Audio Effects Included with Soundtrack Pro

Soundtrack Pro includes the following types of audio effects:

Dynamics Effects

Dynamics effects let you adjust the dynamic range (the range between the softest and loudest sounds) of your projects. You can use dynamics effects to make sounds more focused and to optimize the sound for specific playback situations. Dynamics effects include compressors, limiters, and noise gates.

Compressors

Compressors work like an automatic volume control, lowering the volume whenever it rises above a certain level, called the threshold. But why would you want to reduce the dynamic level? By cutting the peak levels, the compressor lets you raise the overall volume of the signal. This gives the sound more focus by making the foreground parts stand out while preventing the background parts from becoming lost in the mix. Compression also tends to make sounds tighter or “punchier.” Because the peaks are lower, the maximum volume is reached more quickly.

Figure. Compressor advanced settings window.

In addition, a compressor can make a project sound better when played back in different situations. For example, the speakers on a television set or in a car sound system typically reproduce a narrower dynamic range than does the sound system in a theater. Compressing the overall mix can help make the sound reproduce more clearly in lower-fidelity situations.

Compressors have two main parameters. The threshold lets you set the amplitude above which the compressor lowers the volume. The ratio lets you control the amount by which sounds above the threshold will be lowered, as a percentage of the original signal.

For example, if you set the threshold to –12 dB, and the ratio to 2:1, a sound at –7 dB (5 dB above the threshold) is reduced by 2.5 dB, and a sound at –2 dB (10 dB above the threshold) is reduced by 5 dB.

Compressors can also include parameters for attack and release. These parameters let you set how quickly the compressor reacts once the threshold is reached (for attack) or once the signal falls below the threshold again (for release). Use these parameters to make the compressor’s effect more subtle or more pronounced. Another parameter on some compressors is the knee (or soft knee), which lets you control how gradually the compressor transitions between no compression and the compression ratio at the threshold.

Compressors are typically used on vocal tracks to make the vocals prominent in the overall mix. They can also be used on music and sound effects tracks, but are rarely used on ambience tracks.

Limiters

Limiters (also called peak limiters) prevent the audio signal from exceeding a maximum volume level. A compressor gradually attenuates levels above the threshold, but a limiter puts a hard limit on any signal louder than the threshold, usually at a high ratio. You use a limiter mainly to prevent clipping.

Noise Gates

A noise gate alters the signal in the opposite direction from a compressor. While a compressor lowers the volume of sounds above the threshold, a noise gate lowers the sounds below the threshold. Loud signals pass through unchanged, but softer signals, such as the decay of a loud instrument, are cut off. Noise gates can be used to eliminate low-level noise or hum from an audio signal.

Distortion Effects

Distortion effects simulate the sound of analog and digital distortion. After working to eliminate the digital distortion caused by clipping from a project, why would you add distortion as an effect? The distortion produced by overdriven vacuum tubes (which were used in amplifiers and music recording equipment before the development of digital recording technology) produces an effect which many people find pleasing, and which is integral to many styles of popular music. Analog tube distortion adds a distinctive warmth and bite to the signal.

There are also distortion effects that intentionally cause clipping and digital distortion of the signal. These can be used to modify vocal, music, and other tracks to produce an intense, unnatural effect, or for creating sound effects.

Distortion effects include parameters for tone, which let you shape the way in which the distortion alters the signal, and for gain, which let you control how much the distortion increases the output level of the signal.

EQ and Filter Effects

EQ is likely the most common audio effect used in post-production. You can use EQ to shape the sound of a project by adjusting specific frequencies or frequency ranges. Using EQ, you can create both subtle and extreme changes to the sound of your projects.

Most EQ effects make use of filters. As the name suggests, a filter allows certain frequencies to “pass through” to the output while stopping or attenuating other frequencies. EQ effects include highpass, lowpass, and band pass filters.

