track list: Lists the tracks in the project and displays information for each track. Region and Mix Plug-In lists: You add effects plug-ins and can organize and adjust
plug-in parameters in the Region and Mix Plug-In lists.

Counter

The counter displays the number of the track, and displays the current position of the Position Line in minutes and seconds. The counter updates in real time as you play the project. If the project includes index markers, the current index point is also displayed.

Overview

The Overview displays a miniature version of the Wave View area showing the entire duration of the project. A red rectangle in the Overview shows part of the project currently visible in the Wave View area. You can move to a different part of the project, or zoom in and out, by dragging the rectangle.

Wave View Area

The Wave View area is the main workspace for your projects, You view and graphically edit regions and markers in the Wave View area. Each region is displayed as a green rectangle with a waveform, and an envelope of the regions volume level that you can adjust.
The Wave View area includes the following features:
Marker bars: You add and arrange markers in the marker bars, located at the top and
bottom of the Wave View area.
Time rulers: These show the divisions of time so you can place regions and
transitions at a specific point in time.
Position Line: This shows the point in time currently audible if the project is playing,
or the point playback starts when you click Play.
Transport controls: You control project playback and the position of the Position Line
using the Transport controls.
Edit pointer buttons: Click one of the buttons to change the pointer to an edit pointer

for editing regions.

Marker buttons: Click one of the buttons to change the pointer to a marker pointer,
which you can use to add a marker to the marker bar.
Time, Track, and Index fields: These fields show the current location of the Position
Line, the current track, and the current index point (if the project contains index markers). Track height control: Click to set the track height. Horizontal zoom control: Click the control or drag the slider to zoom in or out. Scroll bar: Drag to change the visible area of the project.

Region List

You add and organize the regions in your project in the Region list. Regions are listed in time order from top to bottom. When you select a region in the Region list, the region is also selected in the Wave View area. You can drag regions to change their order, and Option-drag to copy them.

The Region list displays the following information about each region:
Number Start time Title Length

Track List

The track list displays the track order, and shows information about each track.
The Track list includes the following information for each track:
Number Start time Title Length Pre-emphasis state ISRC code Pause start time Comments
Region and Mix Plug-In Lists
You add and organize plug-ins in the two Plug-In lists. The Region Plug-In list shows the plug-ins for individual regions, and the Mix Plug-In list shows plug-ins for the overall project. You can change the order of plug-ins and view plug-in parameters.
The WaveBurner Pro interface includes several additional windows where you can view region, track, and index point information. There are also windows for the Level Meter and for individual effects plug-ins.

Region Info Window

The Region Info window displays information about the currently selected region and the source audio file for the region. You can edit the information in the Region Info window.
The Parameters pane of the Region Info window displays the following information:
Region name Comments Region length The gap from the end of the previous region Trim settings (at start and at stop) Gain settings for left and right channels Peak time and value Fade-in and fade-out information, including fade type and length
The Audio File pane of the Region Info window displays the following information about the regions source audio file:
Name and location of the audio file on disk File size Audio file format, including file type, sample rate, and bit ordering File length Peak time and value
To show the Region Info window, do one of the following: Choose Region > Region Info. Click the Region Info button.

Track Info Window

The track Info window displays information about the current track. You can edit the information in the Track Info window.
The General pane of the Track Info window displays the following information:
Track number Disc Time (the absolute time where the track starts) Name Comment Pause duration (the pause length before the track starts) ISRC code Pre-emphasis SCMS
The CD TEXT pane of the Track Info window displays the following information:
Track number Title Performer Songwriter Composer Arranger Message
To show the Track Info window, do one of the following: Select the track in the Track list, then choose Disc > Track Info Click the track start marker for the track, then choose Disc > Track Info. Double-click the track start marker for the track.

Index Point Info Window

The Index Point Info window displays information about the currently selected index point.You can edit the information in the Index Point Info window.
The Index Point Info window displays the following information:

Elements of a Project

The elements of a WaveBurner Pro project include audio files, regions, and tracks. Audio files: Audio files are the source material for the regions and tracks in your projects. Regions: When you add an audio file to a project, a region is created. The region can include the entire source audio file or any continuous section of the audio file. When you edit a region in the Wave View area or the Region list, the edits affect only the region, not the source audio file. Tracks: Tracks are the individual selections a user sees in the CD display, and chooses using the forward and back buttons and the track buttons on their CD player. Tracks can include multiple regions, and one region can span multiple tracks.
Creating, Opening, and Saving Projects
The first step in creating a CD in WaveBurner Pro is to create a new project. To create a new project: Choose File > New (or press Command-N). A new blank, untitled project appears. You can name the project when you first save it. To open an existing project: 1 Choose File > Open (or press Command-O). 2 Locate and select the project in the Open dialog, then click Open. To save a project: Choose File > Save (or press Command-S).
Adding Audio Files to a Project

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To add an audio file to a project, do one of the following: Drag the audio file from the Finder to the Region list. Drag the audio file from the Finder to the Wave View area. Click the Import button, select the audio file you want to add, then click Open. Choose Region > Add Audio File, select the audio file you want to add, then click Open.
Previewing Projects and Controlling Playback
As you work on a project, you can play the project to hear the regions, tracks, pauses, and crossfades in the project. You can control project playback using the Transport controls or by moving the Position Line. Using the Transport Controls You can control playback of your project and set the position of the Position Line using the Transport controls, which are located at the bottom center of the Wave View area.
The Transport controls include, from left to right:
Go to Region Start: Sets the Position Line to the beginning of the current region. If

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You can move to different parts of your project by dragging the horizontal scroll bar left or right, or by dragging the red rectangle in the Overview left or right. You can also move to the start of the previous track or the next track using keyboard shortcuts. To move to the start of the previous track: Press Command-Left Arrow To move to the start of the next track: Press Command-Right Arrow
Setting WaveBurner Pro Preferences
You can set preferences for various aspects of WaveBurner Pro in the Preferences window. Some preferences apply to the current project, and others apply when you create a new project. You may want to set some preferences before you begin working on your projects. To open the Preferences window: Choose WaveBurner Pro > Preferences (or press Command-comma). General Preferences
Default Pause Length: Sets the default pause length inserted when you add a region.
The format is minutes, seconds, and CDDA frames. Zoom to Position Line: When turned on, the Position line remains centered in the Wave View area. Action at Start: Sets the default behavior when you open WaveBurner Pro. You can choose New document (which creates a new, blank project), Open document (which displays an Open dialog), or Nothing.

Audio Driver Preferences

Driver: Choose the audio device for input and output from the pop-up menu. Input: Choose the input channel or channels from the pop-up menu. The number of
channels available depends on the device selected in the Driver pop-up menu.
Output: Choose the output channel or channels from the pop-up menu. The number
of channels available depends on the device selected in the Driver pop-up menu. Burn Preferences
Write CD TEXT Data: Turn on to include CD TEXT on the CD. Write Index Points: Turn on to include index point information on the CD. Write UPC/EAN Code: Turn on to include UPC and EAN codes on the CD. Write ISRC Codes: Turn on to include ISRC codes on the CD. Write Pre-Emphasis: Turn on to include pre-emphasis on the CD. Write SCMS: Turn on to include SCMS on the CD.

