Learn about the wide array of audio effects and transitions available in Premiere Pro, what they do, and how and when to use them.

For video effects, see Effects and transitions reference.

Audio effects

Amplitude and Compression

Amplify effect boosts or attenuates an audio signal. Since the effect operates in real time, you can combine it with other effects in the Effects Rack.

  • Gain Sliders: Boost or attenuate individual audio channels.
  • Link Sliders: Moves the channel sliders together.

The Channel Mixer effect alters the balance of stereo or surround channels. You can change the apparent position of sounds, correct mismatched levels, or address phasing issues.

  • Channel tabs: Select the output channel. 
  • Input channel sliders: To mix into the output channel, determine the percentage of the current channels. 
  • Invert: Inverts a channel’s phase. Inverting all channels causes no perceived difference in sound. Inverting only one channel, however, can greatly change the sound.

The Channel Volume effect lets you independently control the volume of each channel in a stereo or 5.1 clip or track. Each channel’s level is measured in decibels.

The DeEsser effect removes sibilance and other high frequency “SSS”-type sounds. These sounds are often created when a narrator or vocalist pronounces the letters “s” and “t.” This effect is available for 5.1, stereo, or mono clip.

  • Mode: Choose Broadband to uniformly compress all frequencies or Multiband to only compress the sibilance range. Multiband is best for most audio content but slightly increases processing time.
  • Threshold: Sets the amplitude above which compression occurs.
  • Center Frequency: Specifies the frequency at which sibilance is most intense. To verify, adjust this setting while playing audio.
  • Bandwidth: Determines the frequency range that triggers the compressor.
  • Output Sibilance Only: Lets you hear detected sibilance. Start playback, and fine-tune settings above.
  • Gain Reduction: Shows the compression level of the processed frequencies.

Dynamics Effects consist of four sections. They are Auto Gate, Compressor, Expander, and Limiter. You can individually control each one of the sections. The LED and gain reduction meters helps you get the overview about how the audio signal is processed.

The different parameters under Dynamic Effects are as follows:

  • Auto Gate: Removes noise below a certain amplitude threshold. The LED meter is green when audio passes through the gate. The meter turns red when there is no audio passing, and yellow during the attack, release, and hold times.
  • Compressor: Reduces the dynamic range of the audio signal by attenuating audio that exceeds a specific threshold. The Ratio parameter can be used control the change in dynamic range while Attack and Release parameter changes the temporal behavior. Use the Gain parameter to increase the audio level after compressing the signal. The Gain Reduction meter shows how much the audio level is reduced.
  • Expander: Increases the dynamic range of the audio signal by attenuating audio below the specified threshold. The ratio parameter can be used to control the change in dynamic range. The Gain Reduction meter shows the level of reduction in audio level.
  • Limiter: Attenuate audio that exceeds a specified threshold. The meter LED turns on when the signal is limited.

Dynamics Processing effect can be used as a compressor, limiter, or expander. As a compressor and limiter, this effect reduces dynamic range, producing consistent volume levels. As an expander, it increases dynamic range by reducing the level of low‑level signals. (With extreme expander settings, you can create a noise gate that totally eliminates noise below a specific amplitude threshold.)

Dynamics

  • Graph: Depicts input level along the horizontal ruler (x‑axis) and the new output level along the vertical ruler (y‑axis). The default graph, with a straight line from the lower left to the upper right, depicts a signal that has been left untouched. Every input level has the same output level. Adjusting the graph changes the relationship between input and output levels, altering dynamic range. For example, if a desirable sonic element occurs around ‑20 dB, you can boost the input signal at that level, but leave everything else unchanged. You can also draw an inverse line (from the upper left to the lower right) that boosts quiet sounds and suppress loud ones.
  • Add point: Adds control point in graph using numerical input and output levels you specify. This method is more precise than clicking the graph to add points.
  • Delete point: Removes selected point from the graph.
  • Invert: Flips the graph, converting compression into expansion, or the other way around.
  • Reset: Resets the graph to its default state.
  • Spline Curves: Creates smoother, curved transitions between control points, rather than more abrupt, linear transitions. For more information, see About spline curves for graphs.
  • Make-up Gain: Boosts the processed signal.

Settings 

General: Provides overall settings.

  • Look Ahead Time: Addresses transient spikes that can occur at the onset of loud signals that extend beyond the compressor’s Attack Time settings. Extending Look-Ahead Time causes compression to attack before the audio gets loud, ensuring that amplitude never exceeds a certain level. Conversely, reducing Look-Ahead Time is desirable to enhance the impact of percussive music like drum hits. 
  • Noise Gating: Completely silences signals that are expanded below a 50-to-1 ratio.

Level Detector: Determines the original input amplitude. 

  • Input Gain: Applies gain to the signal before it enters the Level Detector.
  • Attack Time: Determines how many milliseconds it takes for the input signal to register a changed amplitude level. For example, if audio suddenly drops 30 dB, the specified attack time passes before the input registers an amplitude change. This selection avoids erroneous amplitude readings due to temporary changes. 
  • Release Time: Determines how many milliseconds the current amplitude level is maintained before another amplitude change can register. 
  • Peak mode: Determines levels based on amplitude peaks. This mode is a bit more difficult to use than RMS, because peaks aren’t precisely reflected in the Dynamics graph. However, it can be helpful when audio has loud transient peaks you want to subdue.
  • RMS mode: Determines levels based on the root-mean-square formula, an averaging method that more closely matches the way people perceive volume. This mode precisely reflects amplitudes in the Dynamics graph. For example, a limiter (flat horizontal line) at ‑10 dB reflects an average RMS amplitude of ‑10 dB.

