Soundtrack Pro Effects Reference
K Apple Inc. Copyright © 2007 Apple Inc. All rights reserved. Your rights to the software are governed by the accompanying software license agreement. The owner or authorized user of a valid copy of Soundtrack Pro software may reproduce this publication for the purpose of learning to use such software. No part of this publication may be reproduced or transmitted for commercial purposes, such as selling copies of this publication or for providing paid for support services.
1 Preface 7 7 Contents Introduction to the Soundtrack Pro Plug-ins Soundtrack Pro Effects Chapter 1 9 10 28 29 Delay Delay Designer Stereo Delay Tape Delay Chapter 2 31 32 33 34 34 35 36 Distortion Bitcrusher Clip Distortion Distortion Distortion II Overdrive Phase Distortion Chapter 3 39 41 42 45 46 48 49 50 53 55 Dynamics Adaptive Limiter Compressor DeEsser Enveloper Expander Limiter Multipressor Noise Gate Surround Compressor Chapter 4 59 60 64 66 67 72 73 EQ Channel EQ Fat EQ Linear Phas
Chapter 5 75 76 80 82 Filter AutoFilter Spectral Gate Soundtrack Pro Autofilter Chapter 6 83 83 86 Imaging Direction Mixer Stereo Spread Chapter 7 87 87 88 91 92 Metering Correlation Meter MultiMeter Surround MultiMeter Tuner Chapter 8 95 96 96 97 98 99 101 106 107 Modulation Chorus Ensemble Flanger Modulation Delay Phaser RingShifter Scanner Vibrato Tremolo Chapter 9 109 109 110 Pitch Pitch Shifter II Vocal Transformer Chapter 10 113 114 117 Reverb PlatinumVerb Soundtrack Pro Reverb Ch
Chapter 12 141 141 143 144 Specialized Denoiser Exciter SubBass Chapter 13 147 147 148 149 Utility Gain Multichannel Gain Test Oscillator Contents 5
Preface Introduction to the Soundtrack Pro Plug-ins Soundtrack Pro includes a comprehensive collection of powerful effect plug-ins. This manual will introduce you to the individual effects and their parameters. Using plug-ins is much easier if you are familiar with the basic functions of Soundtrack Pro. Information about these can be found in the Soundtrack Pro User Manual. Soundtrack Pro Effects The following table outlines the effects included with Soundtrack Pro.
Effect category Included effects Filter  AutoFilter (p. 76)  Spectral Gate (p. 80) Imaging  Direction Mixer (p. 83)  Stereo Spread (p. 86) Metering     Correlation Meter (p. 87) MultiMeter (p. 88) Surround MultiMeter (p. 91) Tuner (p. 92) Modulation         Chorus (p. 96) Ensemble (p. 96) Flanger (p. 97) Modulation Delay (p. 98) Phaser (p. 99) RingShifter (p. 101) Scanner Vibrato (p. 106) Tremolo (p. 107) Pitch  Pitch Shifter II (p. 109)  Vocal Transformer (p.
1 Delay 1 Delay effects store the input signal and hold it for a short time before sending it to the effect input or output. Most delays allow you to feed a percentage of the delayed signal back to the input, creating a repeating echo effect. Each subsequent repeat is a little quieter than the previous one. The delay time can often be synchronized to the project tempo by matching the grid resolution of the project, usually in note values or milliseconds.
Delay Designer Delay Designer is a multi-tap delay. Each tap is an independent delay. Unlike simple delay effects that only offer one or two delays (or taps), Delay Designer offers you up to 26 individual taps. In other words, you can think of Delay Designer as 26 separate delay processors—in one effect unit.
The Delay Designer interface consists of five main sections: Sync section Tap display Master section Tap parameter bar Tap pads  Tap display: This blue “view screen” display features a graphic representation of all taps. You can see and edit the parameters of each tap in this area. See “The Tap Display,” next, for a more detailed look.  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.
The Tap Display You can see and interact with taps in the Tap display. The display is divided into a number of sections: Â View buttons: Determine the parameter or parameters represented in the Tap display. Â Autozoom: When engaged, the main display is zoomed out, making all taps visible. You can override Autozoom and manually zoom the Tap display by dragging the Zoom slider. Â Overview display: Shows all the taps in the time range.
The View Buttons The View buttons determine which parameter is represented in the main display. Â Cutoff: When clicked, the taps in the main display show the highpass and lowpass filter cutoff frequencies. Â Reso: When clicked, the main display shows the filter resonance value of each tap. Â Transp: Click to show the pitch transposition of each tap in the main display area. Â Pan: Click to show the pan parameter of each tap in the main display.
m Drag the highlighted bars (that are to the left or right of the bright rectangle) to the left or right. Note: The Autozoom button needs to be turned off for this to work. When you zoom in on a small group of taps, the overview display continues to show all taps. The area shown in the Tap display is indicated by the bright rectangle. To move to different sections of the Tap display: m Drag the bright rectangle to the left or right. The zoomed view in the main display updates as you drag.
To create taps with a Tap pad: 1 Click the upper Start pad. Note: Whenever you click the Start pad, it automatically erases all existing taps. Given this behavior, once you have created your initial taps, you will want to create subsequent taps by clicking the identification bar. The upper pad label changes to Tap, and a red tap recording bar appears in the strip below the View buttons. 2 Click the Tap button to record new taps on the fly.
Identifying Taps Taps are assigned letters based on their order of creation. The first tap to be created is assigned as Tap A, the second tap is assigned as Tap B, and so on. Once assigned, each tap is always identified by the same letter, even as taps are moved in time, and therefore reordered. As an example, if you initially create three taps, they are named Tap A, Tap B, and Tap C. If you then change the delay time of Tap B so that it precedes Tap A, it is still called Tap A.
Selecting Taps There is always at least one selected tap. You can easily distinguish selected taps by color—the toggle bar icons and identification bar letters of selected taps are white. To select a tap, do one of the following: m Click a tap in the main display. m Click the desired tap letter in the identification bar. m Click the downward pointing arrow in the Tap field of the Tap parameter bar, then choose the desired tap letter from the pop-up menu.
Moving Taps You can move a tap backward or forward in time. Note: When you move a tap, you are actually editing its delay time. To move a tap in time: m Select the tap in the identification bar, and drag it forward in time (left) or backward in time (right). Note: Editing the Delay Time parameter in the Tap Delay field of the Tap parameter bar also moves a tap in time. See “The Tap Parameter Bar” on page 18 and “Editing Taps” on page 19 for more details on the Tap Delay field and editing taps.