Bounce Preferences

Dithering: Choose the type of dithering to use when the project is bounced or
burned from the pop-up menu. The choices are: POW-r #1 (Dithering), POW-r #2 (Noise Shaping), POW-r #3 (Noise Shaping), and No Dithering. For information about when to use dithering, see the Logic Pro 7 Reference Manual.
Replace Region with Bounce: When turned on, the new region created by choosing
Region > Bounce Region replaces the original region being bounced.
Clipping detection stops Bounce: When turned on, the bounce process is stopped
when clipping is detected in the region or project being bounced.

Working With Regions

Country code (compliant with ISO 3166, in this case, Germany) First owner code (record company, in this case, Polydor) Year of recording code (2 digits) Designation code (5 digits)
The ISRC is used by radio stations to archive recordings. Royalty collection societies such as GEMA or MCPS/PRS also use the code to automate generation of transmission logs, and thus simplify licensing accounting. For commercial recordings, the ISRC should only consist of the code provided by the record label. If the record label company has not been issued a first owner code, you should contact the IFPI. To add an ISRC code: Click the tracks ISRC field, then type the ISRC code in the field.

The CD Track Info Window

You can also view and edit track information in the CD Track Info window, including copy protection and pre-emphasis To show the CD Track Info window, do one of the following: Select the track in the Track list, then choose Disc > Track Info Click the track start marker for the track, then choose Disc > Track Info. Double-click the track start marker for the track. Setting the Copy Prohibit Bit You can set a Copy Prohibit Bit for a track. When a tracks Copy Prohibit Bit is set, the track cannot be copied digitally more than once by recording devices that support the Serial Copy Management System (SCMS). These devices automatically write a copy prohibit bit to prevent further generations of digital copies of the track. Consumer-level DAT recorders (but frequently not professional-level ones) are usually equipped with SCMS. However, because many recording devices do not include this type of copy protection, setting the Copy Prohibit Bit does not guarantee that the track cannot be copied multiple times. WaveBurner Pro provides the option to set the Copy Prohibit Bit in order to comply with the Red Book standard. To set the Copy Prohibit Bit: In the General Pane of the CD Track Info window, choose Protected original or Protected copy from the SCMS pop-up menu.
Setting Track Pre-Emphasis The use of pre-emphasis dates from the early days of digital sound reproduction, when 14-bit A/D converters without oversampling were frequently used. The high frequencies of digital recordings were boosted or emphasized during conversion (similar to the RIAA EQ process for vinyl records), then were attenuated (de-emphasis) after D/A conversion. This process tended to mask the inaccuracies of the conversion. With todays advanced digital recording technology, it is recommended that you leave pre-emphasis turned off, because modern converters are capable of delivering linear conversion that produces substantially fewer errors than de-emphasis filters. WaveBurner Pro supports pre-emphasis only so that old digital recordings that were processed using pre-emphasis can be marked as such. Pre-emphasis is a display-only option, and will not change the way in which the audio is processed when you burn a CD. To turn on pre-emphasis: In the General Pane of the CD Track Info window, turn on Pre-Emphasis.

Bouncing Tracks

You can bounce a track. As with bouncing a region, bouncing a track has two benefits: it lets you save an archive of the track, and it can help conserve processing power when you burn a CD. To bounce a track: 1 In the Track list, select the track. 2 Choose Disc > Bounce Track. 3 In the Save dialog, choose a location to save the track, then click OK.

Working With Markers

Markers are part of the Red Book standard for the CD format. You use markers to indicate a specific point in time on the CD. There are several types of markers, including track start, pause start, and index markers.

Types of Markers

WaveBurner Pro supports the following types of markers: Track start markers: Indicate the start of a track, when someone playing the finished CD uses the forward, back, or track number controls on their CD player. Pause start markers: Indicate the start of a short period of silence before the next track starts. Index markers: Indicate the position of index points, which are additional points in time within a track. Not all CD players can recognize and locate index points.
Inserting and Deleting Markers
When you add an audio file to a project, a track start marker is inserted at the beginning of the region created from the audio file, and a pause start marker is inserted at the end of the preceding region (including the first region). You can use the default placement of the markers, or move them to adjust the pause length. You can insert track start markers manually to define tracks separately from the beginning and end of regions in the project. You insert a track start marker and a pause start marker together, separated by a short period of silence called the pause length. The default pause length is defined in the General Preferences pane. For information about setting the default pause length, see General Preferences on page 15. To insert track start and pause start markers: 1 Click the Track Start Marker button (the purple marker button to the left of the Transport controls). 2 Click either marker bar at the point in time you want to insert the track start marker. The track start marker and pause start marker are inserted at the point where you click. You can move the pause start marker left to create a pause before the track starts. Index markers indicate index points, which are specific time positions within a track. However, not all CD players can recognize index points. To insert an index marker: 1 Click the Index Marker button (the orange marker button to the left of the Transport controls). 2 Click either marker bar at the point in time you want to insert the index marker. To delete a marker: Click the marker to select it, then press the Delete key.

WaveBurner Pro includes a full suite of mastering effects you can use in your projects. The effects bundled with WaveBurner Pro include: Dynamics: Compressor, Multipressor, and Limiter Equalization: Fat EQ and Linear Phase EQ Noise reduction: Denoiser Stereo enhancement: Stereo Spread Audio analysis: Correlation Meter, Multimeter, and Level Meter The following section briefly describes each of the bundled plug-ins.

Compressor

The Compressor tightens up the dynamics of a signal by lowering the volume when it rises above a certain level, called the Threshold. This decreases the difference between the softest and loudest parts of the music, increasing the perceived volume. This can give the sound more focus by making the key parts stand out while preventing the accompanying parts from becoming lost or inaudible. In addition to the Threshold parameter, the most important parameter for the Compressor is the Ratio. The Ratio parameter determines how much sounds above the threshold are lowered. The Ratio is expressed as a percentage of the original signal. For example, if you set the Threshold to 12 dB and set the Ratio to 2:1, a sound at 7 db (5 dB above the Threshold) is reduced by 2.5 dB, and a sound at 2 dB (10 dB above the Threshold) is reduced by 5 dB. Two other important parameters are Attack and Release. The Attack parameter controls how quickly the Compressor reacts when sounds reach the threshold. At higher values, the compressor does not fully compress the signal after the attack time ends. This ensures that the original attack transient (for example, the sound of a pick or finger striking a guitar string) remains intact or clearly audible. The Release parameter determines how quickly the Compressor reacts once the sound falls below the threshold again. If the compressor produces an undesirable pumping sound, adjusting the Release parameter can eliminate or minimize the pumping sound. Because the compressor lowers sounds above the threshold, the overall output signal is often lower than the input signal. You can compensate for the decreased output level by raising the Gain parameter. Turning on Auto Gain ensures that a normalized input signal is amplified so that the output signal is also normalized, regardless of the values for Threshold and Ratio (at least when you are working with relatively static signals). The Compressor can use one of two methods to determine when an input signal reaches or exceeds the threshold: either the Peak or RMS level. RMS level is a better indication of the signals perceived loudness. However, turning on Auto Gain and RMS simultaneously may cause the signal to become saturated. If the signal sounds distorted, turn auto gain off, and adjust the gain parameter instead.

