Soundtrack Pro 3 Effects Reference
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Contents Preface 7 7 9 9 Introduction to the Soundtrack Pro Plug-Ins About the Soundtrack Pro Effects About the Soundtrack Pro Documentation Additional Resources Chapter 1 11 11 31 32 Delay Effects Delay Designer Stereo Delay Tape Delay Chapter 2 35 36 37 38 39 39 40 Distortion Effects Bitcrusher Clip Distortion Distortion Effect Distortion II Overdrive Phase Distortion Chapter 3 41 41 43 44 47 49 50 51 54 57 Dynamics Processors About Dynamics Processors Adaptive Limiter Compressor Enveloper Exp
74 Single-Band EQs 4 Chapter 5 77 77 82 82 Filter Effects AutoFilter Soundtrack Pro Autofilter Spectral Gate Chapter 6 85 Imaging Processors 85 Direction Mixer 88 Stereo Spread Chapter 7 91 91 91 96 102 Metering Tools Correlation Meter MultiMeter Surround MultiMeter Tuner Chapter 8 105 106 106 108 108 110 112 117 118 Modulation Effects Chorus Effect Ensemble Effect Flanger Effect Modulation Delay Phaser Effect Ringshifter Scanner Vibrato Tremolo Effect Chapter 9 121 Pitch Effects 121 Pitch Shi
Chapter 12 153 153 155 157 158 Specialized Effects and Utilities DeEsser Denoiser Exciter SubBass Chapter 13 161 161 162 163 Utilities and Tools Gain Plug-in 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 preface covers the following: • About the Soundtrack Pro Effects (p. 7) • About the Soundtrack Pro Documentation (p. 9) • Additional Resources (p. 9) About the Soundtrack Pro Effects The effects plug-ins included with Soundtrack Pro allow you to process audio in a number of different ways.
Effect category Included effects Dynamics Processors Adaptive Limiter Compressor Enveloper Expander Limiter Multipressor Noise Gate Surround Compressor Equalizers Channel EQ Fat EQ Linear Phase EQ Match EQ Single-Band EQs Filter Effects AutoFilter Soundtrack Pro Autofilter Spectral Gate Imaging Processors Direction Mixer Stereo Spread Metering Tools Correlation Meter MultiMeter Surround MultiMeter Tuner Modulation Effects Chorus Effect Ensemble Effect Flanger Effect Modulation Delay Phaser E
Effect category Included effects Utilities and Tools Gain Plug-in Multichannel Gain Test Oscillator About the Soundtrack Pro Documentation Soundtrack Pro comes with various documentation that will help you get started as well as provide detailed information about the application.
Delay Effects 1 Delay effects store the input signal—and hold it for a short time—before sending it to the effect input or output. The held, and delayed, signal is repeated after a given time period, creating a repeating echo effect. Each subsequent repeat is a little quieter than the previous one. Most delays also allow you to feed a percentage of the delayed signal back to the input. This can result in a subtle, chorus-like effect or cascading, chaotic audio output.
• Pitch transposition (up or down) Further effect-wide parameters include synchronization, quantization, and feedback. As the name implies, Delay Designer offers significant sound design potential. You can use it for everything from a basic echo effect to an audio pattern sequencer. You can create complex, evolving, moving rhythms by synchronizing the placement of taps. This leads to further musical possibilities when coupled with judicious use of transposition and filtering.
• Master section: This area contains the global Mix and Feedback parameters. See Using Delay Designer’s Master Section. Getting to Know Delay Designer’s Main Display Delay Designer’s 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.
• Identification bar: Shows an identification letter for each tap. It also serves as an indicator of time position 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. Using Delay Designer’s View Buttons The view buttons determine which parameter is represented in Delay Designer’s Tap display. • Cutoff button: Shows the highpass and lowpass filter cutoff frequencies of taps.
Tip: If the Overview display is hidden behind a tap, you can move it to the foreground by holding down Shift. To zoom the Tap display Do one of the following: µ Drag the highlighted section (the bright rectangle) of the Overview display up or down. µ Drag the highlighted bars—to the left or right of the bright rectangle—to the left or right. Note: The Autozoom button needs to be disabled for this to work. When you zoom in on a small group of taps, the Overview display continues to show all taps.
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 begin recording new taps. 3 Click the Tap button to create new taps. These are created at the exact moments in time of each click, adopting the rhythm of your click pattern. 4 To finish creating taps, click the Last Tap button.
Delay Designer Tap Creation Suggestions The fastest way to create multiple taps is to use the Tap pads. If you have a specific rhythm in mind, you might find it easier to tap out your rhythm on dedicated hardware controller buttons, instead of using mouse clicks. If you have a MIDI controller, you can assign the Tap pads to buttons on your device. For information about assigning controllers, see the Soundtrack Pro User Manual.
Selecting Taps in Delay Designer There will always be at least one selected tap. You can easily distinguish selected taps by color—the toggle bar icons and the Identification bar letters of selected taps are white. To select a tap Do one of the following: µ µ µ µ Click a tap in the Tap display. Click the desired tap letter in the Identification bar. Click one of the arrows to the left of the Tap name to select the next or previous tap.
Moving and Deleting Taps in Delay Designer You can move a tap backward or forward in time, or completely remove it. Note: When you move a tap, you are actually editing its delay time. µ To move a selected tap in time Select the tap in the Identification bar, and drag it to the left to go forward in time, or to the right to go backward in time. This method also works when more than one tap is selected.
Using Delay Designer’s Tap Toggle Buttons 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: • Cutoff view: Toggle buttons turn the filter on or off. • Reso view: Toggle buttons switch the filter slope between 6 dB and 12 dB. • Pitch view: Toggle buttons switch pitch transposition on or off.
2 Vertically drag the bright line of the tap you wish to edit (or one of the selected taps, if multiple taps are selected). If you have chosen multiple taps, the values of all selected taps will be changed relative to each other. Note: The method outlined above is slightly different for the Filter Cutoff and Pan parameters. See Editing Filter Cutoff in Delay Designer’s Tap Display and Editing Pan in Delay Designer’s Tap Display.
To align the values of several taps 1 Command-click in the Tap display, and move the pointer while holding down the Command key. This will result in a line trailing behind the pointer. 2 Click the desired position to mark the end point of the line. The values of taps that fall between the start and end points are aligned along the line.
