Macromedia Audition - CS5.5 Instruction Manual

Download

Using

ADOBE® AUDITION® CS5.5

Legal notices

For legal notices, see http://help.adobe.com/en_US/legalnotices/index.html.

Last updated 11/7/2011

Chapter 1: What’s new

Chapter 2: Digital audio fundamentals

Understanding sound . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3

Digitizing audio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5

Chapter 3: Workspace and setup

Viewing, zooming, and navigating audio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8

Customizing workspaces . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11

Connecting to audio hardware . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18

Customizing and saving application settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19

Chapter 4: Importing, recording, and playing audio

Creating and opening files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21

Importing with the Files panel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23

Supported import formats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24

Extracting audio from CDs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26

Navigating time and playing audio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27

Recording audio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29

Monitoring recording and playback levels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32

iii

Chapter 5: Editing audio files

Displaying audio in the Waveform Editor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35

Selecting audio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38

Copying, cutting, pasting, and deleting audio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42

Visually fading and changing amplitude . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43

Working with markers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45

Inverting, reversing, and silencing audio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48

Analyzing phase, frequency, and amplitude . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49

Converting sample types . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53

Undo, redo, and history . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55

Chapter 6: Applying effects

Shared effects controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57

Applying effects in the Waveform Editor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60

Applying effects in the Multitrack Editor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61

Adding third-party plug-ins . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63

Chapter 7: Effects reference

Amplitude and compression effects . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65

Delay and echo effects . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 74

Diagnostics effects (Waveform Editor only) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76

Filter and equalizer effects . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 78

Modulation effects . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 81

Last updated 11/7/2011

USING ADOBE AUDITION

Contents

Noise reduction / restoration effects . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 84

Reverb effects . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92

Special effects . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 97

Stereo imagery effects . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100

Time and pitch manipulation effects . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101

Chapter 8: Mixing multitrack sessions

Multitrack Editor overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103

Basic track controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105

Track routing and EQ controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109

Arranging and editing clips . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 114

Looping clips . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 117

Clip volume matching, fading, and mixing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 118

Automating mixes with envelopes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121

Chapter 9: Video and surround sound

Working with video applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 126

Importing video and working with video clips . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 128

5.1 surround sound . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129

Chapter 10: Saving and exporting

Saving and exporting files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132

Viewing and editing XMP metadata . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 136

Chapter 11: Automating common tasks

Favorites . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 139

Match volume across multiple files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 139

Batch process files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 141

Chapter 12: Keyboard shortcuts

Finding and customizing shortcuts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142

Default keyboard shortcuts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 143

Last updated 11/7/2011

Chapter 1: What’s new

Note: Some features in the Windows version of Adobe Audition 3.0 are not available in CS5.5. Examples include CD burning, MIDI, the metronome, some file formats and effects, clip grouping and time stretching, and control surface support. For a complete list, see

Adobe Audition CS5.5 brings the best features from Adobe audio products into a single cross-platform package, with best-in-class editing and multitrack mixing tools, powerful audio sweetening options, and rock-solid performance.

Mac OS support Make the most of the multicore processing, native audio, and DSP power of Apple Macintosh

computers.

High-performance audio engine Multitask efficiently with dramatically improved responsiveness on projects of all

sizes. Open files up to three times faster. Simultaneously work on multiple multitrack sessions and audio files. Import and batch process files in the background while you continue to edit audio. Speed up effects on multiprocessor systems.

Round-trip editing with video applications like Adobe Premiere Pro Tap into audio cleanup and processing tools

directly from Adobe Premiere Pro with roundtrip editing and mixing. Exchange OMF and XML files with digital audio workstations and non-linear editors like Avid Pro Tools and Apple Final Cut Pro. See

applications” on page 126 and “Export sessions to OMF or Final Cut Pro Interchange format” on page 133.

Adobe Audition 3.0 features replaced or not implemented in CS5.5.

“Working with video

Integrated 5.1 surround mixing and editing Mix 5.1 surround directly in the Multitrack Editor. The Track Panner

panel provides intuitive controls and visual feedback that help you locate sounds precisely in the surround field. Open

5.1 mixdown files in the Waveform Editor to edit selected channels. See

Enhanced effects workflows Adjust effect parameters while making selections, playing back audio, or even applying

complex noise reduction. Apply multitrack effects to individual clips. Expand audio processing possibilities with thirdparty VST and Audio Units plug-ins. See

New effects Adobe Audition CS5.5 includes new effects ranging from Surround Reverb, optimized for 5.1 files, to a

suite of Diagnostics effects that correct common audio problems. For more information, see the following:

“Applying effects” on page 57.

“5.1 surround sound” on page 129.

• “Surround Reverb effect” on page 96

• “Vocal Enhancer effect” on page 100

• “Speech Volume Leveler effect” on page 72

• “Single-band Compressor effect” on page 71

• “Diagnostics effects (Waveform Editor only)” on page 76

• “DeHummer effect” on page 89

• “DeEsser effect” on page 65

• “Chorus/Flanger effect” on page 82

• “Phaser effect” on page 84

Expanded library of royalty-free music beds and sound effects Kick-start your soundtrack with more than 10,000

royalty-free files available through the Resource Central panel. Quickly browse and preview files, then simply drag from the panel to audio projects and produce layered, professional soundtracks.