Multipressor

The Multipressor (short for multiband compressor) is used as a mastering tool in many different situations. The Multipressor splits the incoming signal into two to four different frequency bands, each of which is compressed independently. This permits higher levels of compression without producing a pumping effect. As the name suggests, the Multipressor is like a set of compressors, each working independently on a different part of the frequency range of an input signal. For each frequency band, you set the Threshold, Ratio, Attack, and Release parameters as with the Compressor. Using the Multipressor allows you to raise the overall volume level, which can dramatically increase the amount of low-level noise (known as the noise floor). To reduce the noise floor, each frequency band features Expansion (also called downward expansion), which acts as a counterpart to the compressor for that band. The compressor decreases the dynamic range at higher volume levels, and Expansion increases the dynamic range at lower volume levels. When you apply Expansion, the signal is reduced when it is lower than the threshold. The effect is similar to a noise gate, but instead of cutting off the sound abruptly, it smoothly fades the volume using an adjustable ratio. The Multipressor has several other important parameters, including Bands and Lookahead. The Bands parameter lets you choose whether two, three, or four independent frequency bands are compressed; the higher the number, the more processing power the Multipressor uses. Classic multi-band compressors typically use three bands. The Lookahead parameter lets you control how far ahead in time from the current point in the signal the Multipressor looks or analyzes the signal, allowing it to react more quickly to peak volumes. Unlike hardware compressors, using Lookahead does not cause a delay in the signal, because the Multipressor can read the audio file on disk and does not need to analyze the signal in real time.

Limiter

The Limiter prevents the audio signal from exceeding a maximum volume level. While a compressor gradually lowers levels above the threshold, the Limiter puts a hard limit on any signal louder than the Gain level (analogous to the threshold setting on a hardware limiter). By keeping the Gain parameter below 0 dB, the Limiter can help your projects have the greatest perceived loudness without clipping. The Limiter has lookahead and release parameters which work similarly to those for the Compressor and Multipressor. In addition, it has an Output Level slider and a Softknee button. The Output Level parameter controls the output volume, independent of the Gain setting. Turning on the Softknee button, as the name suggests, slightly softens the degree to which sounds above the Gain setting are lowered. When the Softknee button is turned off, sounds above the Gain setting are limited using a completely linear curve.

Fat EQ

The Fat EQ plug-in lets you apply equalization using up to five fully parametric bands, and provides great flexibility in shaping the sound of your projects. In the Fat EQ plug-in window, the bands are arranged in increasing frequency order from left to right. Using the buttons numbered 1 through 5, you can turn each band on or off. The circular slider for each band lets you set the amount to increase (boost) or decrease (cut) the part of the signal in that band. By dragging the number in the value field (located directly above the slider), you can raise or lower the bands frequency. You can set the amount of resonance or Q for each band by dragging the number in the value field located just above the on/off button. Band 1 can function as a low cut or low shelving filter. Bands 2 and 4 can be switched from their normal operating mode as fully parametric band pass filters to low or high shelving filters. The center band (3) always operates as a fully parametric band pass. Band 5 can function as either a high cut or high shelving EQ. You select the filter type for each band by turning on or off the graphic button for the band located at the top of the plug-in window, above the graphic display.

Linear Phase EQ

The Linear Phase EQ offers up to eight bands of equalization and an integrated Fast Fourier Transform (FFT) analyzer. It features linear-phase filtering, which means using the Linear Phase EQ results in no phase distortion on the audio signal. Linear-phase filtering does add latency, about 50 ms. The parameters for the Linear Phase EQ are similar to those for Fat EQ. For each frequency band, you drag the number up or down to raise or lower the bands frequency and resonance (Q). Instead of a slider to adjust the amount of boost or cut, you drag the number (expressed in dB, except for the bottom and top bands, where the low cut and high cut filters are expressed in dB/Octave). You can also drag the curves in the graphic display to edit them directly. You can also turn on the Analyzer to view the frequency content of the signal as it plays. You can set the resolution of the Analyzer by clicking the bottom rectangle button below the Post-Fader button, then choosing the resolution you want from the shortcut menu. High resolutions are recommended to achieve reliable results with very low bass frequencies. The bands derived from FFT analysis are divided in accordance with the frequency linear principlemeaning that there are many more bands for higher octaves than for lower ones. You can reset all parameters by Option-clicking in the display area, or reset individual parameters by Option-clicking in the parameter area. After boosting or cutting frequency bands, you can use the Master Gain fader to adjust the Output level. Use the scales to the left and right of the EQ display to change the vertical scale of the EQ and analyzer curves.

Goniometer The Goniometer helps you to determine the coherence of the stereo image. Using the Goniometer, you can see phase problems as trace cancellations along the center-line (M=mid/mono). Goniometers developed when early two channel oscilloscopes first appeared. Users would connect the left and right stereo channels to the X and Y inputs while rotating the display by 45 degrees, resulting in a useful visualization of the signals stereo phase. The signal trace slowly fades to black, imitating the glow of the tubes found in older Goniometers, and at the same time enhancing readability. Clicking the Goniometer button turns on the Goniometer and turns off the Spectrum Analyzer. You can use the Auto Gain display parameter in order to obtain a higher readout on low-level passages. Auto Gain allows the display to automatically compensate for low input levels. You can set the amount of compensation with the Auto Gain parameter, or set Auto Gain by dragging directly in the display area of the Goniometer. Note that Auto Gain is a display parameter only and increases the display for better readability. The actual audio levels are not touched by this parameter.
Using Audio Effects Plug-Ins
In addition to the included effects plug-ins, you can add plug-ins in the Audio Units format. Audio Units plug-ins are available from Apple and third-party manufacturers. When adding third-party plug-ins to your computer, be sure to read the documentation, including any Read Me and installation files that came with the plug-in. Apple Audio Units plug-ins appear in the Apple submenu of the Add Plug-in pop-up menu. Third-party Audio Units plug-ins appear in the submenu with the manufacturers name in the Add Plug-In pop-up menu. You add Audio Units plug-ins in the same way as the included effects, by choosing the plug-in you want to add from the appropriate submenu of the Add Plug-In menu. You adjust Audio Units plug-in parameters, view a plug-ins window, and delete a plug-in just as you do with the included plug-ins.
Adding and Deleting Plug-Ins

You add plug-ins to regions in the Region Plug-Ins list, and add plug-ins to the overall project in the Mix Plug-Ins list. To add a plug-in to a region: 1 Click the Region Plug-Ins tab to show the Region Plug-Ins list. 2 In the Region list, click the region to which you want to add the plug-in. 3 Choose the category of plug-in you want to add from the Add Plug-In pop-up menu, then choose the plug-in from the submenu. The plug-in appears in the Region Plug-Ins list, below any plug-ins already added to the region. To add a plug-in to the overall project: 1 Make sure no region is selected in the Region Plug-Ins list. 2 Click the Mix Plug-Ins tab to show the Mix Plug-Ins list. 3 Choose the category of plug-in you want to add from the Add Plug-In pop-up menu, then choose the plug-in from the submenu. The plug-in appears in the Mix Plug-Ins list, below any plug-ins already added to the project. If you no longer want to use a plug-in, you can delete it from the list. To delete a plug-in: Select the plug-in in either the Region Plug-Ins or Mix Plug-Ins list, then press the Delete key.