 Flip: Swaps the left and right side of the stereo or surround image. In other words, clicking this button reverses the tap position from left to right, or vice versa. For example, if a tap is set to 55% left, clicking the flip button will swap it to 55% right.  Pan: The Pan parameter controls the pan position for stereo and the surround angle for surround. The pan parameter displays a percentage between 100% (full left) and –100% (full right), which represents the pan position or balance of the tap.
Editing Parameters in the Tap Display You can graphically edit any tap parameter that is represented as a vertical line in the main Tap display. To edit a tap parameter in the Tap display: 1 Click the View button of the parameter you want to edit. 2 Drag the bright line of the tap you wish to edit up or down (or drag one of the selected taps, if multiple taps are selected). If you have multiple taps selected, the values of all selected taps are increased or decreased relative to other taps.
You can also hold down the Command key and click the Tap display before dragging. This results in a line trailing behind the pointer. The values of the taps are aligned along the line when you release the mouse button. Option-clicking a tap resets the chosen parameter to its default setting. If multiple taps are selected, Option-clicking one tap resets that parameter to its default value for all selected taps.
If the highpass filter’s cutoff frequency value is above that of the lowpass filter cutoff frequency, the filter switches from serial operation to parallel operation, meaning the tap passes through both filters simultaneously. In this case, the space between the two cutoff frequencies represents the frequency band being rejected (in other words, the filters act as a band-reject filter).
Using the Toggle Buttons to Edit Tap Parameters Each tap has its own toggle button in the Toggle bar. These buttons offer you a quick way to graphically activate and deactivate parameters. The specific parameter being toggled by the toggle buttons depends on the current View button selection: Â Â Â Â Â Cutoff view: Toggle buttons turn the filter on or off. Reso view: Toggle buttons switch filter slope between 6 dB and 12 dB. Pitch view: Toggle buttons switch pitch transposition on or off.
Parameter Editing Suggestions In general, you’ll find editing in the Tap parameter bar fast and precise when you want to edit the parameters of one tap at a time. All parameters of the selected tap are available, with no need to switch display views or estimate values with vertical lines. If you want to edit the parameters of one tap relative to other taps, use the Tap display. Also, if you want to edit multiple taps at once, you can use the Tap display to select multiple taps and then edit them together.
Setting the Grid Resolution The Grid menu offers several grid resolutions, which correspond to musical note durations. The grid resolution, along with the project tempo, determines the length of each grid increment. To set the grid resolution: m Click the Grid field, then choose the desired grid resolution from the pop-up menu. As you ch ange grid resolutions, the increments shown in the identification bar change accordingly. This also determines a step limitation for all taps.
Saving Sync Settings When you save a Delay Designer setting, the Sync mode status, Grid, and Swing values are all saved. When you save a setting with Sync mode on, the grid position of each tap is also stored. This ensures that a setting loaded into a project with a different tempo (to that of the project that the setting was created in) will retain the relative positions and rhythm of all taps—at the new tempo.
To toggle feedback on or off: m Click the Feedback button. When the Feedback button is turned on, it is lit. The orange track around the Feedback Level knob indicates the current feedback level. Note: If feedback is turned on and you begin creating taps using the Tap pads, feedback is automatically switched off. When you stop creating taps with the Tap pads by clicking the Last Tap button, feedback is automatically turned back on.
Delay Designer always processes each input channel independently. In surround configurations, Delay Designer processes each surround channel independently, and the surround panner lets you adjust the surround balance of each tap in the surround field. Note: The Delay Designer generates separate automation data for stereo pan and surround pan operations. This means that when using the Delay Designer in surround channels, it will not react to existing stereo pan automation data, and vice versa.
 Crossfeed Left to Right and Crossfeed Right to Left: Use to transfer the feedback signal of the left channel to the right channel, and vice versa.  Crossfeed Phase buttons: Use to invert the phase of the crossfed feedback signals. Tape Delay The Tape Delay simulates the warm sound of vintage tape echo machines, with the convenience of easy delay time synchronization to your project tempo.
 LFO Speed: Sets the frequency (speed) of the LFO.  LFO Depth: Sets the amount of LFO modulation. A value of 0 turns delay modulation off.  Flutter parameters: Simulates the speed irregularities of the tape transports used in analog tape delay units. Flutter Rate adjusts the speed, and Flutter Intensity determines how pronounced the effect is.  Smooth: Evens out the LFO and flutter effect.  Dry and Wet: These individually control the amount of original and effect signal.
2 Distortion 2 You can use Distortion effects to recreate the sound of analog or digital distortion, and to radically transform your audio. Distortion effects simulate the distortion created by vacuum tubes, transistors, or digital circuits. Vacuum tubes were used in audio amplifiers before the development of digital audio technology, and are still used in musical instrument amps today.
Bitcrusher The Bitcrusher is a low-resolution digital distortion effect. You can use it to emulate the sound of early digital audio, create artificial aliasing by dividing the sample rate, or distort signals until they are unrecognizable. Bitcrusher Parameters  Drive slider and field: Sets the amount of gain (in decibels) applied to the input signal.  Resolution slider and field: Sets the bit rate (between 1 and 24 bits).
Clip Distortion Clip Distortion is a nonlinear distortion effect that produces unpredictable spectra. You can use it to simulate warm, overdriven tube sounds, and also to create drastic distortion. Clip Distortion features an unusual combination of serially connected filters. After being amplified by the Drive value, the signal passes through a highpass filter, and is then subjected to nonlinear distortion, as controlled by the Symmetry parameter.
Using Clip Distortion If you set the High Shelving Frequency to around 12 kHz, you can use it like the treble control on a mixer channel strip or a stereo hi-fi amplifier. Unlike those types of treble controls, however, you can boost or cut the signal by up to ±30 dB using the Gain parameter. Distortion This Distortion effect simulates the lo-fi, dirty distortion generated by a bipolar transistor.
Distortion II Parameters  PreGain dial: Sets the amount of gain applied to the input signal.  Drive dial: Sets the amount of saturation applied to the signal.  Tone dial: Sets the frequency at which the signal is filtered. Filtering the harmonically rich distorted signal produces a somewhat less grating, softer tone.  Type pop-up menu: Choose the type of distortion you want to apply. The choices are: Growl, Bity, and Nasty.