Editing Filter Cutoff in Delay Designer’s Tap Display Whereas the steps outlined in Editing Parameters in Delay Designer’s Tap Display apply to most graphically editable parameters, the Cutoff and Pan parameters work in a slightly different fashion. In Cutoff view, each tap actually shows two parameters: highpass and lowpass filter cutoff frequency.
Editing Pan in Delay Designer’s Tap Display The way the Pan parameter is represented in the Pan view is entirely dependent on the audio file channel configuration. In stereo configurations, the Pan parameter adjusts the stereo balance. The Pan parameter appears as a dot on the tap, which represents stereo balance. Drag the dot up or down the tap to adjust the stereo balance. By default, stereo spread is set to 100%. To adjust this, drag either side of the dot.
Editing Taps in Delay Designer’s Tap Parameter Bar The Tap parameter bar provides instant access to all parameters of the chosen tap. The Tap parameter bar also provides access to several parameters that are not available in the Tap display, such as Transpose and Flip. Editing in the Tap parameter bar is fast and precise when you want to edit the parameters of a single tap. All parameters of the selected tap are available, with no need to switch display views or estimate values with vertical lines.
• Pan field: Controls the stereo balance for stereo signals, and surround angle when used in surround configurations. • Pan displays a percentage between 100% (full left) and -100% (full right), which represents the balance of the tap. A value of 0% represents the center panorama position. • When used in surround, a surround panner replaces the percentage representation. For more information, see Working with Delay Designer in Surround.
To reset the value of a tap Do one of the following: µ In the Tap display, Option-click a tap to reset the chosen parameter to its default setting. If multiple taps are selected, Option-clicking any tap will reset the chosen parameter to its default value for all selected taps. µ In the Tap parameter bar, Option-click a parameter value to reset it to the default setting.
Note: Delay Designer offers a maximum delay time of 10 seconds. This means that if you load a setting into a project with a slower tempo than the tempo at which it was created, some taps may fall outside the 10-second limit. In such cases, these taps will not be played but will be retained as part of the setting. • Sync button: Enables or disables synchronized mode. • Grid pop-up menu: Provides several grid resolutions, which correspond to musical note durations.
Use subtle variations of the grid position of every second increment (values between 45% and 55%) to create a less rigid rhythmic feel. This can deliver very human timing variations. Use of extremely high Swing values are unsubtle as they place every second increment directly beside the subsequent increment. Make use of higher values to create interesting and intricate double rhythms with some taps, while retaining the grid to lock other taps into more rigid synchronization with the project tempo.
Note: If Feedback is enabled and you begin creating taps with the Tap pads, Feedback is automatically turned off. When you stop creating taps with the Tap pads, Feedback is automatically re-enabled. • Mix sliders: Independently set the levels of the dry input signal and the post-processing wet signal. Working with Delay Designer in Surround Delay Designer is optimally designed for use in surround configurations.
Stereo Delay The Stereo Delay works much like the Tape Delay (see Tape Delay), but allows you to set the Delay, Feedback, and Mix parameters separately for the left and right channels. The effect also features a Crossfeed knob for each stereo side that determines the feedback intensity or the level at which each signal is routed to the opposite stereo side. You can freely use the Stereo Delay on mono tracks or busses when you want to create independent delays for the two stereo sides.
• Crossfeed Left to Right (Crossfeed Right to Left) knob and field: Transfer the feedback signal of the left channel to the right channel, and vice versa. • Feedback Phase button: Use to invert the phase of the corresponding channel’s feedback signal. • Crossfeed Phase button: Use to invert the phase of the crossfed feedback signals. Common Parameters • Beat Sync button: Synchronizes delay repeats to the project tempo, including tempo changes.
• Delay field: Sets the current delay time in milliseconds (this parameter is dimmed when you synchronize the delay time to the project tempo). • Sync button: Synchronizes delay repeats to the project tempo (including tempo changes). • Tempo field: Sets the current delay time in beats per minute (this parameter is dimmed when you synchronize the delay time to the project tempo).
Distortion Effects 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 they are still used in musical instrument amplifiers today.
Bitcrusher Bitcrusher is a low-resolution digital distortion effect. You can use it to emulate the sound of early digital audio devices, to create artificial aliasing by dividing the sample rate, or to distort signals until they are unrecognizable. • Drive slider and field: Set the amount of gain in decibels applied to the input signal. Note: Raising the Drive level tends to increase the amount of clipping at the output of the Bitcrusher as well.
• Displaced: The start, center, and end levels of the signal (above the threshold) are offset, resulting in a distortion that is less severe as signal levels cross the threshold. The center portion of the clipped signal is also softer than in Cut mode. • Clip Level slider and field: Sets the point (below the clipping threshold of the channel strip) at which the signal starts clipping. Clip Distortion Clip Distortion is a nonlinear distortion effect that produces unpredictable spectra.
• Sum LPF knob and field: Sets the cutoff frequency (in Hertz) of the lowpass filter. This processes the mixed signal. • (High Shelving) Frequency knob and field: Sets the frequency (in Hertz) of the high shelving filter. 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 these types of treble controls, however, you can boost or cut the signal by up to plus or minus 30 dB with the Gain parameter.
Distortion II Distortion II emulates the distortion circuit of a Hammond B3 organ. You can use it on musical instruments to recreate this classic effect, or use it creatively when designing new sounds. • PreGain knob: Sets the amount of gain applied to the input signal. • Drive knob: Sets the amount of saturation applied to the signal. • Tone knob: Sets the frequency of the highpass filter. Filtering the harmonically rich distorted signal produces a softer tone.
• Display: Shows the impact of parameters on the signal. • Tone knob and field: Sets the frequency for the high cut filter. Filtering the harmonically rich distorted signal produces a softer tone. • Output slider and field: Sets the output level. This allows you to compensate for increases in loudness caused by using Overdrive. Phase Distortion The Phase Distortion effect is based on a modulated delay line, similar to a chorus or flanger effect (see Modulation Effects).
Dynamics Processors 3 The Dynamics processors 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 highest amplitudes. Dynamics processors enable you to adjust the dynamic range of individual audio files, tracks, or an overall project.