Gain Processor: Amplifies or attenuates the signal depending on the amplitude detected. 

  • Output Gain: Applies gain to the output signal after all dynamics processing.
  • Attack Time: Determines how many milliseconds it takes for the output signal to reach the specified level. For example, if audio suddenly drops 30 dB, the specified attack time passes before the output level changes.
  • Release Time: Determines how many milliseconds the current output level is maintained.
  • Link Channels: Processes all channels equally, preserving the stereo or surround balance. For example, a compressed drum beat on the left channel reduces the right channel level by an equal amount.

Band Limiting: Restricts dynamics processing to a specific frequency range. 

  • Low Cutoff: Is the lowest frequency that dynamics processing affects.
  • High Cutoff: Is the highest frequency that dynamics processing affects.

Hard Limiter effect greatly attenuates audio that rises above a specified threshold. Typically, limiting is applied with an input boost, a technique that increases overall volume while avoiding distortion.

  • Maximum Amplitude: Sets the maximum sample amplitude allowed.
  • Input Boost: Preamplifies audio before you limit it, making a selection louder without clipping it. As you increase this level, compression increases. Try extreme settings to achieve the loud, high‑impact audio heard in contemporary pop music.
  • Look Ahead Time: Sets the amount of time (in milliseconds) for the audio to be attenuated before the loudest peak is hit.
  • Release Time: Sets the time (in milliseconds) for the attenuation to rebound back 12 dB. In general, a setting of around 100 (the default) works well and preserves low bass frequencies.
  • Link Channels: Links the loudness of all channels together, preserving the stereo or surround balance.

Multiband Compressor effect lets you independently compress four different frequency bands. Because each band typically contains unique dynamic content, multiband compression is a powerful tool for audio mastering.

  • Crossover: Sets the crossover frequencies, which determine the width of each band. Either enter specific Low, Midrange, and High frequencies, or drag the crossover markers above the graph.
  • Solo Buttons: Let you hear specific frequency bands. Enable one Solo button at a time to hear bands in isolation, or enable multiple buttons to hear two or more bands together.
  • Bypass Buttons: Bypass individual bands so they pass through without processing.
  • Thresh: Set the input level at which compression begins. Possible values range from ‑60 dB to 0 dB. The best setting depends on audio content and musical style. To compress only extreme peaks and retain more dynamic range, try thresholds around 5 dB below the peak input level. To highly compress audio and greatly reduce dynamic range, try settings around 15 dB below the peak input level.
  • Gain: Boosts or cuts amplitude after compression. Possible values range from ‑18 dB to +18 dB, where 0 is unity gain.
  • Ratio: Sets a compression ratio between 1‑to‑1 and 30‑to‑1. For example, a setting of 3.0 outputs 1 dB for every 3-dB increase above the compression threshold. Typical settings range from 2.0 to 5.0; higher settings produce the compressed sound often heard in pop music.
  • Attack: Determines how quickly compression is applied when audio exceeds the threshold. Possible values range from 0 milliseconds to 500 milliseconds. The default, 10 milliseconds, works well for a wide range of audio. Faster settings work better for audio with fast transients, but such settings sound unnatural for less percussive audio.
  • Release: Determines how quickly compression stops after audio drops below the threshold. Possible values range from 0 milliseconds to 5000 milliseconds. The default, 100 milliseconds, works well for a wide range of audio. Try faster settings for audio with fast transients, and slower settings for less percussive audio.
  • Output Gain: Boosts or cuts overall output level after compression. Possible values range from ‑18 dB to +18 dB, where 0 is unity gain. To reset peak and clip indicators, double‑click the meters.
  • Gain: Boosts or cuts amplitude after compression. Possible values range from ‑18 dB to +18 dB, where 0 is unity gain.
  • Limiter: Applies limiting after Output Gain, at the end of the signal path, optimizing overall levels. Specify Threshold, Attack, and Release settings that are less aggressive than similar band‑specific settings. Then specify a Margin setting to determine the absolute ceiling relative to 0 dBFS.

Options

  • Spectrum On InputDisplays the frequency spectrum of the input signal, rather than the output signal, in the multiband graph. To quickly see the amount of compression applied to each band, toggle this option on and off.
  • Brickwall Limiter: Applies immediate, hard limiting at the current Margin setting. (Deselect this option to apply slower soft limiting, which sounds less compressed but can exceed the Margin setting.) Note: The maximum Attack time for brickwall limiting is 5 ms.
  • Link Band Controls: Lets you globally adjust the compression settings for all bands, while retaining relative differences between bands.

Single-band Compressor effect reduces dynamic range, producing consistent volume levels and increasing perceived loudness. Single-band compression is effective for voiceovers, because it helps the speaker stand out over musical soundtracks and background audio.

Bemærk:

For examples of highly compressed audio, listen to recordings of modern pop music. By contrast, most jazz recordings are lightly compressed, while typical classical recordings feature no compression at all.