Phase Distortion The Phase Distortion effect is based on a modulated delay line, similar to a chorus or flanger effect (for more information about these effects, see Chapter 8, “Modulation,” on page 95). Unlike these effects, however, in the Phase Distortion effect the delay time is not modulated by a low frequency oscillator (LFO), but rather by a lowpass-filtered version of the input signal itself. This means that the signal modulates its own phase position.
Using the Phase Distortion The input signal only passes the delay line and is not affected by any other process. The Mix parameter blends the effected signal with the original signal. The delay time is modulated by a side chain signal—namely, the input signal. The input signal passes through a resonant lowpass filter, with dedicated Cutoff frequency and Resonance controls. You can listen to the filtered side chain (instead of the Mix signal) by turning on the Monitor button.
3 Dynamics 3 You can use Dynamics effects to control the perceived loudness of your audio, add focus and punch to tracks and projects, and optimize the sound for playback in different situations. The dynamic range of an audio signal is the range between the softest and loudest parts of the signal (technically, between the lowest and the highest amplitude).
Some compressors, called multiband compressors, can divide the incoming signal into different frequency bands, and apply different compression settings to each band. This helps achieve the maximum level without introducing compression artifacts, and is typically used on an overall project mix. Expanders Expanders are similar to compressors, except that they raise, rather than lower, the signal when it exceeds the threshold. Expanders are used to enliven the audio signal.
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 clipping (exceeding 0 dBFS).
Compressor The Compressor is designed to emulate the sound and response of a professional-level analog (hardware) compressor. It tightens up your audio by reducing sounds that exceed a certain threshold level, smoothing out the dynamics and increasing the overall volume—the perceived loudness. Compression helps bring the key parts of a track or a mix into focus while preventing softer parts from being inaudible. It is probably the most versatile and widely used sound-shaping tool used in mixing, next to EQ.
 Peak/RMS buttons: Turn on one or the other to set whether the Compressor analyzes the signal using the Peak or RMS method when using the Platinum Circuit Type.  Gain slider and field: Sets the amount of gain applied to the output signal.  Gain pop-up menu: Choose a value to raise the output level in order to compensate for volume reduction caused by compression. The choices are OFF, 0 dB, and –12 dB.  Limiter Threshold slider and field: Sets the threshold level for the limiter.
Knee The Knee parameter smooths the effect of the Compressor by controlling whether the signal is slightly compressed as it approaches the threshold. Setting the Knee parameter close to 0 (zero) means that levels just below the threshold are not compressed at all (1:1 ratio), while levels at the threshold are compressed by the full Ratio amount (4:1, 10:1, or more). This is what audio engineers call hard knee compression, which can cause the transition to be abrupt as the signal reaches the threshold.
DeEsser The DeEsser is a frequency-specific compressor, designed to compress only a particular frequency band within a complex audio signal. It is used to eliminate hiss (also called sibilance) from the signal. The advantage of using the DeEsser instead of an EQ effect to cut high frequencies is that it compresses the signal dynamically rather than statically. This prevents the sound from becoming darker when no sibilance is present in the signal.
Suppressor Section  Suppressor Frequency knob: Sets the frequency band that is reduced when the Detector frequency sensitivity threshold is exceeded.  Strength knob: Sets the amount of gain reduction around the Suppressor frequency. Center Section  Detector and Suppressor frequency displays: The upper display shows the Detector frequency range, and the lower display shows the Suppressor frequency range (in Hz).  Smoothing slider: Sets the reaction speed of the gain reduction start and end phases.
Using the Enveloper The most important parameters of the Enveloper are the two Gain sliders, one on each side of the central display area, that govern Attack (left) and Release (right). Raising the gain emphasizes the attack or release phase, respectively, while lowering the gain attenuates the corresponding phase. For example, boosting the attack gives a drum sound more snap, or amplifies the initial pluck (or pick) sound of a stringed instrument.
Expander The Expander is similar to a compressor except that it increases, rather than reduces, the dynamic range above the Threshold level. You can use the Expander to add liveliness and freshness to your audio, specifically by emphasizing the transients of highly compressed signals. Expander Parameters  Threshold slider and field: Sets the level above which the Expander expands the signal.  Ratio slider and field: Sets the ratio by which the signal is expanded when it exceeds the threshold.
Limiter The Limiter functions similarly to a compressor with one important difference: where a compressor proportionally reduces the signal when it exceeds the threshold, a limiter reduces any peak above the threshold to the threshold level, effectively limiting the signal to this level. The Limiter is used primarily as a mastering effect. Limiter Parameters  Gain reduction meter: Shows the amount of limiting while the signal plays.
Multipressor The Multipressor (short for multiband compressor) is an extremely versatile tool used in mastering audio. It splits the incoming signal into different frequency bands (from one to four), and allows you to apply compression to each band independently. After compression is applied, the bands are combined into a single output signal.
Graphic Display Section  Graphic display: Each frequency band is represented graphically. The amount of gain change from 0 dB is shown graphically by the blue bars. For active bands, the band number appears in the center of its area. 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.
Output Parameters  Auto Gain pop-up menu: Controls whether the Multipressor references the overall processing of the signal to 0 dB, making the output louder (On), or produces more standard compression, with compressed bands attenuated by the amount the dynamic range is reduced (Off ).  Lookahead value field: Adjusts how far the processor looks forward in the audio, in order to react earlier to peak volumes for smoother transitions.  Out Gain slider: Sets the overall gain at the output.
Noise Gate The Noise Gate is commonly used to suppress unwanted noise that is audible when the audio signal is at a low level. You can use it to remove background noise, crosstalk from other signal sources, and low-level hum. The Noise Gate works by allowing signals above the Threshold level to pass unimpeded while reducing signals below the Threshold level. This allows you to remove lower-level parts of the signal, while allowing the intended parts of the audio to pass.
Using the Noise Gate In most situations, setting the Reduction slider to the lowest possible value ensures that sounds below the Threshold are completely suppressed. Setting it to a higher value attenuates low-level sounds but still allows them to pass. You can also set Reduction to a value greater than 0 (zero) to boost the signal by up to 20 dB. This is useful for ducking effects. The three rotary knobs for Attack, Hold, and Release modify the dynamic response of the Noise Gate.