By reducing the highest parts of the signal, called peaks, a compressor raises the overall level of the signal, increasing the perceived volume. This gives the signal more focus by making the louder (foreground) parts stand out, while keeping the softer background parts from becoming inaudible. Compression also tends to make sounds tighter or punchier because transients are emphasized, depending on attack and release settings, and because the maximum volume is reached more swiftly.
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.
• 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. Compressor The Compressor is designed to emulate the sound and response of a professional-level analog (hardware) compressor.
• Compression curve display: Shows the compression curve created by the combination of Ratio and Knee parameter values. Input (level) is shown on the x axis and output (level) on the y axis. • Release knob and field: Determines the amount of time it takes for the compressor to stop reducing the signal after the signal level falls below the threshold. • Auto button: When the Auto button is active, the release time dynamically adjusts to the audio material.
Setting Suitable Compressor Envelope Times The Attack and Release parameters shape the dynamic response of the Compressor. The Attack parameter determines the time it takes after the signal exceeds the threshold level before the Compressor starts reducing the signal. Many sounds, including voices and musical instruments, rely on the initial attack phase to define the core timbre and characteristic of the sound.
Note: If you activate Auto Gain and RMS simultaneously, the signal may become oversaturated. If you hear any distortion, switch Auto Gain off and adjust the Gain slider until the distortion is inaudible. Enveloper The Enveloper is an unusual processor that lets you shape the attack and release phases of a signal—the signal’s transients, in other words. This makes it a unique tool that can be used to achieve results that differ from other dynamic processors.
Using the Enveloper The most important parameters of the Enveloper are the two Gain sliders, one on each side of the central display. These govern the Attack and Release levels of each respective phase. Boosting the attack phase can add snap to a drum sound, or it can amplify the initial pluck or pick sound of a stringed instrument. Attenuating the attack causes percussive signals to fade in more softly. You can also mute the attack, making it virtually inaudible.
Expander The Expander is similar in concept to a compressor, but increases, rather than reduces, the dynamic range above the threshold level. You can use the Expander to add liveliness and freshness to your audio signals. • Threshold slider and field: Sets the threshold level. Signals above this level are expanded. • Peak/RMS buttons: Determine whether the Peak or RMS method is used to analyze the signal.
Limiter The Limiter works much like a compressor but 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 when mastering.
Multipressor The Multipressor (short for multiband compressor) is an extremely versatile audio mastering tool. It splits the incoming signal into different frequency bands—up to four—and 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.
Multipressor Parameters The parameters in the Multipressor window are grouped into three main areas: the graphical display in the upper section, the set of controls for each frequency band in the lower section, and the output parameters on the right. Graphical display section Frequency band section Output section Multipressor Graphical Display Section • Graphical display: Each frequency band is represented graphically. The amount of gain change from 0 dB is indicated by blue bars.
• Expnd Thrsh(old) fields: Set the expansion threshold for the selected band. Setting the parameter to its minimum value (-60 dB), means that only signals that fall below this level are expanded. • Expnd Ratio fields: Set the expansion ratio for the selected band. • Expnd Reduction fields: Set the amount of downward expansion for the selected band. • Peak/RMS fields: Enter a smaller value for shorter peak detection, or a larger value for RMS detection, in milliseconds.
Setting Multipressor Compression Parameters The Compression Threshold and Compression Ratio parameters are the key parameters for controlling compression. Usually the most useful combinations of these two settings are a low Compression Threshold with a low Compression Ratio, or a high Compression Threshold with a high Compression Ratio.
Noise Gate Parameters The Noise Gate has the following parameters. • Threshold slider and field: Sets the threshold level. Signals that fall below the threshold will be reduced in level. • Reduction slider and field: Sets the amount of signal reduction. • Attack knob and field: Sets the amount of time it takes to fully open the gate after the signal exceeds the threshold. • Hold knob and field: Sets the amount of time the gate is kept open after the signal falls below the threshold.
Using the Noise Gate In most situations, setting the Reduction slider to the lowest possible value ensures that sounds below the Threshold value are completely suppressed. Setting Reduction to a higher value attenuates low-level sounds but still allows them to pass. You can also use Reduction to boost the signal by up to 20 dB, which is useful for ducking effects. The Attack, Hold, and Release knobs modify the dynamic response of the Noise Gate.
The filters allow only very high (loud) signal peaks to pass. In the drum kit example, you could remove the hi-hat signal, which is higher in frequency, with the High Cut filter and allow the snare signal to pass. Turn monitoring off to set a suitable Threshold level more easily. Surround Compressor The Surround Compressor, based on the Compressor, is specifically designed for compression of complete surround mixes.
Surround Compressor Link Section Parameters The Surround Compressor’s Link section provides the following parameters: • Circuit Type pop-up menu: Choose the type of circuit emulated by the Compressor. The choices are Platinum, Classic A_R, Classic A_U, VCA, FET, and Opto (optical). • Grp. (Group) pop-up menus: Set group membership for each channel (A, B, C, or no group, indicated by -).
• Knee knob and field: Determines the ratio of compression at levels close to the threshold. • Attack knob and field: Sets the amount of time it takes to reach full compression, after the signal exceeds the threshold. • Release knob and field: Sets the amount of time it takes to return to 0 compression, after the signal falls below the threshold. • Auto button: When the Auto button is enabled, the release time dynamically adjusts to the audio material.
• Threshold knob and field: Sets the threshold for the limiter on the LFE channel. • Limiter button: Enables and disables limiting for the LFE channel.
Equalizers 4 An equalizer (commonly abbreviated as EQ) shapes the sound of incoming audio by changing the level of specific frequency bands. Equalization is one of the most commonly used audio processes, both for music projects and in post-production work for video. You can use EQ to subtly or significantly shape the sound of an audio file, instrument, or project by adjusting specific frequencies or frequency ranges.
Channel EQ The Channel EQ is a highly versatile multiband EQ. It provides eight frequency bands, including lowpass and highpass filters, low and high shelving filters, and four flexible parametric bands. It also features an integrated Fast Fourier Transform (FFT) Analyzer that you can use to view the frequency curve of the audio you want to modify, allowing you to see which parts of the frequency spectrum may need adjustment.
Channel EQ Graphical Display Section • Band On/Off buttons: Click to turn the corresponding band on or off. Each button icon indicates the filter type: Band 1 is a highpass filter. Band 2 is a low shelving filter. Bands 3 through 6 are parametric bell filters. Band 7 is a high shelving filter. Band 8 is a lowpass filter. • Graphical display: Shows the current curve of each EQ band. • Drag horizontally in the section of the display that encompasses each band to adjust the frequency of the band.