  • Threshold: Sets the input level at which compression begins. The best setting depends on audio content and style. To compress only extreme peaks and retain more dynamic range, try thresholds around 5 dB below the peak input level. To highly compress audio and greatly reduce dynamic range, try settings around 15 dB below the peak input level.
  • Ratio: Sets a compression ratio between 1‑to‑1 and 30‑to‑1. For example, a setting of three outputs 1 dB for every 3-dB increase above the threshold. Typical settings range from 2 to 5; higher settings produce the compressed sound often heard in pop music.
  • Attack: Determines how quickly compression starts after audio exceeds the Threshold setting. The default, 10 milliseconds, works well for a wide range of source material. Use faster settings only for audio with quick transients, such as percussion recordings.
  • Release: Determines how quickly compression stops when audio drops below the Threshold setting. The default, 100 milliseconds, works well for a wide range of audio. Try faster settings for audio with fast transients, and slower settings for less percussive audio.
  • Output Gain: Boosts or cuts amplitude after compression. Possible values range from ‑30 dB to +30 dB, where 0 is unity gain.

Tube-modeled Compressor effect simulates the warmth of vintage hardware compressors. Use this effect to add subtle distortion that pleasantly colors audio.

  • Threshold: Sets the input level at which compression begins. Possible values range from ‑60 dB to 0 dB. The best setting depends on audio content and musical style. To compress only extreme peaks and retain more dynamic range, try thresholds around 5 dB below the peak input level. To highly compress audio and greatly reduce dynamic range, try settings around 15 dB below the peak input level.
  • Output Gain: Boosts or cuts overall output level after compression. Possible values range from ‑18 dB to +18 dB, where 0 is unity gain. To reset peak and clip indicators, double‑click the meters.
  • Ratio: Sets a compression ratio between 1‑to‑1 and 30‑to‑1. For example, a setting of 3.0 outputs 1 dB for every 3-dB increase above the compression threshold. Typical settings range from 2.0 to 5.0; higher settings produce the compressed sound often heard in pop music.
  • Attack: Determines how quickly compression is applied when audio exceeds the threshold. Possible values range from 0 milliseconds to 500 milliseconds. The default, 10 milliseconds, works well for a wide range of audio. Faster settings work better for audio with fast transients, but such settings sound unnatural for less percussive audio.
  • Release: Determines how quickly compression stops after audio drops below the threshold. Possible values range from 0 milliseconds to 5000 milliseconds. The default, 100 milliseconds, works well for a wide range of audio. Try faster settings for audio with fast transients, and slower settings for less percussive audio.

Delay and Echo

Analog Delay effect simulates the sonic warmth of vintage hardware delay units. Unique options apply characteristic distortion and adjust the stereo spread. To create discrete echoes, specify delay times of 35 milliseconds or more; to create more subtle effects, specify shorter times.

  • Mode: Specifies the type of hardware emulation, determining equalization and distortion characteristics. Tape and Tube reflect the sonic character of vintage delay units, while Analog reflects later electronic delay lines.
  • Dry Out: Determines the level of original, unprocessed audio.
  • Wet Out: Determines the level of delayed, processed audio.
  • Delay: Specifies the delay length in milliseconds.
  • Feedback: Creates repeating echoes by resending delayed audio through the delay line. For example, a setting of 20% sends delayed audio at one-fifth of its original volume, creating echoes that gently fade away. A setting of 200% sends delayed audio at double its original volume, creating echoes that quickly grow in intensity.
  • Trash: Increases distortion and boosts low frequencies, adding warmth.
  • Spread: Determines the stereo width of the delayed signal.

Delay effect can be used to create single echoes, and various other effects. Delays of 35 milliseconds or more create discrete echoes, while delays between 15‑34 milliseconds can create a simple chorus or flanging effect.

The Multitap Delay effect adds up to four echoes of the original audio in the clip. This effect is available for 5.1, stereo, or mono clips.

Filter and EQ

The Bandpass effect removes frequencies that occur outside the specified range, or band of frequencies. This effect is available for 5.1, stereo, or mono clips.

The Bass effect lets you increase or decrease lower frequencies (200 Hz and below). Boost specifies the number of decibels by which to increase the lower frequencies. This effect is available for 5.1, stereo, or mono clips.

FFT Filter effect makes it easy to draw curves or notches that reject or boost specific frequencies. FFT stands for Fast Fourier Transform, an algorithm that quickly analyzes frequency and amplitude.

This effect can produce:

  • broad high‑ or low‑pass filters (to maintain high or low frequencies) 
  • narrow band‑pass filters (to simulate the sound of a telephone call) 
  • notch filters (to eliminate small, precise frequency bands)
  • Scale: Determines how frequencies are arranged along the horizontal x‑axis:
    • For finer control over low frequencies, select Logarithmic. A logarithmic scale more closely resembles how people hear sound.
    • For detailed, high‑frequency work with evenly spaced intervals in frequency, select Linear.
  • Spline Curves: Creates smoother, curved transitions between control points, rather than more abrupt, linear transitions. For more information, see About spline curves for graphs.
  • Reset: Reverts the graph to the default state, removing filtering.
  • Advanced: Click the triangle to access these settings:
    • FFT Size: Specifies the Fast Fourier Transform size, determining the tradeoff between frequency and time accuracy. For steep, precise frequency filters, choose higher values. For reduced transient artifacts in percussive audio, choose lower values. Values from 1024 through 8192 work well for most material.
    • Window: Determines the Fast Fourier Transform shape, with each option resulting in a different frequency response curve. These functions are listed in order from narrowest to widest. Narrower functions include fewer surrounding, or sidelobe, frequencies but less precisely reflect center frequencies. Wider functions include more surrounding frequencies but more precisely reflect center frequencies. The Hamming and Blackman options provide excellent overall results.