Bypassing Plug-Ins

You can bypass a plug-in in order to hear the region or project without the plug-in, without losing the changes youve made to the plug-ins parameters. To bypass a plug-in: In the Plug-In list containing the plug-in, click the checkbox at the right of the row. Click the checkbox again to hear the plug-in again.
Adjusting Plug-In Parameters
Each plug-in contains a set of parameters you use to control the way the plug-in shapes the sound. You can view and adjust a plug-ins parameters in its Plug-In window. To view a plug-ins parameters: In the Plug-In list in which the plug-in appears, click the plug-ins disclosure triangle. To view the Plug-In window for a plug-in: In the Plug-In list in which the plug-in appears, double-click the name of the plug-in. The Plug-In window for the plug-in appears. You can adjust the plug-ins parameters in the Plug-In window. To adjust plug-in parameters: In the Plug-In window, move the slider, type in the field, or click the button for the parameter you want to adjust.

Using the Level Meter

The Level Meter shows the level of the input signal in real time. When recording digital audio, it is important that the input level not exceed 0 dB, or clipping will occur. Unlike analog clipping, which can produce musically desirable results, digital clipping results in harsh distortion that is undesirable in most situations. You can monitor the audio output levels of your projects using the Level Meter. As a project plays back, the levels in each stereo channel change constantly with the rising and falling of the audio signal. The Level Meter shows these changes as blue bars moving from left to right; the farther right, the higher the level for that channel. Momentary peaks are shown as thin yellow bars in each channel. If there is clipping at any point in the project, the bars in the Level Meter become red past the 0 dB mark, acting as clipping indicators. You can check your project for clipping. To check a project for clipping, do one of the following: Choose Disc > Check Disc for Clipping. Click the Check Disc for Clipping button.

Burning a Project to a CD
When your project is complete, you burn the project to a CD. The resulting audio CD conforms to Red Book standards and can be played on any audio CD player.
Getting Ready to Burn a CD
Before you burn your project to a CD, there are several things you should do to prepare: Check that the CD burner is connected, turned on, and working. Check that WaveBurner Pro recognizes the CD burner. Preview the transitions between tracks. Set Disc Options for the project.

Supported CD Burners

WaveBurner Pro supports all CD burners supported by Mac OS X, including SuperDrives and external third-party CD burners.

Setting Disc Options

You can set many disc options for a project, including the following: Adding a UPC/EAN code Setting offsets for start points, stop points, and indexes Setting the default pause length Adding a period of silence at the end of the CD Adding CD TEXT information, including title, performer, songwriter, composer, and arranger information, and a text message. Adding a UPC/EAN Code Commercially produced CDs typically include a Universal Product Code (UPC) and a European Article Number (EAN). These codes contain information about the record company producing the CD, and may contain additional information. To add a UPC/EAN code: 1 Choose Disc > Disc Options, then click the General tab. 2 Type the code in the UPC/EAN Code field. 3 When you are finished, click OK.
Setting Offsets You can change the position of start point, end point, and index markers globally by setting an offset. The unit of value for offsets is one CDDA frame (1/75th of a second). When you set an offset, the markers are moved immediately before the project is burned to a CD, then moved back to their original positions after the CD is burned. The purpose of setting offsets is to compensate for inaccurate timing in some CD players, particularly older players. The Disc Options window includes offset fields for the first start point, other start points, stop points, and index points, allowing you to offset any or all of these markers. The first start point is a special case, because the Red Book standard stipulates that there must be a two- to three-second pause before the first track begins. Typing a value greater than 0 (zero) in the First Start Point field, WaveBurner Pro inserts a corresponding period of silence before the first track, and every track is shifted by the amount of time required to maintain the period of silence. The overall length of the CD is increased by the number of CDDA frames required to maintain the initial period of silence. To set offsets: 1 Choose Disc > Disc Options, then click the General tab. 2 Type the offsets you want to set in the First Start Point, Other Start Points, End Points, and Indexes fields. (in CDDA frames) 3 When you are finished, click OK.

Model Small British Combo Boutique British Combo
Description A 1 x 12" combo with half the power of the British Combo, this amp offers a slightly darker, less open tone. A 2 x 12" combo that is a modern take on the original 1960s sound. The tone is thicker, with stronger lows and milder highs than the other British Combos.
Tip: Using high Treble and Presence knob settings that might become strident on other amp types can sound great with the British Combos.

British Alternatives

The late 1960s amplifier heads and combos that inspired the Sunshine models are loud and aggressive, with full-bodied mid frequencies. These amps are not just for single note solos and power chords, as they can sound great with big, open chordsone reason why they were embraced by the Brit-pop bands of the 1990s. The Stadium amps are famed for their ability to play ultra-loud without dissolving into mushy distortion. They retain crisp treble and superb note definition, even at maximum gain settings.
Model Sunshine Stack Small Sunshine Combo Description A robust-sounding head paired with a 4 x 12" cabinet. Its a great choice for powerful pop-rock chords. A 1 x 12" combo based on a modern amp known for a big amp sound. It is brighter than the Sunshine Stack head, with a touch of 1960s British Combo flavor. A classic head and 4 x 12" cabinet configuration popular with 1970s arena rock bands. Its tones are cleaner than other Amp Designer 4 x 12" stacks, while still retaining body and impact. A good choice if you need power and clarity. A 2 x 12" combo based on a modern amp. The tone is a little smoother and rounder than that of the Stadium Stack.

Stadium Stack

Stadium Combo
Tip: The tone of the Sunshine Stack can seem dark at times, but a high Treble knob setting opens up the sound. While the Small Sunshine Combo sounds great with its default 1 x 12" cabinet, it also shines through a 4 x 12" cabinet. The Stadium amps can be slow to distort, so most famous users have paired them with aggressive fuzz pedals. Try combining it with Pedalboards Candy Fuzz or Fuzz Machine stompboxes. See Distortion Pedals and Pedalboard.

Metal Stacks

The Metal Stack models are inspired by the powerful, ultra-high gain amplifier heads that put the chunk into modern hard rock and metal music. All are paired with 4 x 12" cabinets. Their signature tones range from heavy distortion to extremely heavy distortion. If you want powerful lows, razor-edged highs, and serious sustain, these are the models you should look to first.