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 frequency—either 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.
Fat EQ The Fat EQ is a versatile multiband EQ that can be used on individual sources or overall mixes. The Fat EQ provides up to five individual frequency bands, includes a graphical display of the EQ curves, and includes a set of parameters for each band. The Fat EQ offers the following parameters. • Band Type buttons: Located above the graphical display. For bands 1–2 and 4–5, click one of the paired buttons to select the EQ type for the corresponding band.
Note: For bands 1 and 5, this changes the slope of the filter. • Band On/Off buttons: Enable/disable the corresponding band. • Master Gain slider and field: Sets the overall output level of the signal. Use it after boosting or cutting individual frequency bands. Linear Phase EQ The high-quality Linear Phase EQ effect is similar to the Channel EQ, sharing the same parameters and eight-band layout.
Linear Phase EQ Gain and Analyzer Controls • Master Gain slider and field: Sets the overall output level of the signal. Use it after boosting or cutting individual frequency bands. • Analyzer button: Turns the Analyzer on or off. • Pre/Post EQ menu: Determines whether the Analyzer shows the frequency curve before or after EQ is applied, when Analyzer mode is active.
• Analyzer Mode buttons (Extended Parameters area): Choose Peak or RMS. Using the Linear Phase EQ The Linear Phase EQ is typically used as a mastering tool and is, therefore, generally inserted into master or output channel strips.
The bands derived from FFT analysis are divided in accordance with the frequency linear principle—there 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 graphical display. The visible area represents a dynamic range of 60 dB. Drag vertically to set the maximum value to anywhere between +20 dB and -40 dB. The Analyzer display is always dB-linear.
Match EQ Parameters The Match EQ offers the following parameters. Match EQ Analyzer Parameters • Analyzer button: Turns the Analyzer function on or off. • Pre/Post menu: Determines whether the Analyzer looks at the signal before (Pre) or after (Post) the filter curve is applied. • View pop-up menu: Sets the information shown in the graphical display. Choices are: • Automatic: Displays information for the current function, as determined by the active button below the graphical display.
• Select pop-up menu (Surround instances only): The Select buttons are replaced by the Select pop-up menu, enabling you to choose an individual channel or all channels. Changes to the filter curve will affect the chosen channel when a single channel is selected. • Channel Link slider and field: Refines the settings made with the Select buttons or Select pop-up menu. • When set to 100%, all channels (L and R for stereo, or all surround channels) are represented by a common EQ curve.
Note: Smoothing has no effect on any manual changes you make to the filter curve. Using the Match EQ Following is a common usage example that you can adapt to your own workflow. In this example, the frequency spectrum of a mix is matched with the spectrum of a source audio file. To learn or create a Match EQ template 1 Choose Process > Effects > EQ > Match EQ. The Match EQ HUD appears. 2 Adjust the Match EQ settings to your liking, or drag source audio onto the Template 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 click 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 incoming audio signal (Current Material) begins.
The Q factor of the filter is determined (and set) by the vertical distance between the clicked position and the curve. µ µ To set the Match EQ Q factor Click the curve directly to set the maximum Q value of 10 (for notch-like filters). Click above or below the curve to decrease the Q value. The farther you click from the curve, the smaller the value (down to the minimum of 0.3). The colors and modes of the dB scales on the left and right of the display are automatically adapted to the active function.
High Cut and Low Cut Filter The Low Cut Filter attenuates the frequency range that falls below the selected frequency. The High Cut Filter attenuates the frequency range above the selected frequency. Use the Frequency field and slider to set the cutoff frequency. High Pass and Low Pass Filter The High Pass Filter affects the frequency range below the set frequency. Higher frequencies pass through the filter. You can use the High Pass Filter to eliminate the bass below a selectable frequency.
High Shelving and Low Shelving EQ The Low Shelving EQ affects only the frequency range that falls below the selected frequency. The High Shelving EQ affects only the frequency range above the selected frequency. • Gain field and slider: Sets the amount of cut or boost. • Frequency field and slider: Sets the cutoff frequency. Parametric EQ The Parametric EQ is a simple filter with a variable center frequency.
Filter Effects 5 Filters are used to emphasize or suppress frequencies in an audio signal, resulting in a change to the tonal color of the audio. Soundtrack Pro contains a variety of advanced filter-based effects that you can use to creatively modify your audio. These effects are most often used to radically alter the frequency spectrum of a sound or mix. Note: Equalizers (EQs) are special types of filters.
Getting to Know the AutoFilter Interface The main areas of the AutoFilter window are the Threshold, Envelope, LFO, Filter, Distortion, and Output parameter sections. Threshold parameter Envelope parameters Filter parameters Output parameters LFO parameters Distortion parameters • Threshold parameter: Sets an input level that—if exceeded—triggers the envelope or LFO, which are used to dynamically modulate the filter cutoff frequency.
The envelope and LFO can be used to modulate the filter cutoff frequency. AutoFilter Envelope Parameters The envelope is used to shape the filter cutoff over time. When the input signal exceeds the set threshold level, the envelope is triggered. • Attack knob and field: Sets the attack time for the envelope. • Decay knob and field: Sets the decay time for the envelope. • Sustain knob and field: Sets the sustain time for the envelope.
• Beat Sync button: Activate to synchronize the LFO to the host application tempo. You can choose from bar values, triplet values, and more. These are determined by the Rate knob. • Phase knob: Shifts the phase relationship between the LFO rate and the host application tempo—when Beat Sync is active. • Decay/Delay knob and field: Sets the amount of time it takes for the LFO to go from 0 to its maximum value. • Rate Mod.
• Resonance knob: Boosts or cuts the signals in the frequency band that surrounds the cutoff frequency. Use of very high Resonance values causes the filter to begin oscillating at the cutoff frequency. This self-oscillation occurs before you reach the maximum Resonance value. • Fatness slider and field: Boosts the level of low frequency content. When you set Fatness to its maximum value, adjusting Resonance has no effect on frequencies below the cutoff frequency.
AutoFilter Output Parameters The Output parameters are used to set the wet/dry balance and overall level. • Dry Signal slider and field: Sets the amount of the original dry signal added to the filtered signal. • Main Out slider and field: Sets the overall output level of the AutoFilter, allowing you to compensate for higher levels caused by adding distortion or by the filtering process itself.