Surround Compressor The Surround Compressor, based on the Compressor, is specially adapted for compressing complete surround mixes. The Surround Compressor is especially useful when inserted in a surround output or on channels and busses that carry multichannel audio. You can adjust the compression ration, knee, attack, and release for both the main channels and for the LFE channel. Both the main channels and the LFE channel include an integrated limiter.
Link Section  Grp. (Group) pop-up menus: For each channel, choose whether the channel is in group A, B, or C, or in no group (–). Moving the Threshold or Output Level slider for any channel assigned to a group will move the sliders for all channels assigned to that group.  Byp (Bypass) buttons: For each channel, click to bypass that channel. Main Section  Ratio slider and field: Sets the ratio by which the signal is reduced when it exceeds the threshold.
Using the Surround Compressor Using the Link controls, you can assign each channel independently to one of three groups (Group A, B, or C). When you adjust the Threshold or Output Level slider for any channel in a group, the sliders for all channels in the same group are adjusted by the same amount. Also, clicking the Bypass button for any grouped channel bypasses all channels in the group.
4 EQ 4 Equalization (EQ) lets you shape the sound of your audio by changing the level of specific frequency bands. EQ is one of the most commonly used audio effects, both for music projects and in post-production work for video. You can use EQ to shape the sound of an audio file, track, or project by adjusting specific frequencies or frequency ranges. Using EQ, you can create both subtle and extreme changes to the sound of your projects.
Multiband EQs Multiband EQs give you control over a set of filters that together cover a large part of the frequency spectrum. On multiband EQs, you can set the frequency, bandwidth, and Q of each band independently. Using a multiband EQ (such as the Channel EQ, Fat EQ, or Linear Phase EQ), you can perform extensive tone-shaping on any audio source. Multiband EQs are equally useful for shaping the sound of an individual track or an overall project mix.
Channel EQ Parameters On the left side of the Channel EQ window is the Gain control and parameters for the Analyzer, while the central area of the window includes the graphic display and parameters for shaping each EQ band. Â Master Gain slider and field: Sets the output level of the signal. After boosting or cutting individual frequency bands, you can use the Master Gain fader to adjust the output level. Â Analyzer button: Turns the Analyzer on or off.
 Link button: Activates Gain-Q coupling, which automatically adjusts the Q (bandwidth) when you raise or lower the gain on any EQ band, to preserve the perceived bandwidth of the bell curve. Setting Gain-Q Couple to strong preserves the perceived bandwidth almost entirely, while light and medium settings allow some change as you raise or lower the gain.
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. The overall range is always ±30, but small values are easier to recognize. When you work with the Channel EQ, you can turn off any bands you are not using to shape the sound. Inactive bands do not use any computer resources.
Fat EQ The Fat EQ effect is a versatile multiband EQ with up to five individual frequency bands. You can use Fat EQ for individual tracks or for overall mixes. The Fat EQ includes a graphic display of the EQ curves and a set of parameters for each band. Fat EQ Parameters The main area of the Fat EQ window includes a graphic display area and a set of strips with parameters for each frequency band. To the right of the parameter section are the Master Gain slider and field.
Parameter Section Below the graphic display area are controls that you can use to show the settings for each band and adjust each band’s settings. Â Frequency fields: Sets the frequency for each band. Â Gain knobs: Sets the amount of gain for each band. Â Q/Order fields: Sets the Q or bandwidth for each band (the range of frequencies around the center frequency that are altered). For bands 1 and 5, this changes the slope of the filter.
Linear Phase EQ The high-quality Linear Phase EQ effect is similar in appearance to the Channel EQ, sharing the same parameters and eight-band layout. However, the Linear Phase EQ uses a different underlying technology, which preserves the phase of the audio signal 100%—even when you apply the wildest EQ curves to the sharpest signal transients! The Linear Phase EQ uses more CPU resources than the Channel EQ, and introduces greater amounts of latency.
Match EQ The Match EQ effect allows you to store the average frequency spectrum of an audio file as a template and apply the template to your project, so that it matches the spectrum of the original file. Using Match EQ you can acoustically match the sound of different songs you plan to include on an album, or impart the particular sound of any source recording to your own projects.
 Format button: Sets whether the Analyzer displays audio channels via separate curves (L&R for stereo, All Cha for surround) or the summed maximum level (LR Max for stereo, Cha Max for surround).  Select buttons: Click one of the buttons to control whether any changes you make to the filter curve created by matching the template with the current material are applied only to the left channel (L), the right channel (R), or both channels (L+R).
 Smoothing slider and field: Sets the amount of smoothing for the filter curve. A value of 0.0 leaves the filter curve unchanged. At all other settings, the filter curve is smoothed at a constant bandwidth, determined by the set value in semitones. For example, a value of 1.0 means that the smoothing uses a bandwidth of one semitone. A value of 4.0 produces a smoothing bandwidth of four semitones (a major third), a value of 12.0 produces a bandwidth of one octave, and so on.
To use the matched EQ on a track: 1 Set the track you want to match as a sidechain input to the Match EQ. 2 Click the “Template Learn” button, play the entire source audio track from start to finish, then click the “Template Learn” button again. 3 Return to the start of your mix, click Current Material Learn, then play your mix (the current material) from start to finish. 4 When done, click Current Material Match (this automatically disengages the Current Material Learn button).
Only one of the Learn buttons can be active at a time. For example, if the Learn button in the Template section is active and you press the Learn button in the Current Material section, the analysis of the template file stops, the current status is used as the spectral template, and analysis of the track (Current Material) begins.
Single Band EQs Following are descriptions of each of the effects found in the Single Band submenu. High Cut and Low Cut Filter As their names suggest, the Low Cut Filter attenuates the frequency range below the selected frequency, while the High Cut Filter attenuates the frequency range above the selected frequency. Each has a single parameter letting you set the cutoff frequency. High Pass and Low Pass Filter The High Pass Filter affects the frequency range below the set frequency.
Frequency Ranges Used With EQ All sounds can be thought of as falling into one of three basic frequency ranges: bass, midrange, or high (treble). These can each be further divided to include low bass, low and high midrange, and low and high highs. The following table describes some of the sounds that fall into each range: Name Frequency range Description High High 8–20 kHz Includes cymbal sounds and highest harmonics of instruments.
5 Filter 5 In addition to the filters in EQ effects, you can use filters to change the character of your audio in both familiar and unusual ways. The Filter submenu contains a variety of filter-based effects that you can use to creatively modify your audio, including autofilters, filter banks, vocoders, wah-wah effects, and a gate that uses frequency rather than the amplitude (volume) as the criteria for which part of the signal is allowed to pass through.