Choosing a Guitar Amp Pro Equalizer Setting Guitar Amp Pro Microphone Parameters
Choosing a Guitar Amp Pro Amplifier
You can choose an amplifier model from the Amp pop-up menu near the top of the interface. UK Combo 30W: Neutral-sounding amp, well-suited for clean or crunchy rhythm parts. UK Top 50W: Quite aggressive in the high frequency range, well-suited for classical rock sounds. US Combo 40W: Clean sounding amp model, well-suited for funk and jazz sounds. US Hot Combo 40W: Emphasizes the high mid-frequency range, making this model ideal for solo sounds. US Hot Top 100W: This amp produces very fat sounds, even at low Master settings, that result in broad sounds with a lot of oomph. Custom 50W: With the Presence parameter set to 0, this amp model is well-suited for smooth fusion lead sounds. British Clean (GarageBand): Simulates the classic British Class A combos used continuously since the 1960s for rock music, without any significant modification. This model is ideally suited for clean or crunchy rhythm parts. British Gain (GarageBand): Emulates the sound of a British tube head and is synonymous with rocking, powerful rhythm parts and lead guitars with a rich sustain. American Clean (GarageBand): Emulates the traditional full tube combos used for clean and crunchy sounds. American Gain (GarageBand): Emulates a modern Hi-Gain head, making it suitable for distorted rhythm and lead parts. Clean Tube Amp: Emulates a tube amp model with very low gain (distortion only when using very high input levels or Gain/Master settings).
Choosing a Guitar Amp Pro Speaker Cabinet
The speaker cabinet can have a huge bearing on the type of tones you can extract from your chosen amplifier. The speaker parameters are found near the top of the interface. Speaker pop-up menu: You can choose one of the 15 speaker models: UK 1 x 12 open back: Classic open enclosure with one 12" speaker, neutral, well-balanced, multifunctional. UK 2 x 12 open back: Classic open enclosure with two 12" speakers, neutral, well-balanced, multifunctional. UK 2 x 12 closed: Loads of resonance in the low frequency range, therefore well-suited for Combos: crunchy sounds are also possible with low Bass control settings.
UK 4 x 12 closed slanted: when used in combination with off-center miking, you will get an interesting mid frequency range; therefore, this model works well when combined with High Gain amps. US 1 x 10 open back: Not much resonance in the low frequency range. Suitable for use with blues harmonicas. US 1 x 12 open back 1: Open enclosure of an American lead combo with a single 12" speaker. US 1 x 12 open back 2: Open enclosure of an American clean/crunch combo with a single 12" speaker. US 1 x 12 open back 3: Open enclosure of another American clean/crunch combo with a single 12" speaker. US broad range: Simulation of a classic electric piano speaker. Analog simulation: Internal speaker simulation of a well-known British tube preamplifier. UK 1 x 12 (GarageBand): A British Class A tube open back with a single 12" speaker. UK 4 x 12 (GarageBand): Classic closed enclosure with four 12" speakers (black series), suitable for rock. US 1 x 12 open back (GarageBand): Open enclosure of an American lead combo with a single 12" speaker. US 1 x 12 bass reflex (GarageBand): Closed bass reflex cabinet with a single 12" speaker. DI Box: This option allows you to bypass the speaker simulation section. Amp-Speaker Link button: Located between the Amp and Speaker pop-up menus, links these pop-up menus so that when you change the amp model, the speaker associated with that amp is loaded automatically.

Using Guitar Amp Pros Tremolo and Vibrato Effects
Tremolo and vibrato are controlled by an On button, the FX pop-up menu, the Depth and Speed knobs, and the Sync button in the Effects section. Tremolo modulates the amplitude or volume of the sound, and vibrato modulates the pitch. FX pop-up menu: You can choose either Tremolo or Vibrato. Depth knob: Sets the intensity of the modulation. Speed knob: Sets the speed of the modulation in Hertz. Lower settings produce a smooth and floating sound, while higher settings produce a rotor-like effect. Sync button: When the Sync button is turned on, the modulation speed is synchronized to the project tempo. You can adjust the Speed knob to select bar, beat, and musical note values (including triplet and dotted notes). When the Sync button is turned off, the modulation speed can be set to any available value with the Speed knob.
Using Guitar Amp Pros Reverb Effect
Reverb is controlled by an On button, the Reverb pop-up menu, and a Level knob in the Reverb section near the bottom. Reverb can be added to either the Tremolo or Vibrato effect, or used independently. Reverb pop-up menu: Choose one of the three types of spring reverb. Level knob: Sets the amount of reverb applied to the pre-amplified amp signal.
Setting Guitar Amp Pro Microphone Parameters
After choosing a speaker cabinet from the Speaker menu, you can set the type of microphone you want to be emulated, and where the microphone is placed in relation to the speaker. The Microphone Position parameters are available in the yellow area to the left, and the Microphone Type parameters in the yellow area to the right. Microphone Position Parameters Centered button: Places the microphone in the center of the speaker cone, also called on-axis. This placement produces a fuller, more powerful sound, suitable for blues or jazz guitar tones. Off-Center button: Places the microphone on the edge of the speaker, also referred to as off-axis. This placement produces a tone that is brighter and sharper, but also thinnersuitable for cutting rock or R & B guitar parts. When you select either button, the graphic speaker display reflects your choice. Microphone Type Parameters Condenser button: Emulates the sound of a studio condenser microphone. The sound of condenser microphones is fine, transparent, and well-balanced.

Robo Flanger

Roswell Ringer

Roto Phase

Spin Box

Stompbox Total Tremolo

Description A flexible tremolo effect (modulation of the signal level). Rate sets the modulation speed and can run freely, or be synchronized with the host application tempo by enabling the Sync button. When synchronized, you can specify bar, beat and note values (including triplets and dotted notes). Depth sets the strength of the effect. Wave and Smooth work in combination to alter the waveform shape of the LFO. This enables you to create floating changes in level, or abrupt steps. Volume determines the output level of the effect. The 1/2 and 2 x Speed buttons immediately halve or double the current Rate value. Hold down the Speed Up and Slow Down buttons to gradually accelerate or reduce the current Rate value to the maximum or minimum possible values. A tremolo effect (modulation of the signal level). Rate sets the modulation speed and can run freely, or be synchronized with the host application tempo by enabling the Sync button. When synchronized, you can specify bar, beat and note values (including triplets and dotted notes). Depth sets the strength of the effect. Level sets the post-tremolo gain. A vibrato/chorus effect based on the Scanner Vibrato unit found in the Hammond B3 organ. You can choose from three vibrato (V13) or chorus (C13) variations with the Type knob. Rate sets the modulation speed and can run freely, or be synchronized with the host application tempo by enabling the Sync button. When synchronized, you can specify bar, beat and note values (including triplets and dotted notes). Depth sets the strength of the effect. See Scanner Vibrato Effect for background information on this effect.

Trem-o-Tone

the Vibe

Delay Pedals

This section describes the Delay effects pedals.
Stompbox Blue Echo Description A delay effect. Time sets the modulation speed and can run freely, or be synchronized with the host application tempo by enabling the Sync button. When synchronized, you can specify bar, beat and note values (including triplets and dotted notes). The Repeats knob determines the number of delay repeats. Mix sets the balance between the delayed and source signals. The Tone Cut switch controls a fixed frequency internal filter circuit that allows more low (Lo) or high (Hi) frequency content to be heard. You can also disable this filter circuit by choosing Off. A spring reverb pedal. Time sets the length of the reverberation to short, medium, or long values. Tone controls the cutoff frequency, making the effect brighter or darker. Style switches between algorithms, each with different characteristics. You can choose from: Boutique, Simple, Vintage, Bright, and Resonant. Mix sets the ratio between the source and effect signals.

Getting to Know the Delay Designer Interface
The Delay Designer interface consists of five main sections:
Sync section Main display Master section

Tap pads

Tap parameter bar
Main display: Provides a graphic representation of all taps. You can see, and edit, the parameters of each tap in this area. See Getting to Know Delay Designers Main Display.