It works by dividing the incoming signal into two frequency ranges—above and below a central frequency band that you specify with the Center Freq and Bandwidth parameters. The signal ranges above and below the defined band can be individually processed with the Low Level and High Level parameters and the Super Energy and Sub Energy parameters.
• Sub Energy and field: Controls the level of the frequency range below the threshold. • Low Level slider and field: Blends the frequencies of the original signal—below the selected frequency band—with the processed signal. • Gain slider and field: Sets the output level of the Spectral Gate. Using the Spectral Gate One way to familiarize yourself with the operation of the Spectral Gate would be to start with a drum loop. Set the Center Freq.
Imaging Processors 6 The Soundtrack Pro Imaging processors are tools for manipulating the stereo image. This enables 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. The following sections describe the imaging processors included with Soundtrack Pro. This chapter covers the following: • Direction Mixer (p. 85) • Stereo Spread (p.
• Spread slider and field: Determines the spread of the stereo base in LR input signals. Determines the level of the side signal in MS input signals. See Using the Direction Mixer’s Spread Parameter. • Direction knob and field: Determines the pan position for the middle—the center of the stereo base—of the recorded stereo signal. See Using the Direction Mixer’s Direction Parameter.
• 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.
Understanding MS Miking To make a Middle Side (MS) 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 you want to record—in 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.
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 alternately—middle 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.
Metering Tools 7 You can use the Metering tools to analyze audio in a variety of ways. These utilities have no effect on the audio signal. They are simply used as diagnostic aids. This chapter covers the following: • Correlation Meter (p. 91) • MultiMeter (p. 91) • Surround MultiMeter (p. 96) • Tuner (p. 102) Correlation Meter The Correlation Meter displays the phase relationship of a stereo signal.
• A Correlation Meter to spot mono phase compatibility • An integrated Level Meter to view the signal level for each channel You can view either the Analyzer or Goniometer results in the main display area. You switch the view and set other MultiMeter parameters with the controls on the left side of the interface.
Using the MultiMeter Analyzer In Analyzer mode, the MultiMeter’s main display shows the frequency spectrum of the input signal as 31 independent frequency bands. Each frequency band represents one-third of an octave. The Analyzer parameters are used to activate Analyzer mode, and to customize the way that the incoming signal is shown in the main display. Analyzer parameters Scale • Analyzer button: Switches the main display to Analyzer mode.
Using the MultiMeter Goniometer A goniometer helps you to judge the coherence of the stereo image and determine phase differences between the left and right channels. Phase problems are easily spotted as trace cancelations along the center line (M—mid/mono). The idea of the goniometer was born with the advent of early two-channel oscilloscopes.
Using the MultiMeter’s Level 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 mark turns red. Current peak values are displayed numerically (in dB increments) above the Level Meter.
Using the MultiMeter Peak Parameters The MultiMeter Peak parameters are used to enable/disable the peak hold function and to reset the peak segments of all meter types. You can also determine a temporary peak hold duration. • Hold button: Activates peak hold for all metering tools in the MultiMeter, as follows: • Analyzer: A small yellow segment above each 1/3 octave level bar indicates the most recent peak level. • Goniometer: All illuminated pixels are held during a peak hold.
Although you can insert the Surround MultiMeter directly into any channel strip, it is more commonly used in the master channel strip of the host application—when you are working on the overall surround mix. Analyzer parameters Goniometer parameters Peak parameters Main display (Goniometer shown) Balance/Correlation button Using the Surround MultiMeter Analyzer In Analyzer mode, the MultiMeter’s main display shows the frequency spectrum of the input signal as 31 independent frequency bands.
• Sum and Max buttons: Determine whether a summed or maximum level is displayed in the Analyzer results in the main display. These buttons are relevant only when multiple channels are selected with the channel buttons. • Channel buttons: Used to select a single channel or a combination of channels for metering. The number and appearance of these buttons varies when different surround modes are chosen.
Because the Surround MultiMeter Goniometer is dealing with multichannel signals, the display is divided into multiple segments, as shown in the image. Each segment indicates a speaker position. When the surround panner is moved in a channel strip, the indicator changes accordingly. This indicates not only left and right channel coherence, but also the front-to-rear coherence. • Goniometer button: Displays the Goniometer results in the main display.
Using the Surround MultiMeter Level 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 mark turns red. Current peak values are displayed numerically (in dB increments) above the Level Meter.
Depending on the chosen surround format, a number of points that indicate speaker positions are shown (L, R, C, Ls, Rs in a 5.1 configuration is displayed in the figure). Lines connect these points. The center position of each connecting line is indicated by a blue marker. A gray ball indicates the surround field/sound placement. As you move the surround panner of the channel strip, the ball in the Correlation Meter mirrors your movements.
• Hold Time pop-up menu: When peak hold is active, sets the hold time for all metering tools to 2, 4, or 6 seconds—or infinite. • Reset button: Click to reset the peak hold segments of all metering tools. Tuner You can tune instruments connected to your system with the Tuner utility. This ensures that your external instrument recordings will be in tune with any software instruments, samples, or existing recordings in your projects.
• Tuning Adjustment slider 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 to the left to lower the pitch corresponding to A. Drag the knob to the right to raise the pitch corresponding to A. The current value is displayed in the field. To use the Tuner 1 Insert the Tuner into an audio channel strip. 2 Play a single note on the instrument and watch the display.
Modulation Effects 8 Modulation effects are used to add motion and depth to your sound. Effects such as chorus, flanging, and phasing are known as modulation effects because they modulate the timing of the incoming signal. Typically, the incoming signal is delayed by a few milliseconds and then an LFO is used to modulate either the delay time, the delayed signal, or both.
Chorus Effect 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 incoming signal and create the impression that multiple instruments or voices are being played in unison. The slight delay time variations generated by the LFO simulate the subtle pitch and timing differences heard when several musicians or vocalists perform together.
The Ensemble effect can add a great deal of richness and movement to sounds, particularly when you use a high number of voices. It is very useful for thickening parts, but it can also be used to emulate more extreme pitch variations between voices, resulting in a detuned quality to processed material. • Intensity sliders and fields: Set the amount of modulation for each LFO. • Rate knobs and fields: Control the frequency of each LFO.