When the audio signal passes through an EQ filter, the frequencies that pass through can be raised or lowered in volume. Raising and lowering frequencies using EQ is often referred to as boosting and cutting frequencies. You can create many changes to the sound of your project by boosting and cutting various frequencies.

Figure. Channel EQ advanced settings window.

Frequency Ranges Used with EQ

Sounds can be categorized into one of three basic frequency ranges: bass, midrange, or high (also called treble). These can be further divided to include low bass, low and high midrange, and low and high highs. The following table describes some of the sounds affected by each range:

Name
Frequency range
Description
High High
8–20 kHz
Includes cymbal sounds and highest harmonics of instruments. Boosting frequencies in this range slightly can add sparkle and presence.
High
5–8 kHz
This range corresponds roughly to the treble tone control on a stereo. Boosting frequencies in this range can add brightness and shine.
Low High
2.5–5 kHz
Includes the higher harmonics of voices and musical instruments. This range is important for adding presence. Excessive boosting in this range can sound shrill or harsh.
High Midrange
1.2–2.5 kHz
Includes the consonants of voices and the high harmonics of musical instruments, especially brass instruments. Excessive boosting in this range can create a pinched, nasal sound.
Midrange
750 Hz–1.2 kHz
Includes the vowels of voices and the harmonics of musical instruments that create tone color.
Low Midrange
250–750 Hz
Includes the fundamentals and lower harmonics of voices and musical instruments; careful EQing of each can keep them from competing. Excessive boosting in this range can result in muddy and unclear audio; excessive cutting can produce thin-sounding audio.
Bass
50–250 Hz
Corresponds roughly to the bass tone control on a stereo. Includes the fundamental frequencies of voices and of musical instruments. Excessive boosting in this range can sound boomy and thick.
Low Bass
50 Hz and below
Also calledsub bass. Very little of the sound of voices or musical instruments falls in this range. Many sound effects used in movies, such as explosions and earthquakes, fall in this range.

Note: The frequencies shown for each range are approximate. Any division of sound into frequency ranges is somewhat arbitrary and is meant only to give a general indication of each range.

Roll-Off Filters

The simplest types of EQ effects are roll-off filters, which include lowpass, highpass, bandpass, and shelf filters. Lowpass filters affect all frequencies above a specific frequency, called the cutoff frequency. Frequencies above the cutoff are attenuated or “rolled off” gradually, usually by a fixed number of decibels per octave. Highpass filters, by contrast, affect all frequencies below their cutoff frequency. Bandpass filters exclude all frequencies close to their center frequency. You can set the center frequency, and also set the bandwidth or Q, which specifies how wide a range of frequencies around the center frequency is affected.

These EQs include parameters for setting the cutoff frequency. Shelf filters add parameters to control the gain (the amount of boost or cut). You can use roll-off filters as “broad brush” effects to boost or cut a large range of frequencies.

Graphic EQs

Graphic EQs give you a set of filters (often with 10 or 31 filters), each with a set center frequency and bandwidth. Using a graphic EQ, you can shape a wide variety of frequencies throughout the frequency range. Graphic EQs can be used to shape the sound of the overall project mix.

Parametric EQs

Parametric EQs are similar to bandpass EQs, but provide a greater amount of control, and can be used for extremely precise adjustments. With a parametric EQ, you can set the center frequency, the gain, and the bandwidth. Used carefully, a parametric EQ can help a track cut through the mix, or help a track or project sound fuller. Parametric EQs can also be used to remove specific, unwanted frequencies from a mix.

Modulation Effects

Modulation effects begin with a delayed signal, like time-based effects, but vary (or modulate) the delay time, typically using a low-frequency oscillator (LFO). This can be used to double a sound, making it seem stronger and “fatter,” to simulate a group of voices or instruments playing together, or to add a distinctive character to the sound. Modulation effects include chorus, phase shifters, and flangers.