Chapter 2 Delay Effects

Tap parameter bar: Offers a numeric overview of the current parameter settings for the selected tap. You can view and edit the parameters of each tap in this area. See Editing Taps in Delay Designers Tap Parameter Bar. Tap pads: You can use these two pads to create taps in Delay Designer. See Creating Taps in Delay Designer. Sync section: You can set all Delay Designer synchronization and quantization parameters in this section. See Synchronizing Taps in Delay Designer. Master section: This area contains the global Mix and Feedback parameters. See Using Delay Designers Master Section.
Getting to Know Delay Designers Main Display
Delay Designers main display is used to view and edit tap parameters. You can freely determine the parameter shown, and quickly zoom or navigate through all taps.
Toggle buttons Tap display View buttons Autozoom button

Identification bar

Overview display
View buttons: Determine the parameter or parameters represented in the Tap display. See Using Delay Designers View Buttons. Autozoom button: Zooms the Tap display out, making all taps visible. Turn Autozoom off if you want to zoom the display in (by dragging vertically on the Overview display) to view specific taps. Overview display: Shows all taps in the time range. See Zooming and Navigating Delay Designers Tap Display.
Toggle buttons: Click to enable or disable the parameters of a particular tap. The parameter being toggled is chosen with the view buttons. The label at the left of the toggle bar always indicates the parameter being toggled. For more information, see Using Delay Designers Tap Toggle Buttons. Tap display: Represents each tap as a shaded line. Each tap contains a bright bar (or dot for stereo panning) that indicates the value of the parameter. You can directly edit tap parameters in the Tap display area. For more details, see Editing Parameters in Delay Designers Tap Display. Identification bar: Shows an identification letter for each tap. It also serves as a time position indicator for each tap. You may freely move taps backward or forward in time along this bar/timeline. See Moving and Deleting Taps in Delay Designer.

Chapter 4 Dynamics Processors

Adaptive Limiter

The Adaptive Limiter is a versatile tool for controlling the perceived loudness of sounds. It works by rounding and smoothing peaks in the signal, producing an effect similar to an analog amplifier being driven hard. Like an amplifier, it can slightly color the sound of the signal. You can use the Adaptive Limiter to achieve maximum gain, without introducing generally unwanted distortion and clipping, which can occur when the signal exceeds 0 dBFS. The Adaptive Limiter is typically used on the final mix, where it may be placed after a compressor, such as the Multipressor, and before a final gain control, resulting in a mix of maximum loudness. The Adaptive Limiter can produce a louder-sounding mix than can be achieved by normalizing the signal. Note: Using the Adaptive Limiter adds latency when the Lookahead parameter is active. Usually it should be used for mixing and mastering previously recorded tracks, not while recording.
Input meters (to the left): Show the input levels in real time as the file or project plays. The Margin field shows the highest input level. You can reset the Margin field by clicking it. Input Scale knob and field: Scales the input level. Scaling is useful for handling very high-level or low-level input signals. It essentially squeezes the higher and lower signal levels into a range that allows the Gain knob to work effectively. In general, the input level should never exceed 0 dBFS, which can result in unwanted distortion. Gain knob and field: Sets the amount of gain after input scaling.
Out Ceiling knob and field: Sets the maximum output level, or ceiling. The signal will not rise above this. Output meters (to the right): Show output levels, allowing you to see the results of the limiting process. The Margin field shows the highest output level. You can reset the Margin field by clicking it. Mode buttons (Extended Parameters area): Choose the type of peak smoothing: OptFit: Limiting follows a linear curve, which allows signal peaks above 0 dB. NoOver: Avoids distortion artifacts from the output hardware by ensuring that the signal does not exceed 0 dB. Lookahead field and slider (Extended Parameters area): Adjusts how far ahead the Adaptive Limiter analyzes the file for peaks. Remove DC checkbox (Extended Parameters area): Enable to activate a highpass filter that removes direct current (DC) from the signal. DC can be introduced by lower-quality audio hardware.

Ducker

Ducking is a common technique used in radio and television broadcasting: When the DJ or announcer speaks while music is playing, the music level is automatically reduced. When the announcement has finished, the music is automatically raised to its original volume level. Ducker provides a simple means of achieving this result with existing recordings. It does not work in real time. Note: For technical reasons, Ducker can only be inserted in output and aux channel strips.

Ducker Parameters

The Ducker has the following parameters:
Ducking On and Off buttons: Enable or disable ducking. Lookahead On and Off buttons: Enable to ensure that the Ducker reads the incoming signal before processing. This results in no latencyit is primarily intended for slower computers. Amount slider and field: Defines the amount of volume reduction of the music mix channel strip, which is, in effect, the output signal. Threshold slider and field: Determines the lowest level that a side-chain signal must attain before it begins to reduce the music mix output levelby the amount set with the Intensity slider. If the side-chain signal level doesnt reach the threshold, the music mix channel strip volume is not affected.
Attack slider and field: Controls how quickly the volume is reduced. If you want the music mix signal to be gently faded out, set this slider to a high value. This value also controls whether or not the signal level is reduced before the threshold is reached. The earlier this occurs, the more latency is introduced. Note: This only works if the ducking signal is not livethe ducking signal must be an existing recording. The host application needs to analyze the signal level before it is played back in order to predefine the point where ducking begins. Hold slider and field: Determines the duration for which the music mix channel strip volume is reduced. This control prevents a chattering effect that can be caused by a rapidly changing side-chain level. If the side-chain level hovers around the threshold value rather than clearly exceeding or falling short of it, set the Hold parameter to a high value to compensate for any rapid volume reductions. Release slider and field: Controls how quickly the volume returns to the original level. Set it to a high value if you want the music mix to slowly fade up after the announcement.

Using the Ducker

The steps below show how to use the Ducker on existing recordings. Note: For technical reasons, the Ducker plug-in can be inserted only in output and aux channel strips. To use the Ducker plug-in 1 Insert the plug-in into an aux channel strip. 2 Assign all channel strip outputs that are supposed to duck (dynamically lower the volume of the mix) to a busthe aux channel strip chosen in step 1. 3 Choose the bus that carries the ducking (vocal) signal in the Side Chain menu of the Ducker plug-in. Note: Unlike all other side-chain-capable plug-ins, the Ducker side chain is mixed with the output signal after passing through the plug-in. This ensures that the ducking side-chain signalthe voice-overis heard at the output. 4 Adjust the Ducker parameters.

Gain reduction meter: Shows the amount of limiting in real time. Gain slider and field: Sets the amount of gain applied to the input signal. Lookahead slider and field: Adjusts how far ahead in milliseconds the Limiter analyzes the audio signal. This enables it to react earlier to peak volumes by adjusting the amount of reduction. Note: Lookahead causes latency, but this has no perceptible effect when you use the Limiter as a mastering effect on prerecorded material. Set it to higher values if you want the limiting effect to occur before the maximum level is reached, thus creating a smoother transition. Release slider and field: Sets the amount of time, after the signal falls below the threshold level, before the Limiter stops processing. Output Level knob and field: Sets the output level of the signal. Softknee button: When active, the signal is limited only when it reaches the threshold. The transition to full limiting is nonlinear, producing a softer, less abrupt effect, and reducing distortion artifacts that can be produced by hard limiting.

Multipressor

The Multipressor (an abbreviation for multiband compressor) is an extremely versatile audio mastering tool. It splits the incoming signal into different frequency bandsup to fourand enables you to independently compress each band. After compression is applied, the bands are combined into a single output signal. The advantage of compressing different frequency bands separately is that it allows you to apply more compression to the bands that need it, without affecting other bands. This avoids the pumping effect often associated with high amounts of compression. As the Multipressor allows the use of higher compression ratios on specific frequency bands, it can achieve a higher average volume without causing audible artifacts. Raising the overall volume level can result in a corresponding increase in the existing noise floor. Each frequency band features downward expansion, which allows you to reduce or suppress this noise. Downward expansion works as a counterpart to compression. Whereas the compressor compresses the dynamic range of higher volume levels, the downward expander expands the dynamic range of the lower volume levels. With downward expansion, the signal is reduced in level when it falls below the threshold level. This works in a similar way to a noise gate, but rather than abruptly cutting off the sound, it smoothly fades the volume with an adjustable ratio.