Flanger Effect The Flanger effect works in much the same way as the Chorus effect, but it uses a significantly shorter delay time. In addition, the effect signal can be fed back into the input of the delay line. Flanging is typically used to create changes that are described as adding a spacey or underwater quality to input signals. • Feedback slider and field: Determines the amount of the effect signal that is routed back into the input.
Although rich, combined flanging and chorus effects are possible, the Modulation Delay is capable of producing some extreme modulation effects. These include emulations of tape speed fluctuations and metallic, robot-like modulations of incoming signals. • Feedback slider and field: Determines the amount of the effect signal that is routed back to the input. If you’re going for radical flanging effects, enter a high Feedback value. If simple doubling is what you’re after, don’t use any feedback.
• 180° or –180° is equal to the greatest possible distance between the modulation phases of the channels. Note: The LFO Phase parameter is available only if the LFO Left Right Link button is active. • Distribution pop-up menu: Available only in surround instances, it defines how the phase offsets between the individual channels are distributed in the surround field. You can choose from “circular,” “left->right,” “front->rear,” “random,” and “new random” distributions.
• Feedback slider and field: Determines the amount of the effect signal that is routed back into the input of the effect. Phaser Sweep Section • Ceiling and Floor slider and fields: Use the individual slider handles to determine the frequency range affected by the LFO modulations. • Order slider and field: Allows you to choose between different phaser algorithms. The more orders a phaser has, the heavier the effect. The 4, 6, 8, 10, and 12 settings put five different phaser algorithms at your fingertips.
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. 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.
• Delay parameters: Use these to delay the effect signal. See Using the Ringshifter’s Delay. • Envelope follower parameters: The oscillator frequency and output signal can be modulated with an envelope follower. See Modulating the Ringshifter with the Envelope Follower. • LFO parameters: The oscillator frequency and output signal can be modulated with an LFO. See Modulating the Ringshifter with the LFO.
• In the ring modulator OSC mode, the Frequency parameter controls the frequency content (timbre) of the resulting effect. This timbre can range from subtle tremolo effects to clangorous metallic sounds. Following is a list of controls for the ringshifter oscillator: • Frequency control: Sets the frequency of the sine oscillator.
• Level knob and field: Sets the level of the delay added to the ring-modulated or frequency-shifted signal. A Level value of 0 passes the effect signal directly to the output (bypass). Modulating the Ringshifter with the Envelope Follower The oscillator Frequency and Dry/Wet parameters can be modulated with the internal envelope follower—and the LFO (see Modulating the Ringshifter with the LFO).
Modulating the Ringshifter with the LFO The oscillator Frequency and Dry/Wet parameters can be modulated with the LFO—and the envelope follower (see Modulating the Ringshifter with the Envelope Follower). The oscillator frequency even allows modulation through the 0 Hz point, thus changing the oscillation direction. The LFO produces continuous, cycled control signals. • Power button: Turns the LFO on or off and enables the following parameters.
• Feedback knob and field: Sets the amount of the signal that is routed back to the effect input. Feedback adds an edge to the Ringshifter sound and is useful for a variety of special effects. It produces a rich phasing sound when used in combination with a slow oscillator sweep. Comb filtering effects are created by using high Feedback settings with a short delay time (less than 10 ms).
• Chorus Int knob: Sets the intensity of a chosen chorus effect type. If a vibrato effect type is chosen, this parameter has no effect. • Stereo Phase knob: When set to a value between 0° and 360°, Stereo Phase determines the phase relationship between left and right channel modulations, thus enabling synchronized stereo effects. If you set the knob to “free,” you can set the modulation speed of the left and right channel independently.
• Symmetry and Smoothing knobs and fields: Use these to alter the shape of the LFO waveform. If Symmetry is set to 50% and Smoothing to 0%, the LFO waveform has a rectangular shape. This means that the timing of the highest and lowest volume signals is equal, with the switch between both states occurring abruptly. • Phase knob and field: Controls the phase relationship between the individual channel modulations. At 0, modulation values are reached simultaneously for all channels.
Pitch Effects 9 You can use the Pitch effects of Soundtrack Pro to transpose the pitch of audio signals. These effects can also be used for creating unison or slightly thickened parts, or even for creating harmony voices. This chapter covers the following: • Pitch Shifter II (p. 121) • Vocal Transformer (p. 122) Pitch Shifter II The Pitch Shifter II provides a simple way to combine a pitch-shifted version of the signal with the original signal.
• Vocals: Retains the intonation of the source, making it well-suited for signals that are inherently harmonic or melodious, such as string pads. • Mix slider and field: Sets the balance between the effect and original signals. To pitch shift 1 Set the Semi Tones slider for the amount of transposition, or pitch shift. 2 Set the Cents slider for the amount of detuning. 3 Click the Drums, Speech, or Vocals button to select the algorithm that best matches the material you are working with.
Vocal Transformer Parameters The Vocal Transformer offers the following parameters. • Pitch knob and field: Determines the amount of transposition applied to the input signal. • Robotize button: Enables Robotize mode, which is used to augment, diminish, or mirror the melody. See Using Vocal Transformer’s Robotize Mode. • Pitch Base slider and field (available only in Robotize mode): Use to transpose the note that the Tracking parameter (see below) is following.
The Pitch parameter is expressly used to change the pitch of a voice, not its character. If you set negative Pitch values for a female soprano voice, you can turn it into an alto voice without changing the specific character of the singer’s voice. The Formant parameter shifts the formants, while maintaining—or independently altering—the pitch. If you set this parameter to positive values, the singer sounds like Mickey Mouse.
10 Reverb Effects You can use Reverb effects to simulate the sound of acoustic environments such as rooms, concert halls, caverns, or an open space. Sound waves repeatedly bounce off the surfaces—walls, ceilings, windows, and so on—of any space, or off objects within a space, gradually dying out until they are inaudible. These bouncing sound waves 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 metal plates and springs, were also 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.
PlatinumVerb The PlatinumVerb allows you to edit both the early reflections and diffuse reverb tail separately, making it easy to precisely emulate real rooms. Its dual-band Reverb section splits the incoming signal into two bands, each of which is processed and can be edited separately.
PlatinumVerb Early Reflections Parameters The PlatinumVerb offers the following Early Reflections parameters: • Predelay slider and field: Determines the amount of time between the start of the original signal and the arrival of the early reflections. Extremely short Predelay settings can color the sound and make it difficult to pinpoint the position of the signal source.