All of the modulation effects include parameters for the delay rate (also called speed or frequency), which let you set the minimum delay time; depth (also called width or intensity), which you use to set how much the LFO modulates the delay time; and mix, which you use to control the ratio of the effected (wet) signal to the original (dry) signal. They can also include parameters for feedback (or regeneration), which add part of the output back into the input signal.

Chorus

Chorus effects play back multiple repetitions of the delayed signal (like reverbs), but vary the delay time for each one, using an LFO. As the name implies, this effect can strengthen the sound, and create the impression that the sound is being played by many instruments or voices in unison. The slight variations in delay time created by the LFO simulate the subtle differences in timing and pitch heard when several people play together. Using chorus also adds fullness or richness to the signal and can add movement to low or sustained sounds.

Phase Shifters

Phase shifters produce a characteristic “whooshing” sound by combining the original signal with a copy of the signal that is slightly out of phase with the original. This means that the amplitudes of the two signals’ sound waves reach their highest and lowest points at slightly different times. The time between the two signals is modulated, typically using an LFO. As the two signals go in and out of phase, certain frequencies, called notch frequencies, are created, which give phase shifters their distinctive sound.

The main difference between chorus and phase shifting is the amount of delay time. Chorus effects typically use delay times between 20 and 30 milliseconds (ms), while phase shifters (and flangers, discussed next) typically use shorter delay times, between 1 and 10 ms.

Flangers

Flangers work in much the same way as do phase shifters, but additionally change the pitch of the delayed signal slightly. Flanging is typically used to create a more extreme change than phase shifting, sometimes described as adding a “spacey” or “underwater” effect.

Reverb and Delay Effects

Reverbs and delays work by copying a part of the audio signal, delaying it for a brief period of time, and then playing it back with the original signal. The delayed signal can be played back multiple times and can be modified in a variety of ways.

Delay

A delay effect stores the audio signal and then plays back each repetition at a regular rate of time after the original signal. Delays can be used to double individual sounds (for example, making it sound as if a group of instruments is playing the same melody), to achieve echo effects (making it sound as though the sound was occurring in an immense space), and to enhance the stereo position of tracks in a mix. Delay effects are not commonly used on an overall mix except to achieve special effects (such as to create an “otherworldly” sound).

Delay effects let you set the delay time, the time between the original signal and the delayed signal. Delays often provide parameters for feedback (also called regeneration), which let you set how much of the delayed signal is fed back into the delay’s input, creating more repetitions of the delay (like the number of “bounces” in an echo). Specific types of delay have other parameters: tap tempo delays let you set the delay time by physically tapping a key or controller; stereo delays include parameters for the pan position of the output signal, which can be shifted over time using a low-frequency oscillator (called an LFO).

Reverb

Reverberation, usually shortened to reverb, simulates the sound of acoustic environments such as rooms, concert halls, caverns, or the sound of infinite space. In any acoustic space, sounds echo off the surfaces of the space (the floor, walls, and ceiling) over and over, gradually dying out until they become inaudible. Reverb effects consist of thousands of delays, of varying lengths and intensities, that simulate these natural echoes. Reverb helps define the sense of space in which sounds take place and can be used to simulate both realistic and fantastic acoustic environments.

Figure. Space Designer plug-in advanced settings window.

The first form of reverb actually used a room with hard surfaces (called an echo chamber) to add echoes to the signal. Mechanical devices, including plates and springs, were also used to add reverberation to the output of instruments and microphones. Digital sound recording has made it possible to use digital reverbs, which use complex algorithms (sets of equations) to simulate various acoustic environments with greater accuracy and flexibility.