Multipressor Parameters

The parameters in the Multipressor window are grouped into three main areas: the graphic display in the upper section, the set of controls for each frequency band in the lower section, and the output parameters on the right.

Graphic display section

Frequency band section

Output section

Multipressor Graphic Display Section Graphic display: Each frequency band is represented graphically. The amount of gain change from 0 dB is indicated by blue bars. The band number appears in the center of active bands. You can adjust each frequency band independently in the following ways: Drag the horizontal bar up or down to adjust the gain makeup for that band. Drag the vertical edges of a band to the left or right to set the crossover frequencies, which adjusts the bands frequency range. Crossover fields: Set the crossover frequency between adjacent bands. Gain Make-up fields: Set the amount of the gain make-up for each band. Multipressor Frequency Band Section Compr(ession) Thrsh(old) fields: Set the compression threshold for the selected band. Setting the parameter to 0 dB results in no compression of the band. Compr(ession) Ratio fields: Set the compression ratio for the selected band. Setting the parameter to 1:1 results in no compression of the band.

Gain-Q Couple Strength pop-up menu (Extended Parameters area): Choose the amount of Gain-Q coupling. Choose strong to preserve most of the perceived bandwidth. Choose light or medium to allow some change as you raise or lower the gain. The asymmetric settings feature a stronger coupling for negative gain values than for positive values, so the perceived bandwidth is more closely preserved when you cut, rather than boost, gain. Note: If you play back automation of the Q parameter with a different Gain-Q Couple setting, the actual Q values will be different than when the automation was recorded.

Using the Channel EQ

The way you use the Channel EQ is obviously dependent on the audio material and what you intend to do with it, but a useful workflow for many situations is as follows: Set the Channel EQ to a flat response (no frequencies boosted or cut), turn on the Analyzer and play the audio signal. Keep an eye on the graphic display to see which parts of the frequency spectrum have frequent peaks and which parts of the spectrum stay at a low level. Pay particular attention to sections where the signal distorts or clips. Use the graphic display or parameter controls to adjust the frequency bands as desired. You can reduce or eliminate unwanted frequencies, and you can raise quieter frequencies to make them more pronounced. You can adjust the center frequencies of bands 2 through 7 to affect a specific frequencyeither one you want to emphasize, such as the root note of the music, or one you want to eliminate, such as hum or other noise. While doing so, change the Q parameter(s) so that only a narrow range of frequencies are affected, or widen it to alter a broad area. Each EQ band has a different color in the graphic display. You can graphically adjust the frequency of a band by dragging horizontally. Drag vertically to adjust the amount of gain for the band. For bands 1 and 8, the slope values can be changed only in the parameter area below the graphic display. Each band has a pivot point (a small circle on the curve) at the location of the bands frequency; you can adjust the Q or width of the band by dragging the pivot point vertically. You can also adjust the decibel scale of the graphic display by vertically dragging either the left or right edge of the display, where the dB scale is shown, when the Analyzer is not active. When the Analyzer is active, dragging the left edge adjusts the linear dB scale, and dragging the right edge adjusts the Analyzer dB scale. To increase the resolution of the EQ curve display in the most interesting area around the zero line, drag the dB scale, on the left side of the graphic display, upward. Drag downward to decrease the resolution.
Using the Channel EQ Analyzer
The Analyzer, when active, makes uses of a mathematical process called a Fast Fourier Transform (FFT) to provide a real-time curve of all frequency components in the incoming signal. This is superimposed over any EQ curves you have set. The Analyzer curve uses the same scale as the EQ curves, making it easy to recognize important frequencies in the incoming audio. This also simplifies the task of setting EQ curves to raise or lower the levels of frequencies/frequency ranges. The bands derived from FFT analysis are divided in a logarithmic scalethere are more bands in higher octaves than in lower ones. As soon as the Analyzer is activated, you can change the scaling with the Analyzer Top parameter, on the right side of the graphic display. The visible area represents a dynamic range of 60 dB. Drag vertically to set the maximum value to anywhere between +20 dB and 80 dB. The Analyzer display is always dB-linear. Note: When choosing a resolution, be aware that higher resolutions require significantly more processing power. High resolution is necessary when trying to obtain accurate analysis of very low bass frequencies, for example. It is recommended that you disable the Analyzer or close the Channel EQ window after setting the appropriate EQ parameters. This will free up CPU resources for other tasks.

EVOC 20 TrackOscillator Output Parameters
The Output section provides control over the type, stereo width, and level of signal that is sent from the EVOC 20 TrackOscillator.
Signal menu: Determines the signal that is sent to the EVOC 20 TrackOscillator main outputs. You can choose among the following settings: Voc(oder): Choose to hear the vocoder effect. Syn(thesis): Choose to hear only the synthesizer signal. Ana(lysis): Choose to hear only the analysis signal. Note: The last two settings are mainly useful for monitoring purposes. Level slider: Controls the volume of the EVOC 20 TrackOscillator output signal.
Stereo Mode pop-up menu: Sets the input/output mode of the EVOC 20 Filterbank. The choices are m/s (mono input to stereo output), and s/s (stereo input to stereo output). Note: Set Stereo Mode to m/s if the input signal is mono, and to s/s if the input signal is stereo. In s/s mode, the left and right stereo channels are processed by separate filter banks. When you use m/s mode on a stereo input signal, the signal is first summed to mono before it is passed to the filter banks. Stereo Width knob: Distributes the output signals of the Synthesis sections filter bands in the stereo field. At the left position, the outputs of all bands are centered. At the centered position, the outputs of all bands ascend from left to right. At the right position, the bands are outputalternatelyto the left and right channels.

Fuzz-Wah

The Fuzz-Wah plug-in emulates classic wah wah effects often used with a clavinet, and it adds compression and fuzz distortion effects as well. The name wah wah comes from the sound it produces. It has been a popular effectusually a pedal effectwith electric guitarists since the days of Jimi Hendrix. The pedal controls the cutoff frequency of a bandpass, lowpass, orless commonlyhighpass filter.
Getting to Know the Fuzz-Wah Interface
The Fuzz-Wah interface is broken down into the following sections.

Effect Order buttons

Fuzz parameters

Wah parameters

Auto Wah parameters
Effect Order buttons: Select whether the wah wah effect precedes the fuzz effect in the signal chainWah-Fuzzor vice versaFuzz-Wah. See Effect Order Buttons.
Wah parameters: Provide control over the type and tone of the wah wah effect. See Wah Parameters. Auto Wah parameters: Set the depth and envelope times for the automatic wah wah effect. See Auto Wah Parameters. Fuzz parameters: Set the compression ratio, and control the tone and level of the integrated distortion circuit. See Fuzz Parameters.