AutoFilter The AutoFilter is a versatile filter effect with several unique features. You can use it to create classic, analog-style synthesizer effects, or as a tool for creative sound design. The filter cutoff can be dynamically modulated using both a synthesizer-style ADSR envelope and a low frequency oscillator (LFO).
LFO Section  Coarse and Fine Rate knobs and field: Use together to set the frequency of the LFO. Drag the Coarse slider to set the LFO frequency in Hertz, then drag the Fine slider to fine tune the frequency in 1/000s of a Hertz.  Beat Sync button: When selected, the LFO is synchronized to the sequencer’s tempo.  Phase knob: Lets you shift the phase relationship between the LFO and the sequencer when Beat Sync is active.
Using the AutoFilter The following section provides additional information on using the parameters in the AutoFilter window. Filter Parameters The most important parameters are located on the right side of the AutoFilter window. The Filter Cutoff knob determines the point where the filter kicks in. Higher frequencies are attenuated, while lower frequencies are allowed to pass through. The Resonance knob controls how much frequencies around the cutoff frequency are emphasized.
LFO Parameters You set the waveform of the LFO by clicking one of the Waveform buttons. The choices are: descending sawtooth (saw down), ascending sawtooth (saw up), triangle, pulse wave, or random (random values, Sample & Hold). Once you select a waveform, you can shape the curve with the Pulsewidth slider. Use the Coarse and Fine Frequency knobs to set the LFO frequency. The Rate Mod. (Rate Modulation) knob controls modulation of the LFO frequency independent of the input signal level.
Spectral Gate The Spectral Gate separates the signal above and below the Threshold level into two independent frequency ranges that you can modulate separately. It can produce some unusual and rich filtering effects. Spectral Gate Parameters  Threshold slider and field: Sets the threshold level at which the frequency band defined by the Center Freq. and Bandwidth parameters is divided into upper and lower frequency ranges.
Using the Spectral Gate Using the Center Freq. and Bandwidth parameters, set the frequency band you want to process using the Spectral Gate. The graphic display visually indicates the band defined by these two parameters. Once the frequency band is defined, use the Threshold parameter to set the level above and below which the frequency band is divided into upper and lower ranges.
Soundtrack Pro Autofilter The Soundtrack Pro Autofilter is a simple lowpass resonance filter that offers the following parameters: Â Cutoff Frequency slider and field: Sets the cutoff frequency for the lowpass filter. Â Resonance slider and field: Adjusts the amount of emphasis in the frequency band around the cutoff frequency.
6 Imaging 6 You can use the Soundtrack Pro Imaging plug-ins to extend the stereo base of a recording, and to alter perceived signal positions. These effects enable you to make certain sounds, or the overall mix, seem wider and more spacious. You can also alter the phase of individual sounds within a mix, to enhance or suppress particular transients.
Using the Direction Mixer The Direction Mixer is a simple plug-in to use, as it only offers two parameters: Spread and Direction. Each alters the incoming signal differently when either the LR or MS Input buttons are active. Using the Spread Parameter on LR Input Signals At a neutral value of 1, the left side of the signal is positioned precisely on the left, and the right side precisely on the right. As you decrease the Spread value, the two sides move toward the center of the stereo image.
What is MS? Relegated to obscurity for a good long while, MS stereo (middle-side as opposed to left-right) has recently enjoyed a renaissance of sorts. Making a Middle Side Recording Two microphones are positioned as closely together as possible (usually on a stand or hung from the studio ceiling). One is a cardioid (or omnidirectional) microphone that directly faces the sound source that you want to record—in a straight alignment.
Stereo Spread The Stereo Spread effect is typically used for mastering. There are several ways to extend the stereo base (or perception of space), including use of reverbs and other effects and altering the signal’s phase. They can all sound great, but can also weaken the overall sound of your mix by ruining transient responses, for example. The Stereo Spread plug-in extends the stereo base by distributing a selectable number of frequency bands from the middle frequency range to the left and right channels.
7 Metering 7 You can use the Metering plug-ins of Soundtrack Pro to analyze audio in a variety of ways. Each Metering plug-in allows you to view different characteristics of an audio signal. For example: The BPM Counter displays the pitch of a note, the Correlation Meter displays the phase relationship, and the Level Meter displays the level of an audio recording.
MultiMeter The MultiMeter provides a collection of professional gauge and analysis tools in a single window. It includes: Â An Analyzer to view the level of each 1/3-octave frequency band. Â A Goniometer for judging the phase coherency in the stereo sound field. Â a Correlation Meter to spot mono phase compatibility. Â An integrated Level Meter to view the signal level for each channel. There is also a surround version of the MultiMeter, with parameters for each channel and a slightly different layout.
Goniometer Section  Goniometer button: When selected, displays the Goniometer in the center of the window.  Auto Gain field (display only): Sets the amount by which the display compensates for low input levels. You can set Auto Gain levels in 10% increments, or set it to off.  Decay field: Sets the amount of time it takes for the Goniometer trace to fade to black. Peak Section  Hold button: When selected, activates peak hold for the all of the metering tools in the MultiMeter.
You can alter the scale of values displayed in the Analyzer in several ways. Use the View parameters, which let you set the maximum level displayed and the overall dynamic range, by vertically dragging the dB scale on the left edge of the Analyzer. Adjusting the scale is useful when analyzing highly compressed material, so that you can identify smaller level differences more easily by moving and/or reducing the display range.
Level Meter (Peak/RMS Meter) The Level Meter displays the current signal level on a logarithmic decibel scale. The signal level for each channel is represented by a blue bar. RMS and Peak levels are shown simultaneously, with RMS levels appearing as dark blue bars, and Peak levels appearing as light blue bars. When the level exceeds 0 dB, the portion of the bar above the 0 dB point becomes red. The current peak values are displayed numerically (in dB increments) above the Level Meter.
Balance/Correlation Section In the Surround MultiMeter, the Correlation Meter is not visible when the Analyzer or Goniometer are active, but is displayed separately when you select the Balance/Correlation button. The highlighted area indicates the overall balance of the surround signal. The Balance/Correlation Meter combines two meters into one compact, easy-to-read display.