Graphic Equalizer effect boosts or cuts specific frequency bands and provides a visual representation of the resulting EQ curve. Unlike the Parametric Equalizer, the Graphic Equalizer uses preset frequency bands for quick and easy equalization.

You can space frequency bands at the following intervals:

  • One octave (10 bands)
  • One‑half octave (20 bands)
  • One‑third octave (30 bands)

Graphic equalizers with fewer bands provide quicker adjustment; more bands provide greater precision.

  • Gain sliders: Sets the exact boost or attenuation (measured in decibels) for the chosen band.
  • Range: Defines the range of the slider controls. Enter any value between 1.5 dB and 120 dB. (By comparison, standard hardware equalizers have a range of about 12 dB to 30 dB.)
  • Accuracy: Sets the accuracy level for equalization. Higher accuracy levels give better frequency response in the lower ranges, but they require more processing time. If you equalize only higher frequencies, you can use lower accuracy levels.
  • Master Gain: Compensates for an overall volume level that is too soft or too loud after the EQ settings are adjusted. The default value of 0 dB represents no master gain adjustment.

The Highpass effect removes frequencies below the specified Cutoff frequency. The Highpass effects are available for 5.1, stereo, or mono clips.

The Lowpass effect eliminates frequencies above the specified Cutoff frequency. The Lowpass effects are available for 5.1, stereo, or mono clips.

Notch Filter effect removes up to six user‑defined frequency bands. Use this effect to remove narrow frequency bands, such as a 60-Hz hum, while leaving all surrounding frequencies untouched.

Bemærk:

To remove shrill “ess” sounds, use the Sibilance Softener preset. Or use DTMF presets to remove standard tones for analog telephone systems.

  • Frequency: Specifies the center frequency for each notch.
  • Gain: Specifies the amplitude for each notch.
  • Enable: Enable the button to pass without processing.
  • Notch Width: Determines frequency range for all notches. The three options range from Narrow to Super Narrow. Narrow is for a second order filter, which removes some adjacent frequencies. Super Narrow is for a sixth order filter, which is specific.
  • Ultra Quiet: Virtually eliminates noise and artifacts, but requires more processing. This option is audible only on high-end headphones and monitoring systems.
  • Fix Gain to: Determines if notches have equal or individual gain levels.

Parametric Equalizer effect provides maximum control over tonal equalization. It gives you total control over frequency, Q, and gain settings. 

  • Frequency: Sets the center frequency for bands 1-5, and the corner frequencies for the band-pass and shelving filters.
  • Gain: Sets the boost or attenuation for frequency bands, and the per-octave slope of the band-pass filters.
  • Q/ Width: Controls the width of the affected frequency band. Low Q values affect a larger range of frequencies. High Q values (close to 100) affect a narrow band and are ideal for notch filters removing particular frequencies, like 60-Hz hum.
  • Band: Enables up to five intermediate bands, and high-pass, low-pass, and shelving filters, giving you fine control over the equalization curve. To activate the corresponding settings, click the band button. The low and high shelving filters provide slope buttons that adjust the low and high shelves by 12 dB per octave, rather than the default 6 dB per octave.
  • Constant: Describes a frequency band’s width as either a Q value (which is a ratio of width to center frequency) or an absolute width value in Hz. Constant Q is the most common setting.
  • Ultra-Quiet: Virtually eliminates noise and artifacts, but requires more processing. This option is audible only on high-end headphones and monitoring systems.
  • Range: Sets the graph to a 30-dB range for more precise adjustments, or a 96-dB range for more extreme adjustments.

Use the Scientific Filter effect for advanced manipulation of audio. You can also access the effect from the Effects Rack for single assets in the waveform editor, or for tracks and clips in the Multitrack editor.