Simple reverb effects provide parameters for the decay time or reverb time, which let you set how long the reverb lasts before dying away, and the mix or level, which you use to set the ratio of the effected signal (called the wet signal) to the original (the dry signal). More sophisticated reverbs can include the following parameters:

  • Room type: Lets you set the type of space the reverb will simulate: a small or large room, a hall, or another type of acoustic space.
  • Predelay time: In an acoustic space, there is a short period of silence between a sound and the time when the initial echoes of the reverb begin. Different spaces have different amounts of predelay, which helps “tell” our ears how large the space is. Longer predelay settings also help separate the original (dry) signal from the effected (wet) signal, making it sound clearer and sometimes larger.
  • Early reflections: The first echoes to arrive from the surrounding surfaces in a space are determined by the size and shape of the space, and “tell” our ears what type of space it is.
  • Diffusion: Lets you set the number of the echoes in the reverb. Hall reverbs typically have low diffusion settings, while plate reverbs typically have high diffusion settings.
  • High-frequency and low-frequency reverb time: These parameters let you specify the decay of higher and lower frequencies separately. Different surfaces, such as wood floors and concrete walls, absorb high and low frequencies at different rates, and these parameters let you simulate the sound of different environments more closely.
  • Reverb envelope: Lets you control how much the volume of the reverb changes over time. In natural acoustic situations, the reverb echoes decay gradually over time. You can re-create this gradual decay, or gate the reverb so that it cuts off more abruptly.

Meters and Diagnostic Effects

Diagnostic effects help you analyze and clean up audio in a variety of ways. Each type of diagnostic effect provides a different way to “look at” an audio clip or file, and each has a unique set of parameters. These effects are available only as realtime effects in the Effects tab and the Mixer, not as processing effects.

Correlation Meter

The Correlation meter displays the phase relationship of a stereo signal. A correlation of +1 (plus one, the far right position) means that the left and right channels “correlate” 100% (that is, they are completely in phase). A correlation of 0 (zero, the center position) indicates the widest permissible left/right divergence, often audible as an extremely wide stereo effect. Correlation values less than zero indicate that out-of-phase material is present, which can lead to phase cancelations if the stereo signal is combined into a monaural signal.

MultiMeter

The MultiMeter combines the functions of the Level Meter and Correlation Meter (as described above) with several other analysis tools:

  • A Spectrum Analyzer

  • A Goniometer for judging the phase coherency in the stereo sound field

The control panel to the left of the display allows you to switch between the Analyzer and Goniometer and contains parameter controls for the MultiMeter. The Stereo Level and Correlation Meter are always visible.

Figure. MultiMeter advanced settings window with Spectrum Analyzer active.
Spectrum Analyzer

The Spectrum Analyzer divides the audio signal into 31 independent frequency bands. Each frequency band represents one third of an octave. The filter curves comply to IEC document 1260.

You turn on the Spectrum Analyzer by clicking the Analyzer button. Turning on the Spectrum Analyzer turns off the Goniometer. The four buttons under the Analyzer button determine what portion of the input signal the Analyzer is displaying. You can choose between Left or Right channel only. LR max shows the maximum band levels of either channel, while Mono displays the levels of the stereo signal summed to mono.

The View options determine the level represented by the top line of the scale in the display (Top; range: -40 to +20 dB) and the overall dynamic range of the Spectrum Analyzer (Range; range: 20 to 80 dB). These two parameters can also be set directly in the display: by dragging directly on the bar graph, you can shift the top line of the display. Dragging directly on the dB scale allows you to compress or expand the scale’s range. The View options are useful when analyzing highly compressed material as you can identify smaller level differences more easily by moving or reducing the display range.

There are three display respond modes: RMS Slow, RMS Fast, and Peak. RMS Slow and RMS Fast modes show the effective signal average (Root Mean Square) and offer a good representation of the perceived volume levels. Peak mode shows level peaks accurately.

Goniometer

The Goniometer helps you to determine the coherence of the stereo image. Using the Goniometer, you can see phase problems as trace cancelations along the center line (M=mid/mono). Goniometers developed when early two channel oscilloscopes first appeared. Users would connect the left and right stereo channels to the X and Y inputs while rotating the display by 45 degrees, resulting in a useful visualization of the signal’s stereo phase.

Figure. MultiMeter advanced settings window with Goniometer active.