 Tuning Adjustment knob and field: Sets the pitch of the note used as the basis for tuning. By default, the Tuner is set to concert pitch A = 440 Hz. Drag the knob left to lower the pitch corresponding to A, or drag the knob right to raise the pitch corresponding to A. The current value is displayed in the field. Using the Tuner Using the Tuner is simple.
8 Modulation 8 Modulation effects are used to add motion and depth to your sound. Modulation effects include chorus, flanging, and phasing among others, which make sounds richer or more animated. This is often achieved through the use of an LFO, which is controlled with parameters such as speed or frequency, and depth (also called width, amount, or intensity). You can also control the ratio of the affected (wet) signal and the original (dry) signal.
Chorus The Chorus effect delays the original signal. The delay time is modulated with an LFO. The delayed, modulated signal is mixed with the original, dry signal. You can use the Chorus effect to enrich the sound and create the impression that it’s being played by multiple instruments or voices, in unison. The slight delay time variations generated by the LFO simulate the subtle pitch and timing differences heard when several people perform together.
 Phase knob and field: Controls the phase relationship between the individual voice modulations. The value that you choose here is dependent on the number of voices, which is why it is shown as a percentage value rather than degrees. The value 100 (or –100) is equal to the greatest possible distance between the modulation phases of all voices.  Spread slider and field: Used to distribute the voices across the stereo or surround field.
Modulation Delay The Modulation Delay effect is based on the same principles as the Flanger and Chorus effects, but you can set the delay time, thereby allowing both chorus and flanging effects to be generated. It can also be used—without modulation—to create resonator or doubling effects. The modulation section consists of two LFOs with variable frequencies. Â Feedback slider and field: Determines the amount of the effect signal that is routed back to the input.
 LFO Phase knob and field (only available in stereo and surround instances): Controls the phase relationship between the individual channel modulations. At 0°, the extreme values of the modulation are achieved simultaneously for all channels. 180° or –180° is equal to the greatest possible distance between the modulation phases of the channels. The LFO Phase parameter is available only if the LFO Left Right Link button is active.
 Order slider and field: Allows you to choose between different phaser algorithms. The more orders a phaser has, the heavier the effect.  Env Follow slider and field (Sweep section): Determines how much the frequency range (as set with the Ceiling and Floor controls) is modulated by the level of the input signal.  LFO 1 and LFO 2 Rate knobs and fields: Use to set the speed for each LFO independently.  LFO Mix slider and fields: Determines the balance between the two LFOs.
RingShifter The RingShifter effect combines a ring modulator with a frequency shifter effect. Both effects were popular during the 1970s, and are currently experiencing something of a renaissance. Â The ring modulator modulates the amplitude of the input signal using either the internal oscillator or a side chain signal. The frequency spectrum of the resulting effect signal equals the sum and difference of the frequency content in the two original signals.
Modes The four mode buttons determine whether the RingShifter operates as a frequency shifter or as a ring modulator. Â Single (Frequency Shifter) button: The frequency shifter generates a single, shifted effect signal. The oscillator Frequency control determines whether the signal is shifted up (positive value) or down (negative value). Â Dual (Frequency Shifter) button: The frequency shifting process produces one shifted effect signal for each stereo channel—one is shifted up, the other is shifted down.
 Frequency control: Sets the frequency of the sine oscillator.  Lin(ear) and Exp(onential) buttons: Use these buttons to switch the scaling of the Frequency control:  The exponential scaling offers extremely small increments around the 0 point, which is useful for programming slow moving phasing and tremolo effects.  In the Lin(ear) mode, the resolution of the scale is even across the entire control range.
Output  Feedback knob and field: Sets the amount of the signal that is routed back to the effect input.  Stereo Width knob and field: Determines the breadth of the effect signal in the stereo field. Stereo Width affects only the effect signal of the RingShifter, not the dry input signal.  Dry/Wet knob and field: Set the mix ratio of the dry input signal and the wet effect signal.
Modulation Sources The oscillator Frequency and Dry/Wet parameters can be modulated via the internal envelope follower and LFO. The oscillator frequency even allows modulation through the 0 Hz point, thus changing the oscillation direction. Envelope Follower The envelope follower analyzes the amplitude (volume) of the input signal and uses this to create a continuously changing control signal—a dynamic volume envelope of the input signal. This control signal can be used for modulation purposes.
Scanner Vibrato The Scanner Vibrato effect simulates the scanner vibrato section of a Hammond organ. You can choose between three different vibrato and chorus types. The stereo version of the effect features two additional parameters: Stereo Phase and Rate Right. These allow you to set the modulation speed independently for the left and right channels. The stereo parameters of the mono version of the Scanner Vibrato are hidden behind a transparent cover.
Tremolo The Tremolo effect modulates the amplitude of a signal, resulting in periodic volume changes. You’ll recognize this effect from vintage guitar combo amps (where it is sometimes incorrectly referred to as vibrato). The graphic display shows all parameters, except Rate. Â Â Â Â Depth slider and field: Determines the modulation amount. Rate knob and field: Defines the frequency, and therefore the speed, of the LFO. Symmetry and Smoothing knobs and fields: Use these to set the shape of the modulation.
9 Pitch 9 You can use the Pitch effects of Soundtrack Pro to transpose the pitch of audio tracks. These effects can also be used for creating unison or slightly thickened parts, or even the creation of harmony voices. Soundtrack Pro includes the following Pitch effects: Â “Pitch Shifter II” on page 109 Â “Vocal Transformer” on page 110 Pitch Shifter II The Pitch Shifter II provides a simple way to combine a pitch-shifted version of the signal with the original signal.
 Drums, Speech, and Vocals buttons: Select one of the three presets to optimize Pitch Shifter II operation for common types of audio material:  Drums leaves the groove of the original track intact.  Vocals retains the intonation of the original with no change. Thus, Vocals is well-suited for any signals that are inherently harmonic or melodious, such as string pads.  Speech provides a compromise between the two by attempting to retain both the rhythmic and harmonic aspects of the signal.
Vocal Transformer Parameters  Pitch knob and field: Determines the amount of transposition applied to the input signal.  Formant knob and field: Shifts the formants of the input signal.  Robotize button: Click to switch the Vocal Transformer to Robotize mode. Robotize mode is used for augmenting, diminishing, or mirroring the melody.  Tracking slider and buttons (available only in Robotize mode): Control how the melody is changed in Robotize mode.