  • Types: Specifies the type of scientific filter. The available options are as follows.
    • Bessel: Provides accurate phase response with no ringing or overshoot. However, the pass band slopes at its edges, where rejection of the stop band is the poorest of all filter types. These qualities make Bessel a good choice for percussive, pulse-like signals. For other filtering tasks, use Butterworth. 
    • Butterworth: Provides a flat pass band with minimal phase shift, ringing, and overshoot. This filter type also rejects the stop band much better than Bessel and only slightly worse than Chebychev 1 or 2. These overall qualities make Butterworth the best choice for most filtering tasks. 
    • Chebychev: Provides the best stop band rejection but the worst phase response, ringing, and overshoot in the pass band. Use this filter type only if rejecting the stop band is more important than maintaining an accurate pass band. 
    • Elliptical: Provides a sharp cut-off and narrow transition width. It can also notch out frequencies, unlike the Butterworth and Chebychev filters. It can introduce ripples in both the stop band and the pass band
  • Modes: Specify a mode for the filter. The available options are as follows.
    • LowPass: Passes the low frequencies and removes high frequencies. Specify the cutoff point at which the frequencies are removed. 
    • HighPass: Passes high frequencies and removes low frequencies. Specify the cutoff point at which the frequencies are removed.
    • BandPass: Preserves a band, a range of frequencies, while removing all other frequencies. Specify two cutoff points to define the edges of the band.
    • BandStop: Rejects any frequencies within the specified range. Also known as a notch filter, Band Stop is the opposite of Band Pass. Specify two cutoff points to define the edges of the band.
  • Master Gain: Compensates for an overall volume level that might be too loud or too soft after you adjust the filter settings.
  • Cutoff: Defines the frequency that serves as a border between passed and removed frequencies. At this point the filter switches from passing to attenuating, or conversely. In filters requiring a range (Band Pass and Band Stop), Cutoff defines the low frequency border, while High Cutoff defines the high frequency border.
  • High Cutoff: Defines the high frequency border in filters that require a range (Band Pass and Band Stop).
  • Order: Determines the filter’s precision. The higher the order, the more precise the filter (with steeper slopes at the cutoff points, and so on). However, high orders can also have high levels of phase distortion.
  • Transition Bandwidth: (Butterworth and Chebychev only) Sets the width of the transition band. (Lower values have steeper slopes.) If you specify a transition bandwidth, the Order setting is filled in automatically, and conversely. In filters that require a range (Band Pass and Band Stop), this option serves as the lower frequency transition, while High Width defines the higher frequency transition.
  • High width: (Butterworth and Chebychev only) In filters that require a range (Band Pass and Band Stop), this option serves as the higher frequency transition, while Transition Bandwidth defines the lower frequency transition.
  • Stop Attn: (Butterworth and Chebychev only) Determines how much gain reduction to use when frequencies are removed.
  • Pass/Actual ripple: (Chebychev only) Determines the maximum allowable amount of ripple. Ripple is the effect of unwanted boosting and cutting of frequencies near the cutoff point.

The Treble effects let you increase or decrease higher frequencies (4000 Hz and above). The Boost control specifies the amount, measured in decibels, to increase or decrease. This effect is available for 5.1, stereo, or mono clips.

Modulation

Chorus/Flanger effect combines two popular delay-based effects. The Chorus option simulates several voices or instruments played at once by adding multiple short delays with a small amount of feedback. The result is lush, rich sound. Use this effect to enhance vocal tracks or add stereo spaciousness to mono audio.

  • Mode: The following modes are available. 
    • Chorus: Simulates several voices or instruments playing at once.
    • Flanger: Simulates the delayed, phase-shifted sound originally heard in psychedelic music.
  • Speed: Controls the rate at which the delay time cycles from zero to the maximum setting.
  • Width: Specifies the maximum amount of delay.
  • Intensity: Controls the ratio of original to processed audio.
  • Transience: Emphasizes transients, giving them a sharper, more distinct sound.

Flanging is an audio effect caused by mixing a varying, short delay in roughly equal proportion to the original signal. It was originally achieved by sending an identical audio signal to two reel‑to‑reel tape recorders, and then pressing the flange of one reel to slow it down. Combining the two resulting recordings produced a phase‑shifted, time‑delay effect, characteristic of psychedelic music of the 1960s and 1970s. The Flanger effect lets you create a similar result by slightly delaying and phasing a signal at specific or random intervals.

  • Initial Delay Time: Sets the point in milliseconds at which flanging starts behind the original signal. The flanging effect occurs by cycling over time from an initial delay setting to a second (or final) delay setting.
  • Final Delay Time: Sets the point in milliseconds at which flanging ends behind the original signal.
  • Stereo Phasing: Sets the left and right delays at separate values, measured in degrees. For example, 180° sets the initial delay of the right channel to occur at the same time as the final delay of the left channel. You can set this option to reverse the initial/final delay settings for the left and right channels, creating a circular, psychedelic effect.
  • Feedback: Determines the percentage of the flanged signal that is fed back into the flanger. With no feedback, the effect uses only the original signal. With feedback added, the effect uses a percentage of the affected signal from before the current point of playback.
  • Modulation Rate: Determines how quickly the delay cycles from the initial to final delay times, measured either in cycles per second (Hz) or beats per minute (beats). Small setting adjustments produce widely varying effects.
  • Mode: Provides three ways of flanging:
    • Inverted: Inverts the delayed signal, canceling out audio periodically instead of reinforcing the signal. If the Original ‑ Expanded mix settings are set at 50/50, the waves cancel out to silence whenever the delay is at zero.
    • Special Effects: Mixes the normal and inverted flanging effects. The delayed signal is added to the effect while the leading signal is subtracted.
    • Sinusoidal: Makes the transition from initial delay to final delay and back follow a sine curve. Otherwise, the transition is linear, and the delays from the initial setting to the final setting are at a constant rate. If Sinusoidal is selected, the signal is at the initial and final delays more often than it is between delays.
  • Mix: Adjusts the mix of original (Dry) and flanged (Wet) signal. You need some of both signals to achieve the characteristic cancellation and reinforcement that occurs during flanging. With Original at 100%, no flanging occurs at all. With Delayed at 100%, the result is a wavering sound, like a bad tape player.

Similar to flanging, phasing shifts the phase of an audio signal and recombines it with the original, creating psychedelic effects first popularized by musicians of the 1960s. But unlike the Flanger effect, which uses variable delays, the Modulation > Phaser effect sweeps a series of phase-shifting filters to and from an upper frequency. Phasing can dramatically alter the stereo image, creating unearthly sounds.