PlatinumVerb Reverb Parameters The PlatinumVerb offers the following Reverb parameters: • Initial Delay slider and field: Sets the time between the original signal and the diffuse reverb tail. • Spread slider and field: Controls the stereo image of the reverb. At 0% the effect generates a monaural reverb. At 200% the stereo base is artificially expanded. • Crossover slider and field: Defines the frequency at which the input signal is split into two frequency bands, for separate processing.
• Diffusion slider and field: Sets the diffusion of the reverb tail. High Diffusion values represent a regular density, with few alterations in level, times, and panorama position over the course of the diffuse reverb signal. Low Diffusion values result in the reflection density becoming irregular and grainy. This also affects the stereo spectrum. As with Density, find the best balance for the signal. • Reverb Time slider and field: Determines the reverb time of the high band.
Space Designer Convolution Reverb 11 Space Designer is a convolution reverb effect. You can use it to place your audio signals in exceptionally realistic recreations of real-world acoustic environments. Space Designer generates reverb by convolving, or combining, an audio signal with an impulse response (IR) reverb sample. An impulse response is a recording of a room’s reverb characteristics or, to be more precise, a recording of all reflections in a given room, following an initial signal spike.
Getting to Know the Space Designer Interface The Space Designer interface consists of four main sections: Impulse response Envelope and EQ parameters parameters Main display Button bar Global parameters Global parameters Filter parameters Parameter bar • Impulse response parameters: Used to load, save, or manipulate (recorded or synthesized) impulse response files. The chosen IR file determines what Space Designer will use to convolve with your audio signal.
Working with Space Designer’s Impulse Response Parameters Space Designer can use either recorded impulse response files or its own synthesized impulse responses. The circular area to the left of the main display contains the impulse response parameters. These are used to determine the Impulse Response mode (IR Sample mode or Synthesized IR mode), load or create impulse responses, and set the sample rate and length.
Important: To convolve audio in real time, Space Designer must first calculate any parameter adjustments to the impulse response. This requires a moment or two, following parameter edits, and is indicated by a blue progress bar. During this parameter edit processing time you can continue to adjust the parameter. When calculation starts, the blue bar is replaced by a red bar, advising you that calculation is taking place.
Any mono, stereo, AIFF, SDII, or WAV file can be used as an IR. In addition, surround formats up to 7.1, discrete audio files, and B-format audio files that comprise a single surround IR can also be used. Working in Space Designer’s Synthesized IR Mode In Synthesized IR mode, Space Designer generates a synthesized impulse response based on the values of the Length, Envelope, Filter, EQ, and Spread parameters.
• If the project sample rate is 44.1 kHz, the options will be 22.05 kHz, 11.025 kHz, and 5512.5 Hz. Changing the sample rate upward increases—or changing it downward decreases—the frequency response (and length) of the impulse response, and to a degree the overall sound quality of the reverb. Upward sample rate changes are of benefit only if the original IR sample actually contains higher frequencies. When you are reducing the sample rate, use your ears to decide if the sonic quality meets your needs.
All envelopes are automatically calculated as a percentage of the overall length, which means that if this parameter is altered, your envelope curves will stretch or shrink to fit, saving you time and effort. When you are using an impulse response file, the Length parameter value cannot exceed the length of the actual impulse response sample. Longer impulse responses (sampled or synthesized) place a higher strain on the CPU.
• All button: Resets all envelopes and the EQ to default values. • Volume Env button: Displays the volume envelope in the foreground of the main display. The other envelope curves are shown as transparencies in the background. See Working with Space Designer’s Volume Envelope. • Filter Env button: Displays the filter envelope in the foreground of the main display. The other envelope curves are shown as transparencies in the background. See Working with Space Designer’s Filter.
Whereas some parameters are envelope-specific, all envelopes consist of the Attack Time and Decay Time parameters. The combined total of the Attack Time and Decay Time parameters is equal to the total length of the synthesized or sampled impulse response (see Setting Impulse Response Lengths in Space Designer), unless the Decay time is reduced. The large nodes are value indicators of the parameters shown in the parameter bar below—Init Level, Attack Time, Decay Time, and so on.
Working with Space Designer’s Volume Envelope The volume envelope is used to set the reverb’s initial level and adjust how the volume will change over time. You can edit all volume envelope parameters numerically, and many can also be edited graphically (see Setting Space Designer’s Envelope Parameters). Init Level node Decay Time/End Level node Attack/Decay Time node • Init Level field: Sets the initial volume level of the impulse response attack phase.
Using Space Designer’s Density Envelope The density envelope allows you to control the density of the synthesized impulse response over time. You can adjust the density envelope numerically in the parameter bar, and you can edit the Init Level, Ramp Time, and End Level parameters using the techniques described in Setting Space Designer’s Envelope Parameters. Note: The density envelope is available only in Synthesized IR mode.
Working with Space Designer’s EQ Space Designer features a four-band EQ comprised of two parametric mid-bands plus two shelving filters (one low shelving filter and one high shelving filter). You can edit the EQ parameters numerically in the parameter bar, or graphically in the main display. EQ On/Off button Individual EQ band buttons • EQ On/Off button: Enables or disables the entire EQ section. • Individual EQ band buttons: Enable or disable individual EQ bands.
2 Drag the pointer horizontally over the main display. When the pointer is in the access area of a band, the corresponding curve and parameter area are automatically highlighted and a pivot point is displayed. 3 Drag horizontally to adjust the frequency of the band. 4 Drag vertically to increase or decrease the Gain of the band. 5 Vertically drag the (illuminated) pivot point of a parametric EQ band to raise or lower the Q value.
Using Space Designer’s Main Filter Parameters The main filter parameters are found at the lower-left corner of the interface. • Filter On/Off button: Switches the filter section on and off. • Filter Mode knob: Determines the filter mode. • 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. • 12 dB (LP): Useful where you want a warmer sound, without drastic filter effects.
Note: Activation of the filter envelope automatically enables the main filter. Controls the Attack Time endpoint (and Decay Time startpoint) and Break Level parameters simultaneously. Controls the Decay endpoint and End Level parameters simultaneously. • Init Level field: Sets the initial cutoff frequency of the filter envelope. • Attack Time field: Determines the time required to reach the Break Level (see below). • Break Level field: Sets the maximum filter cutoff frequency that the envelope reaches.