Using the Direction Mixers Spread Parameter
The Direction Mixers Spread parameter behavior changes when fed LR or MS signals. These differences are outlined below: When working with LR signals, the following applies to the Direction Mixers Spread parameter: At a neutral value of 1, the left side of the signal is positioned precisely to the left and the right side precisely to the right. As you decrease the Spread value, the two sides move toward the center of the stereo image. A value of 0 produces a summed mono signalboth sides of the input signal are routed to the two outputs at the same level. At values greater than 1, the stereo base is extended out to an imaginary point beyond the spatial limits of the speakers. The following applies when working with MS signals: Values of 1 or higher increase the level of the side signal, making it louder than the middle signal. At a value of 2, you hear only the side signal.
Using the Direction Mixers Direction Parameter
When Direction is set to a value of 0, the midpoint of the stereo base in a stereo recording is perfectly centered within the mix. The following applies when working with LR signals: At 90, the center of the stereo base is panned hard left. At 90, the center of the stereo base is panned hard right. Higher values move the center of the stereo base back toward the center of the stereo mix, but this also has the effect of swapping the stereo sides of the recording. For example, at values of 180 or 180, the center of the stereo base is dead center in the mix, but the left and right sides of the recording are swapped. The following applies when working with MS signals: At 90, the middle signal is panned hard left. At 90, the middle signal is panned hard right. Higher values move the middle signal back toward the center of the stereo mix, but this also has the effect of swapping the side signals of the recording. For example, at values of 180 or 180, the middle signal is dead center in the mix, but the left and right sides of the side signal are swapped.
Getting to Know Stereo Miking Techniques
There are three commonly used stereo miking variants used in recording: AB, XY, and MS. A stereo recording, put simply, is one that contains two channel signals. AB and XY recordings both record left and right channel signals, but the middle signal is the result of combining both channels. MS recordings record a real middle signal, but the left and right channels need to be decoded from the side signal, which is the sum of both left and right channel signals. Understanding AB Miking In an AB recording, two microphonescommonly omnidirectional, but any polarity can be usedare equally spaced from the center and pointed directly at the sound source. Spacing between microphones is extremely important for the overall stereo width and perceived positioning of instruments within the stereo field. The AB technique is commonly used for recording one section of an orchestra, such as the string section, or perhaps a small group of vocalists. It is also useful for recording piano or acoustic guitar. AB is not well suited to recording a full orchestra or group as it tends to smear the stereo imaging/positioning of off-center instruments. It is also unsuitable for mixing down to mono, as you run the risk of phase cancellations between channels.

Understanding XY Miking In an XY recording, two directional microphones are symmetrically angled, from the center of the stereo field. The right-hand microphone is aimed at a point between the left side and the center of the sound source. The left-hand microphone is aimed at a point between the right side and the center of the sound source. This results in a 45 to 60 off-axis recording on each channel (or 90 to 120 between channels). XY recordings tend to be balanced in both channels, with good positional information being encoded. It is commonly used for drum recording. XY recording is also suitable for larger ensembles and many individual instruments. Typically, XY recordings have a narrower sound field than AB recordings, so they can lack a sense of perceived width when played back. XY recordings can be mixed down to mono. Understanding MS Miking To make a Middle Side (MS) recording, two microphones are positioned as closely together as possibleusually on a stand or hung from the studio ceiling. One is a cardioid (or omnidirectional) microphone that directly faces the sound source you want to recordin a straight alignment. The other is a bidirectional microphone, with its axes pointing to the left and right of the sound source at 90 angles. The cardioid microphone records the middle signal to one side of a stereo recording. The bidirectional microphone records the side signal to the other side of a stereo recording. MS recordings made in this way can be decoded by the Direction Mixer. When MS recordings are played back, the side signal is used twice: As recorded Panned hard left and phase reversed, panned hard right MS is ideal for all situations where you need to retain absolute mono compatibility. The advantage of MS recordings over XY recordings is that the stereo middle is positioned on the main recording direction (on-axis) of the cardioid microphone. This means that slight fluctuations in frequency response that occur off the on-axisas is the case with every microphoneare less troublesome, because the recording always retains mono compatibility.

Stereo Spread

Stereo Spread is typically used when mastering. There are several ways to extend the stereo base (or perception of space), including use of reverbs or other effects and altering the signals phase. These options can all sound great, but may also weaken the overall sound of your mix by ruining transient responses, for example.
Stereo Spread extends the stereo base by distributing a selectable number of frequency bands from the middle frequency range to the left and right channels. This is done alternatelymiddle frequencies to the left channel, middle frequencies to the right channel, and so on. This greatly increases the perception of stereo width without making the sound totally unnatural, especially when used on mono recordings.

Defining the Pitch Correction Effects Quantization Grid
Use the Pitch Correction effects Normal and Low buttons to determine the pitch range that you want to scan for notes that need correction. Normal is the default range and works for most audio material. Low should be used only for audio material that contains extremely low frequencies (below 100 Hz), which may result in inaccurate pitch detection. These parameters have no effect on the sound; they are simply optimized tracking options for the chosen target pitch range. The Scale pop-up menu allows you to choose different pitch quantization grids. The scale that is set manually (with the keyboard graphic in the plug-in window) is called the User Scale. The default setting is the chromatic scale. If youre unsure of the intervals used in any given scale, choose it in the Scale menu and look at the keyboard graphic. You can alter any note in the chosen scale by clicking the keyboard keys. Any such adjustments overwrite the existing user scale settings. There is only one user scale per project. You can, however, create multiple user scales and save them as Pitch Correction plug-in settings files. Tip: The drone scale uses a fifth as a quantization grid, and the single scale defines a single note. Neither of these scales is meant to result in realistic singing voices, so if youre after interesting effects, you should give them both a try. Open the Root pop-up menu to choose the root note of the scale. (If you chose user scale or chromatic in the Scale pop-up menu, the Root pop-up menu is non-functional.) You may freely transpose the major and minor scales, and scales named after chords.
Excluding Notes from Pitch Correction
You can use the Pitch Correction effects onscreen keyboard to exclude notes from the pitch quantization grid. When you first open the effect, all notes of the chromatic scale are selected. This means that every incoming note will be altered to fit the next semitone step of the chromatic scale. If the intonation of the singer is poor, this might lead to notes being incorrectly identified and corrected to an unwanted pitch. For example, the singer may have intended to sing an E, but the note is actually closer to a D#. If you dont want the D# in the song, the D# key can be disabled on the keyboard. Because the original pitch was sung closer to an E than a D, it will be corrected to an E. Note: The settings are valid for all octave ranges. Individual settings for different octaves arent provided. Use of the small bypass buttons (byp) above the green (black) and below the blue (white) keys excludes notes from correction. This is useful for blue notes. Blue notes are notes that slide between pitches, making the major and minor status of the keys difficult to identify. As you may know, one of the major differences between C minor and C major is the Eb (E flat) and Bb (B flat), instead of the E and B. Blues singers glide between these notes, creating an uncertainty or tension between the scales. Use of the bypass buttons allows you to exclude particular keys from changes, leaving them as they were. If you enable the Bypass All button, the input signal is passed through unprocessed and uncorrected. This is useful for spot corrections to pitch through use of automation. Bypass All is optimized for seamless bypass enabling or disabling in all situations. Tip: Youll often find that its best to correct only the notes with the most harmonic gravity. For example, choose sus4 from the Scale pop-up menu, and set the Root note to match the project key. This will limit correction to the root note, the fourth, and the fifth of the key scale. Activate the bypass buttons for all other notes and only the most important and sensitive notes will be corrected, while all other singing remains untouched.