  • Stages: Specifies the number of phase-shifting filters. A higher setting produces denser phasing effects.
  • Intensity: Determines the amount of phase‑shifting applied to the signal.
  • Depth: Determines how far the filters travel below the upper frequency. Larger settings produce a wider tremolo effect; 100% sweeps from the upper frequency to zero Hz.
  • Mod Rate: Modulation rate controls how fast the filters travel to and from the upper frequency. Specify a value in Hz (cycles per second).
  • Phase Diff: Determines the phase difference between stereo channels. Positive values start phase shifts in the left channel, negative values in the right. The maximum values of +180° and -180° produce a complete difference and are sonically identical.
  • Upper Freq: Sets the upper-most frequency from which the filters sweep. To produce the most dramatic results, select a frequency near the middle of the selected audio’s range.
  • Feedback: Feeds a percentage of the phaser output back to the input, intensifying the effect. Negative values invert phase before feeding audio back.
  • Mix: Controls the ratio of original to processed audio.
  • Output Gain: Adjusts the output level after processing.

Noise Reduction/Restoration

To quickly remove crackle and static from vinyl recordings, use the Automatic Click Remover effect. You can correct a large area of audio or a single click or pop.

  • Threshold: Determines sensitivity to noise. Lower settings detect more clicks and pops but may include audio you want to retain. Settings range from 1 to 100; the default is 30.
  • Complexity: Indicates the complexity of noise. Higher settings apply more processing but can degrade audio quality. Settings range from 1 to 100; the default is 16.

The DeHummer effect removes narrow frequency bands and their harmonics. The most common application addresses power line hum from lighting and electronics. But the DeHummer can also apply a notch filter that removes an overly resonant frequency from source audio.

Bemærk:

To visually adjust root frequency and gain, drag directly in the graph.

  • Frequency: Sets the root frequency of the hum. If you’re unsure of the precise frequency, drag this setting back and forth while previewing audio.
    • Q: Sets the width of the root frequency and harmonics above. Higher values affect a narrower range of frequencies, and lower values affect a wider range.
    • Gain: Determines the amount of hum attenuation.
  • Number of Harmonics: Specifies how many harmonic frequencies to affect.
  • Harmonic Slope: Changes the attenuation ratio for harmonic frequencies.
  • Output Hum Only: Lets you preview removed hum to determine if it contains any desirable audio.

The DeNoise effect reduces or completely removes noise from your audio file. This noise could be unwanted hum and hiss, fans, air conditioner, or any other background noise. You can control the amount of noise reduced using a slider. The values range from 0% to 100% and control the amount of processing applied to the audio signal.

The DeReverb effect estimates the reverberation profile and helps adjust the reverberation amount. The values range from 0% to 100% and control the amount of processing applied to the audio signal.

Reverb

The Convolution Reverb effect reproduces rooms ranging from coat closets to concert halls. Convolution-based reverbs use impulse files to simulate acoustic spaces. The results are incredibly realistic and life-like.

Bemærk:

Because Convolution Reverb requires significant processing, you may hear clicks or pops when previewing it on slower systems. These artifacts disappear after you apply the effect.

  • Impulse: Specifies a file that simulates an acoustic space. Click Load to add a custom impulse file in WAV or AIFF format.
  • Mix: Controls the ratio of original to reverberant sound.
  • Room Size: Specifies a percentage of the full room defined by the impulse file. The larger the percentage, the longer the reverb.
  • Damping LF: Reduces low-frequency, bass-heavy components in reverb, avoiding muddiness and producing a clearer, more articulate sound.
  • Damping HF: Reduces high-frequency, transient components in reverb, avoiding harshness and producing a warmer, lusher sound.
  • Pre-Delay: Determines how many milliseconds the reverb takes to build to maximum amplitude. To produce the most natural sound, specify a short pre-delay of 0–10 milliseconds. To produce interesting special effects, specify a long pre-delay of 50 milliseconds or more.
  • Width: Controls the stereo spread. A setting of 0 produces a mono reverb signal.
  • Gain: Boosts or attenuates amplitude after processing.

The Studio Reverb effect simulates acoustic spaces. It is faster and less processor‑intensive than the other reverb effects, however, because it isn’t convolution‑based. As a result, you can make real‑time changes quickly and effectively in the Multitrack Editor, without pre-rendering effects on a track.

Characteristics

  • Room Size: Sets the room size.
  • Decay: Adjusts the amount of reverberation decay in milliseconds.
  • Early Refections: Controls the percentage of echoes that first reach the ear, giving a sense of the overall room size. Too high a value can result in an artificial sound, while too low a value can lose the audio cues for the room’s size. Half the volume of the original signal is a good starting point.
  • Width: Controls the spread across the stereo channels. 0% produces a mono reverb signal; 100% produces maximum stereo separation.
  • High Frequency Cut: Specifies the highest frequency at which reverb can occur.
  • Low Frequency Cut: Specifies the lowest frequency at which reverb can occur.
  • Damping: Adjusts the amount of attenuation applied to the high frequencies of the reverb signal over time. Higher percentages create more damping for a warmer reverb tone.
  • Diffusion: Simulates the absorption of the reverberated signal as it is reflected off surfaces, such as carpeting and drapes. Lower settings create more echoes, while higher settings produce a smoother reverberation with fewer echoes.

Output Level

  • Dry: Sets the percentage of source audio to output with the effect.
  • Wet: Sets the percentage of reverb to output.