∏ Tip: If you set Pitch to 0 semitones, Mix to 50%, and Formant to +1 (with Robotize switched off ), you can effectively place a singer (with these different vocal characteristics) next to the original singer. Both will sing with the same voice—in a choir of two. This choir effect is quite effective, and is easily controlled with the Mix parameter. Using Robotize Mode If you switch Robotize on, the Vocal Transformer can augment or diminish the melody.
10 10 Reverb You can use Reverb effects to simulate the sound of acoustic environments such as rooms, concert halls, caverns, or the sound of infinite space. Sounds bounce off the surfaces of any space, or off objects within a space, repeatedly, gradually dying out until they are inaudible. The bouncing soundwaves result in a reflection pattern, more commonly known as a reverberation (or reverb).
Plates, Digital Reverb Effects, and Convolution Reverb The first form of reverb used in music production was actually a special room with hard surfaces (called an echo chamber). It was used to add echoes to the signal. Mechanical devices, including plates and springs, were used to add reverberation to the output of musical instruments and microphones. Digital recording introduced digital reverb effects, which consist of thousands of delays of varying lengths and intensities.
The interface can be divided into four parameter groups: Â Early Reflections parameters: Emulates the original signal’s first reflections as they bounce off the walls, ceiling, and floor of a natural room. Â Reverb parameters: Controls the diffuse reverberations. Â Balance ER/Reverb parameter: Controls the balance between the Early Reflections and Reverb section. When you set the slider to either of its extreme positions, the unused section is deactivated.
Output Parameters  Dry: Controls the amount of the original signal.  Wet: Controls the amount of the effect signal. Setting Predelay and Initial Delay In practice, too short a Predelay tends to make it difficult to pinpoint the position of the signal. It can also color the sound of the original signal. On the other hand, too long a Predelay can be perceived as an unnatural echo. It can also divorce the original signal from its early reflections, which leaves an audible gap.
Setting the Reverb Time and Level of the Low Frequency Band You can use the Low Ratio control to offset the reverb time of the low frequency band. At 100%, the reverb times for the two bands are identical. At lower values, the reverb time of the frequencies below the crossover frequency is shorter. At values greater than 100%, the reverb time for low frequencies is longer. Both of these phenomena occur in nature. In most mixes, a shorter reverb time for bass frequencies is preferable.
11 Convolution Reverb: Space Designer 11 Space Designer is a convolution reverb effect. You can use it to create exceptionally realistic reverberations. Space Designer generates reverb by convolving, or combining, an audio signal with an impulse response (IR) reverb sample. For example, imagine that you apply the Space Designer to a vocal track.
Impulse response parameters Envelope and EQ display Filter parameters Volume envelope parameters Space Designer consists of the following parameter groups: Â Impulse response parameters: Use these parameters to load, save, or manipulate impulse response files. The IR file you choose determines what Space Designer will use to convolve with your audio signal. These parameters will be the initial parameters you use to load your IR file, as well as the last if you want to save your synthesized IR.
 Synthesized impulse response parameters: If you do enough processing of the original IR, you may want to synthesize a new IR from your edited parameters. Use these parameters to adjust the density envelope and other synthesized IR parameters. See “Synthesizer Impulse Response Parameters” on page 136.  EQ: For final sound sculpting, Space Designer includes a built-in four-band EQ: two shelving filters, and two parametric filters. Use these parameters to fine tune the sound of your reverb to your taste.
 Length parameter: Adjusts the length of the impulse response.  Synthesized IR button: Click to switch Space Designer to Synthesized IR mode. In this mode, Space Designer generates a new synthesized impulse response from the values of the Length, envelope, filter, EQ, and Spread parameters. You may freely switch between a loaded impulse response sample and a synthesized impulse response without losing the settings of the other.
Setting the Sample Rate The Sample Rate slider is used to determine the sample rate of an impulse response. You can choose between the following settings: Â Orig: Space Designer uses the current project sample rate. When loading an impulse response, Space Designer automatically converts the sample rate of the impulse response to match the current project sample rate—should it be necessary. For example, this allows you to load a 44.1 kHz impulse response into a project running at 96 kHz, and vice versa.
Activating the Preserve Length button preserves the length of the impulse response when the sample rate is changed. Manipulating these two parameters as you see fit can lead to interesting results. The lower sample rates can also be used for interesting tempo, pitch, and retro-digital sounding effects. If running Space Designer in a project that uses a higher sample rate than the impulse response, you may also want to reduce the impulse response sample rate.
Global Parameters Space Designer’s Envelope and EQ display contains most of Space Designer’s interface elements that change to reflect the current parameter group you are adjusting. The global parameters, spread throughout the interface around and below the Envelope and EQ display, remain constant.
The lower (flat) section of Space Designer contains the following global parameters: Â Filter parameters: Activate, adjust the resonance, and select the mode of Space Designer’s resonant filter. See “Filter Parameters” for more information. Â Pre-Dly knob: Sets the reverb’s predelay time, or time between the original signal and the first reflections from the reverb. See “Predelay” on page 130 for more information. Â IR Start knob: Shifts the point at which the impulse response will play back.
Surround Mode For surround instances of Space Designer, the Input slider determines how much of the LFE signal is mixed with the surround channels feeding the reverb. At its lowest setting, the slider acts as an LFE bypass, with all the LFE signal passed through the reverb unprocessed. Latency Compensation The complex calculations made by Space Designer take time. This time results in a processing latency, or delay, between the direct (input) signal, and the processed (output) signal.
Rev Vol Compensation Rev Vol Compensation (Reverb Volume Compensation) attempts to match the perceived (not actual) volume differences of impulse response files. It is switched on by default and should generally be left in this mode, although you may find that it isn’t successful with all types of impulse responses. In such situations, switch it off and adjust the input and output levels accordingly.
Output Parameters The output parameters let you adjust the mix between the direct (dry) and processed signals. Which parameters are available depends on Space Designer’s input configuration. Mono and Stereo Configurations If you insert Space Designer as mono, mono-to-stereo, or stereo effect, Space Designer offers two output sliders: one for the direct signal, and one for the reverb signal. Â Dry slider: Sets the level of the noneffected (dry) signal.
Predelay Predelay is the amount of time that elapses between the original signal and the initial early reflections from the reverb generated by Space Designer. For a room of any given size and shape, predelay determines the distance between the listener and the walls, ceiling, and floor. Of course, Space Designer allows you to adjust this parameter separately from, and over a greater range than what is considered natural for predelay.