Space Designer Global Parameters: Upper Section These parameters are found around the main display. Output sliders Input slider Latency Compensation button Definition area Rev Vol Compensation button • Input slider: Determines how Space Designer processes a stereo or surround input signal. For more information, see Using Space Designer’s Input Slider. • Latency Compensation button: Switches Space Designer’s internal latency compensation feature on or off.
• Spread and Xover knobs (synthesized IRs only): Spread adjusts the perceived width of the stereo or surround field. Xover sets the crossover frequency in Hertz. Any synthesized impulse response frequency that falls below this value will be affected by the Spread parameter. See Using Space Designer’s Spread Parameters. Using Space Designer’s Input Slider The Input slider behaves differently in stereo or surround instances. • In stereo instances, the Input slider determines how a stereo signal is processed.
Using Space Designer’s Latency Compensation Feature The complex calculations made by Space Designer take time. This time results in a processing delay, or latency, between the direct input signal and the processed output signal. When activated, the Latency Compensation feature delays the direct signal (in the Output section) to match the processing delay of the effect signal. Note: This is not related to latency compensation in the host application.
Using Space Designer’s Rev Vol Compensation Rev Vol Compensation (Reverb Volume Compensation) attempts to match the perceived (not actual) volume differences between 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. If this is the case, switch it off and adjust input and output levels accordingly.
Space Designer Surround Output Configuration Parameters • C(enter) slider: Adjusts the output level of the center channel independently of other surround channels. • Bal(ance) slider: Sets the level balance between the front (L-C-R) and rear (Ls-Rs) channels. • In 7.1 ITU surround, the balance pivots around the Lm-Rm speakers, taking the surround angles into account. • With 7.1 SDDS surround, the Lc-Rc speakers are considered front speakers.
This can be useful for eliminating level peaks at the beginning of the impulse response sample. Its use also affords a number of creative options, particularly when combined with the Reverse function (see Using Space Designer’s Button Bar). Note: The IR Start parameter is not available or required in Synthesized IR mode because, by design, the Length parameter provides identical functionality.
You can, however, record, edit, and play back any movement of the following Space Designer parameters in a suitable host application: • Stereo Crossfeed • Direct Output • Reverb Output 152 Chapter 11 Space Designer Convolution Reverb
Specialized Effects and Utilities 12 Soundtrack Pro includes a bundle of specialized effects and utilities designed to address tasks often encountered during audio production. See the section for each processor for typical usage. This chapter covers the following: • DeEsser (p. 153) • Denoiser (p. 155) • Exciter (p. 157) • SubBass (p. 158) DeEsser The DeEsser is a frequency-specific compressor, designed to compress a particular frequency band within a complex audio signal.
The Detector parameters are on the left side of the DeEsser window, and the Suppressor parameters are on the right. The center section includes the Detector and Suppressor displays and the Smoothing slider. DeEsser Detector Section • Detector Frequency knob and field: Sets the frequency range for analysis. • Detector Sensitivity knob and field: Sets the degree of responsiveness to the input signal.
Denoiser The Denoiser eliminates or reduces any noise below a threshold volume level. The Denoiser uses fast Fourier transform (FFT) analysis to recognize frequency bands of lower volume and less complex harmonic structure. It then reduces these low-level, less complex bands to the desired dB level. If you use the Denoiser too aggressively, however, the algorithm produces artifacts, which are usually less desirable than the existing noise.
• Reduce slider and field: Sets the amount of noise reduction applied to signals that fall below the threshold. When reducing noise, remember that each 6 dB reduction is equivalent to halving the volume level (and each 6 dB increase equals a doubling of the volume level). Note: 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 doesn’t introduce any audible side effects.
Using the Denoiser The following steps are recommended for using the Denoiser effect. To use the Denoiser 1 Locate a section of the audio where only noise is audible, and set the Threshold value so that only signals at, or below, this level are filtered out. 2 Play the audio signal and set the Reduce value to the point where noise reduction is optimal but little of the desired signal is reduced. 3 If you encounter artifacts, use the smoothing parameters.
• Harmonics knob and field: Sets the ratio between the effect and original signals, expressed as a percentage. If the Input button is turned off, this parameter has no effect. Note: In most cases, higher Frequency and Harmonics values are preferable, because human ears cannot easily distinguish between the artificial and original high frequencies. • Color 1 and Color 2 buttons: Color 1 generates a less dense harmonic distortion spectrum. Color 2 generates a more intense harmonic distortion.
SubBass Parameters The SubBass offers the following parameters. • High Ratio knob and field: Adjusts the ratio between the generated signal and the original upper band signal. • High Center knob and field: Sets the center frequency of the upper band. • High Bandwidth knob and field: Sets the width of the upper band. • Graphical display: Shows the selected upper and lower frequency bands. • Freq. Mix slider and field: Adjusts the mix ratio between the upper and lower frequency bands.
Using SubBass Unlike a pitch shifter, the waveform of the signal generated by SubBass is not based on the waveform of the input signal, but is sinusoidal—that is, 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 with the Wet and Dry sliders. Use the High and Low parameters to define the two frequency bands, which SubBass uses to generate tones.
Utilities and Tools 13 The tools found in the Utility category can help with routine tasks and situations that you may encounter during production, such as adjusting the level or phase of input signals, or generating a static frequency or sine sweep. See the section for each tool for typical usage. This chapter covers the following: • Gain Plug-in (p. 161) • Multichannel Gain (p. 162) • Test Oscillator (p. 163) Gain Plug-in Gain amplifies (or reduces) the signal by a specific decibel amount.
• Swap L/R (Left/Right) button: Swaps the left and right output channels. The swapping occurs after the Balance parameter in the signal path. • Mono button: Outputs the summed mono signal on both the left and right channels. Using Phase Inversion Inverting phase is useful for dealing with time alignment problems, particularly those caused by simultaneous recording with multiple microphones. When you invert the phase of a signal heard in isolation, it sounds identical to the original.
Test Oscillator Test Oscillator generates a static frequency or a sine sweep. The latter is a user-defined frequency spectrum tone sweep. The Test Oscillator is useful for tuning studio equipment and instruments. • 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 when used in conjunction with the Anti Aliased button.