The Surround Reverb effect is primarily intended for 5.1 sources, but it can also provide surround ambience to mono or stereo sources. In the Waveform Editor, you can choose Edit > Convert Sample Type to convert a mono or stereo file to 5.1, and then apply Surround Reverb. In the Multitrack Editor, you can send mono or stereo tracks to a 5.1 bus or master with Surround Reverb.

  • Input Center: Determines the percentage of the center channel included in the processed signal.
  • Input LFE: Determines the percentage of the Low Frequency Enhancement channel used to excite reverb for other channels. The LFE signal itself is not reverberated.
  • Reverb Settings
    • Impulse: Specifies a file that simulates an acoustic space. Click Load to add a custom, 6-channel impulse file in WAV or AIFF format.
    • Room Size: Specifies a percentage of the full room defined by the impulse file. The larger the percentage, the longer the reverb.
    • Damping LF: Reduces low-frequency, bass-heavy components in reverb, avoiding muddiness and producing a clearer, more articulate sound.
    • Damping HF: Reduces high-frequency, transient components in reverb, avoiding harshness and producing a warmer, lusher sound.
    • Pre-Delay: Determines how many milliseconds the reverb takes to build to maximum amplitude. To produce the most natural sound, specify a short pre-delay of 0–10 milliseconds. To produce interesting special effects, specify a long pre-delay of 50 milliseconds or more.
    • Front Width: Controls the stereo spread across the front three channels.A width setting of 0 produces a mono reverb signal.
    • Surround Width: Controls the stereo spread across the rear surround channels (Ls and Rs).
  • Output
    • C Wet Level: Controls the amount of reverb added to the Center channel. (Because this channel usually contains dialog, reverb should typically be lower.)
    • L/R Bal: Controls left-right balance for front and rear speakers. 100 outputs reverb to only the left, -100 to only the right.
    • F/B Bal: Controls front-back balance for left and right speakers. 100 outputs reverb to only the front, -100 to only the back.
    • Mix: Controls the ratio of original to reverberant sound. A setting of 100 outputs only reverb.
    • Gain: Boosts or attenuates amplitude after processing.

Special

Use this effect to use a little gravel or saturation effect to any audio. You can use this effect to simulate blown car speakers, muffled microphones, or overdriven amplifiers.

  • Positive and Negative graphs: Specify separate distortion curves for positive and negative sample values. The horizontal ruler (x‑axis) indicates input level in decibels; the vertical ruler (y‑axis) indicates output level. The default diagonal line depicts an undistorted signal, with a one‑to‑one relationship between input and output values. Click-and-drag to create and adjust points on the graphs. Drag points off a graph to remove them.
  • Reset: Returns a graph to its default, undistorted state.
  • Curve Smoothening: Creates curved transitions between control points, sometimes producing a more natural distortion than the default linear transitions.
  • Time Smoothing: Determines how quickly distortion reacts to changes in input levels. Level measurements are based on low-frequency content, creating softer, more musical distortion.
  • dB Range: Changes the amplitude range of the graphs, limiting distortion to that range.
  • Linear Scale: Changes the amplitude scales of the graphs from logarithmic decibels to normalized values.

The Fill Left with Right effect duplicates the left channel information of the audio clip and places it in the right channel, discarding the original clip’s right channel information.

The Fill Right with Left effect duplicates the right channel information and places it in the left channel, discarding the existing left channel information. Apply to stereo audio clips only.

The Guitar Suite effect applies a series of processors that optimize and alter the sound of guitar tracks. The Compressor stage reduces dynamic range, producing a tighter sound with greater impact. FilterDistortion, and Box Modeler stages simulate common effects that guitarists use to create expressive, artistic performances.

  • Compressor: Reduces dynamic range to maintain consistent amplitude and help guitar tracks stand out in a mix.
  • Filter: Choose an option from this menu, and then set options below:
    • Filter: Simulates guitar filters ranging from resonators to talk boxes.
    • Type: Determines which frequencies are filtered. Specify Lowpass to filter high frequencies, Highpass to filter low frequencies, or Bandpass to filter frequencies above and below a center frequency.
    • Frequency: Determines the cutoff frequency for Lowpass and Highpass filtering, or the center frequency for Bandpass filtering.
    • Resonance: Feeds back frequencies near the cutoff frequency, adding crispness with low settings and whistling harmonics with high settings.
  • Distort: Adds a sonic edge often heard in guitar solos. To change the distortion character, choose an option from the Type menu.
  • Amplifier: Simulates various amplifier and speaker combinations that guitarists use to create unique tones.
  • Mix: Controls the ratio of original to processed audio.

The Invert (audio) effect inverts the phase of all channels. This effect is available for 5.1, stereo, or mono clips.

You can measure the audio level of your clips, tracks, or sequences using the Loudness Radar effect. 

For more information, see Using the Loudness Radar effect.

Mastering describes the complete process of optimizing audio files for a particular medium, such as radio, video, CD, or the web.

Before mastering audio, consider the requirements of the destination medium. If the destination is the web, for example, the file will likely be played over computer speakers that poorly reproduce bass sounds. To compensate, you can boost bass frequencies during the equalization stage of the mastering process.