The signal trace slowly fades to black, imitating the glow of the tubes found in older Goniometers, and at the same time enhancing readability.

Clicking the Goniometer button turns on the Goniometer and turns off the Spectrum Analyzer. You can use the Auto Gain display parameter in order to obtain a higher readout on low-level passages. Auto Gain allows the display to automatically compensate for low input levels. You can set the amount of compensation with the Auto Gain parameter, or set Auto Gain by dragging directly in the display area of the Goniometer.

Note: Auto Gain is a display parameter only and increases the display for better readability. The actual audio levels are not touched by this parameter.

Miscellaneous Effects

Miscellaneous effects don’t fall into any of the other categories. They include denoising effects, pitch shifting effects, stereo enhancers, bass enhancers, and effects used to transform the sound of vocals. Each effect gives you a different way to modify the audio, and includes a unique set of parameters.

Denoiser

Using the Denoiser, you can eliminate or reduce many kinds of low-level noise (noise floor) from an audio signal. The main parameters of the Denoiser are Threshold, Reduce, and Noise Type. The Threshold parameter sets how high the noise floor is for the audio signal. The recommended method for setting the Threshold is to find a passage where you hear only noise, then set the Threshold so that signals at this volume level are filtered out.

The Reduce parameter sets the level to which the noise floor is reduced. You use the Noise Type parameter to set the type of noise that the Denoiser reduces. There are three choices of noise type:

  • Setting the Noise Type to 0 (zero) causes the Denoiser to reduce “white noise” (all frequencies reduced equally).

  • Setting the Noise Type to a positive value causes the Denoiser to reduce “pink noise” (harmonic noise; greater bass response).

  • Setting the Noise Type to a negative value causes the Denoiser to reduce “blue noise” (hiss, sibilants, tape noise).

The Denoiser recognizes frequency bands with a lower volume and less complex harmonic structure, and then reduces them to the desired dB value. This method is not completely precise, and neighboring frequencies are also reduced. Using the Denoiser at too-high settings can produce the “glass-noise” effect, which is usually less desirable than the existing noise.

There are three smoothing parameters that you can use to minimize the “glass-noise” effect: Frequency smoothing, Time smoothing, and Level smoothing. Raising the Frequency smoothing slider results in a smoother transition of denoising to the neighboring frequencies. When the Denoiser recognizes that only noise is present in a certain frequency band, the higher the Frequency Smoothing parameter is set, the more it will also change the neighboring frequency bands to avoid glass noise.

By adjusting the Time smoothing slider, you can set the amount of time the Denoiser takes to reach maximum noise reduction. By adjusting the Level smoothing slider, you can set a factor for a smoother transition between adjacent volume levels. When the Denoiser recognizes that only noise is present in a certain volume range, the higher the Transition smoothing parameter is set, the more it will also change similar level values to avoid glass noise.

Stereo Spread

The Stereo Spread plug-in is a useful effect for sound design or audio clean-up. It enhances the perception of stereo by extending the stereo base. Some stereo enhancing algorithms function by changing the phase of the signal, which can distort your mix and produce unpredictable results. Instead, the Stereo Spread plug-in extends the stereo base by distributing a selectable number of bands in the middle frequency range alternately left and right. This increases the perception of stereo without causing unnatural-sounding distortion of the mix.

The main parameters of the Stereo Spread plug-in are Order, Upper Intensity (Upper Int.), and Lower Intensity (Lower Int.). The Order parameter determines number of frequency bands into which the signal is divided. The Upper Intensity parameter controls the intensity of the base extension of the upper frequency bands. The Lower Intensity parameter controls the intensity of the base extension of the lower frequency bands.

Human beings perceive stereo placement of sounds mainly in the middle and high frequencies. If very low frequencies are distributed between the left and right speakers, the energy distribution for both speakers will be significantly worse. Therefore, it is always best to select a lower intensity setting for the lower frequency bands, and avoid setting the Lower Freq. below 300 Hz.