Envelope and EQ Display Space Designer’s Envelope and EQ display features two components: the button bar at the top and the main display (including its parameter bar). The display itself shows either the envelope being edited or the EQ curve, depending on which button you engage.
 EQ button: Click this button to switch the main display to Space Designer’s four-band parametric EQ. See “EQ Parameters” on page 138 for more information on Space Designer’s EQ.  Reverse button: Click to reverse the impulse response together with its envelopes. When you reverse the impulse response, you are effectively using the tail rather than the front end of the sample. As such, you may need to use lower or even negative Predelay values when reversing.
 You can change the curve shape by dragging the envelope curve directly. Use the small nodes attached to a line for finer adjustments to envelope curves. They are tied to the envelope curve itself, so you can view them as envelope handles. Moving the nodes vertically or horizontally will change the shape of the envelope curve.  The large nodes are value indicators of the parameters that appear in the horizontal parameter bar below—Init Level, Attack Time, Decay Time, and so on.
 Volume decay mode buttons: Click to choose the volume decay curve.  Exp: The output of the volume envelope is shaped by an exponential algorithm in order to generate the most natural sounding reverb tail.  Lin: The volume decay will be more linear (and less natural sounding).  End Level: Sets the end volume level. It is expressed as a percentage of the overall volume envelope. If you set this parameter to 0%, the reverb tail cuts off abruptly, which is great for gated reverb effects.
Setting the Filter Mode The Filter Mode knob switches between four modes. Click the desired LP (lowpass) 6 dB and 12 dB, BP (bandpass) or HP (highpass) value. Â 6 dB (LP): Bright, good general-purpose filter mode. It can be used to retain the top end of most material, while still providing some filtering. Â 12 dB (LP): Useful where you want a warmer sound, without drastic filter effects. It is handy for smoothing out bright reverbs. Â BP: 6 dB per octave design.
Synthesizer Impulse Response Parameters In Synthesizer IR mode, Space Designer generates a synthesized impulse response determined by the values of the Length, envelopes, filter, EQ, and spread parameters. To switch to Synthesizer IR mode, enable the Synthesized IR button in the impulse response parameters section. Clicking the activated Synthesized IR button will randomly generate new impulse responses with slightly different reflection patterns.
The density envelope offers the following parameters: Â Init Level: Sets the initial density (the average number of reflections in a given period of time) of the reverb. Lowering the density levels will result in audible reflections patterns and discreet echoes. Â Ramp Time: Adjusts the length of time elapsed between the Initial and End Density levels. Â End Level: Sets the density of the reverb tail. If you select an End Level value that is too low, the reverb tail will sound grainy.
Spread extends the stereo or surround base to frequencies that fall below the frequency determined by the Xover (crossover) parameter. At a Spread value of 0, no stereo or surround information is added (although the inherent stereo or surround information of a signal and reverb will be retained). At a value of 100, the left and right channel divergence is at its maximum. The Xover parameter is set in Hertz.
The EQ has the following parameters: Â EQ On/Off button: Click to switch the entire EQ section on or off. Â Individual EQ buttons (1 through 4): Click to turn individual EQ bands on or off. Â Frequency: Sets the frequency for the selected EQ band. Â Gain: Adjusts the gain cut or boost for the selected EQ band. Â Q: Sets the Q-factor for the two parametric bands. The Q can be adjusted from 0.1 (very narrow) to 10 (very wide).
12 Specialized 12 Soundtrack Pro includes a bundle of specialized plug-ins designed to address tasks often encountered during audio production. Consider using these specialized effects if you want to do any of the following: Â Eliminate or reduce noise below a threshold level (see “Denoiser” on page 141). Â Add life to digital recordings by adding additional high frequency components (see “Exciter” on page 143).
For example, if the noise floor of your recording is very high (more than –68 dB), reducing it to a level of –83 to –78 dB should be sufficient, provided this does not introduce any audible side effects. This effectively reduces the noise by more than 10 dB, to less than half of the original (noise) volume. Â Noise Type slider and field: Set to a value appropriate to the type of noise you want to reduce. Â A value of 0 equals white noise (equal frequency distribution).
Exciter The Exciter generates high-frequency components that are not part of the original signal, using a nonlinear distortion process that resembles overdrive and distortion effects. Unlike those effects, however, the Exciter passes the input signal through a highpass filter before feeding it into the harmonics (distortion) generator. This results in the artificial harmonics added to the signal having frequencies at least one octave above the threshold of the highpass filter.
SubBass The SubBass plug-in generates frequencies below those of the original signal—in other words, an artificial bass. The simplest use for the SubBass is as an octave divider, similar to Octaver effect pedals for electric bass guitars. Where such pedals can only process a monophonic input sound source of clearly defined pitch, SubBass can be used with complex summed signals as well. SubBass creates two bass signals, derived from two separate portions of the incoming signal.
Using the SubBass Unlike a pitch shifter, the waveform of the signal generated by the SubBass is not based on the waveform of the input signal, but is sinusoidal (it uses a sine wave). Given that pure sine waves rarely sit well in complex arrangements, you can control the amount of (and balance between) the generated and original signals using the Dry and Wet sliders. You define the two frequency bands (which the SubBass uses to generate tones) with the High and Low parameters.
13 Utility 13 The Utility plug-ins are handy tools that can help you with routine tasks and situations that you may encounter when producing music. This includes the following tasks: Â Adjusting the level or phase of input signals (see “Gain” on page 147 and “Multichannel Gain” on page 148) Â Generating a static frequency or sine sweep (see “Test Oscillator” on page 149) Gain Gain lets you amplify (or reduce) the signal by a specific decibel amount.
 Swap L/R (Left/Right) button: When selected, swaps the left and right output channels. The swapping occurs after the Balance in the signal path.  Mono button: When selected, outputs the summed mono signal on both the left and right channels. Using Phase Inversion Inverting phase lets you combat time alignment problems, particularly those caused by recording with multiple microphones at the same time. When you invert the phase of a signal heard in isolation, it sounds identical to the original.
Test Oscillator The Test Oscillator generates a static frequency or a sine sweep. The latter is a userdefined frequency spectrum tone sweep. Test Oscillator Parameters  Waveform buttons: Select the type of waveform to be used for test tone generation.  The Square Wave and Needle Pulse waveforms are available as either aliased or anti-aliased versions. The latter is used in conjunction with the Anti Aliased button.  Needle Pulse is a single needle impulse waveform.