  • Equalizer: Adjusts the overall tonal balance.
  • Graph: Shows frequency along the horizontal ruler (x‑axis) and amplitude along the vertical ruler (y‑axis), with the curve representing the amplitude change at specific frequencies. Frequencies in the graph range from lowest to highest in a logarithmic fashion (evenly spaced by octaves).
    • Low Shelf Enable: Activate shelving filters at the low end of the frequency spectrum.
    • Peaking Enable: Activates a peaking filter in the center of the frequency spectrum.
    • High Shelf Enable: Activate shelving filters at the high end of the frequency spectrum.
    • Hz: Indicates the center frequency of each frequency band.
    • dB: Indicates the level of each frequency band.
    • Q: Controls the width of the affected frequency band. Low Q values (up to 3) affect a larger range of frequencies and are best for overall audio enhancement. High Q values (6–12) affect a narrow band and are ideal for removing a particular, problematic frequency, like 60-Hz hum.
  • Reverb: Adds ambience. Drag the Amount slider to change the ratio of original to reverberant sound.
  • Exciter: Exaggerates high-frequency harmonics, adding crispness and clarity. 
    • Retro: Adjusts light distortion
    • Tape: Adjusts bright tone
    • Tube: Adjusts quick, dynamic response
    • Amount: Adjust the level of processing
  • Widener: Adjusts the stereo image (disabled for mono audio). Drag the Width slider to the left to narrow the image and increase central focus. Drag the slider to the right to expand the image and enhance spatial placement of individual sounds.
  • Loudness Maximiser: Applies a limiter that reduces dynamic range, boosting perceived levels. A setting of 0% reflects original levels; 100% applies maximum limiting.
  • Output Gain: Determines output levels after processing. For example, to compensate for EQ adjustments that reduce overall level, boost the output gain.

The Swap Channels effect switches the placement of the left and right channel information. Apply to stereo clips only.

Vocal Enhancer effect quickly improves the quality of voice-over recordings. The Male and Female modes automatically reduce sibilance and plosives, as well as microphone handling noise such as low rumbles. Those modes also apply microphone modeling and compression to give vocals a characteristic radio sound. The Music mode optimizes soundtracks so they better complement a voice-over.

  • Male: Optimizes audio for a man’s voice.
  • Female: Optimizes audio for a woman’s voice.
  • Music: Applies compression and equalization to music or background audio.

Stereo Imagery

The Stereo Imagery effect positions and expands the stereo image. Because the Stereo Expander is VST-based, however, you can combine it with other effects in the Mastering Rack and Effects Rack. In Multitrack View, you can also vary the effect over time by using automation lanes.

  • Center Channel Pan: Positions the center of the stereo image anywhere from hard left (-100%) to hard right (100%).
  • Stereo Expand: Expands the stereo image from Narrow/Normal (0) to Wide (300). Narrow/Normal reflects the original, unprocessed audio.

Time and Pitch

The Pitch Shifter effect changes the musical pitch. It's a real-time effect which can be combined with other effects in the mastering rack or the effects rack. In the Multitrack View, you can also vary pitch over time by using automation lanes.

  • Pitch Transpose: Contains options that adjust pitch.
    • Semi-tones: Transposes pitch in semi-tone increments, which equalmusical half-notes (for example, the note C# is one semi-tone higher than C). A setting of 0 reflects the original pitch; +12 semi-tones are an octave higher; -12 semi-tones are an octave lower.
    • Cents: Adjusts pitch in fractions of semi-tones. Possible values range from -100 (one semi-tone lower) to +100 (one semi-tone higher).
    • Ratio: Determines the relationship between shifted and original frequency. Possible values range from 0.5 (an octave lower) to 2.0 (an octave higher).
  • Precision: Determines sound quality.
    • Low Precison: Use the Low setting for 8 bit or low-quality audio.
    • Medium Precison: Use the medium setting for medium quality audio.
    • High Precison: High setting taking longest to process. Use the High setting for professionally recorded audio.
  • Pitch Settings: Control how audio is processed.
    • Splicing Frequency: Determines the size of each chunk of audio data. (The Pitch Shifter effect divides audio into small chunks for processing.) The higher the value, the more precise the placement of stretched audio over time. However, artifacts become more noticeable as values go up. At higher Precision settings, a lower Splicing Frequency may add stutter or echo. If the frequency is too high, sound becomes tinny and voices have a tunnel-like quality.
    • Overlapping: Determines how much each chunk of audio data overlaps with the previous and next ones. If stretching produces a chorus effect, lower the Overlapping percentage. If doing so produces a choppy sound, adjust the percentage to strike a balance between choppiness and chorusing. Values range from 0% to 50%.
    • Use appropriate default settings: Applies good default values for Splicing Frequency and Overlapping.

Obsolete Audio Effects

If your project has an obsolete effect applied, you are prompted to replace the effect when you open the project. To apply the new version of the effect, select Yes.

Audio crossfade transitions

Constant Gain

The Constant Gain crossfade changes audio at a constant rate in and out as it transitions between clips. This crossfade can sometimes sound abrupt.

Constant Power

The Constant Power crossfade creates a smooth, gradual transition, analogous to the dissolve transition between video clips. This crossfade decreases audio for the first clip slowly at first and then quickly toward the end of the transition. For the second clip, this crossfade increases audio quickly at first and then more slowly toward the end of the transition.

Exponential Fade

Exponential Fade fades out the first clip over a smooth logarithmic curve while fading up the second clip, also over a smooth logarithmic curve. Selecting an option from the Alignment control menu, you can specify the positioning of the transition.

Bemærk:

Though the Exponential Fade transition is similar to the Constant Power transition, it is more gradual.