Streamlined metadata workflow with Broadcast Wave support Simplify metadata editing and management with the

XMP-based Metadata panel. XMP support extends to the Broadcast WAV (BWF) format, enabling automated workflows for radio and TV production systems. See

“Viewing and editing XMP metadata” on page 136.

Last updated 11/7/2011

USING ADOBE AUDITION

What’s new

Native XML session format Save multitrack sessions in the flexible XML format, a human-readable standard that

facilitates conversion to proprietary formats used by different manufacturers. Adobe Audition XML sessions can be opened and edited in text editors, or created programmatically from scripts and other tools. See

“Save multitrack

sessions” on page 133.

Multitrack clip volume matching Easily mix audio from diverse sources. See “Match multitrack clip volume” on

page 119.

Multitrack audio analysis Analyze phase relationships and frequency response in real-time using the Phase Meter and

Frequency Analysis panel. See

Simultaneous waveform and spectral views Evaluate audio amplitude and frequency with maximum precision. See

“Analyze phase” on page 49 and “Analyze frequency range” on page 50.

“View audio waveforms and spectrums” on page 35.

Recordable favorites Store combinations of effects, fades, and amplitude adjustments, and quickly reapply them to

any file or selection in the Waveform Editor. See

History panel Easily roll back edits and mixes to earlier states, comparing different effects processing, noise reduction,

signal flow, and more. Recall your original settings with a single click. See

“Favorites” on page 139.

“Undo, redo, and history” on page 55.

Last updated 11/7/2011

Chapter 2: Digital audio fundamentals

Understanding sound

Sound waves

Sound starts with vibrations in the air, like those produced by guitar strings, vocal cords, or speaker cones. These vibrations push nearby air molecules together, raising the air pressure slightly. The air molecules under pressure then push on the air molecules surrounding them, which push on the next set of molecules, and so on. As high-pressure areas move through the air, they leave low-pressure areas behind them. When these waves of pressure changes reach us, they vibrate the receptors in our ears, and we hear the vibrations as sound.

When you see a visual waveform that represents audio, it reflects these waves of air pressure. The zero line in the waveform is the pressure of air at rest. When the line swings up to a peak, it represents higher pressure; when the line swings down to a trough, it represents lower pressure.

A sound wave represented as a visual waveform A. Zero line B. Low-pressure area C. High-pressure area

Waveform measurements

Several measurements describe waveforms:

Amplitude Reflects the change in pressure from the peak of the waveform to the trough. High-amplitude waveforms

are loud; low-amplitude waveforms are quiet.

Cycle Describes a single, repeated sequence of pressure changes, from zero pressure, to high pressure, to low pressure,

and back to zero.

Frequency Measured in hertz (Hz), describes the number of cycles per second. (For example, a 1000-Hz waveform has

1000 cycles per second.) The higher the frequency, the higher the musical pitch.

Phase Measured in 360 degrees, indicates the position of a waveform in a cycle. Zero degrees is the start point,

followed by 90º at high pressure, 180º at the halfway point, 270º at low pressure, and 360º at the end point.

Wavelength Measured in units such as inches or centimeters, is the distance between two points with the same degree

of phase. As frequency increases, wavelength decreases.

Last updated 11/7/2011

USING ADOBE AUDITION

Digital audio fundamentals

180º360

270

A single cycle at left; a complete, 20-Hz waveform at right A. Wavelength B. Degree of phase C. Amplitude D. One second

How sound waves interact

When two or more sound waves meet, they add to and subtract from each other. If their peaks and troughs are perfectly

in phase, they reinforce each other, resulting in a waveform that has higher amplitude than either individual waveform.

In-phase waves reinforce each other.

If the peaks and troughs of two waveforms are perfectly out of phase, they cancel each other out, resulting in no waveform at all.

Out-of-phase waves cancel each other out.

In most cases, however, waves are out of phase in varying amounts, resulting in a combined waveform that is more complex than individual waveforms. A complex waveform that represents music, voice, noise, and other sounds, for example, combines the waveforms from each sound.

Because of its unique physical structure, a single instrument can create extremely complex waves. That’s why a violin and a trumpet sound different even when playing the same note.

Two simple waves combine to create a complex wave.

Last updated 11/7/2011

USING ADOBE AUDITION

Digital audio fundamentals

Digitizing audio

Comparing analog and digital audio

In analog and digital audio, sound is transmitted and stored very differently.

Analog audio: positive and negative voltage

A microphone converts the pressure waves of sound into voltage changes in a wire: high pressure becomes positive voltage, and low pressure becomes negative voltage. When these voltage changes travel down a microphone wire, they can be recorded onto tape as changes in magnetic strength or onto vinyl records as changes in groove size. A speaker works like a microphone in reverse, taking the voltage signals from an audio recording and vibrating to re-create the pressure wave.

Digital audio: zeroes and ones

Unlike analog storage media such as magnetic tape or vinyl records, computers store audio information digitally as a series of zeroes and ones. In digital storage, the original waveform is broken up into individual snapshots called

samples. This process is typically known as digitizing or sampling the audio, but it is sometimes called analog-to-digital conversion.

When you record from a microphone into a computer, for example, analog-to-digital converters transform the analog signal into digital samples that computers can store and process.

Understanding sample rate

Sample rate indicates the number of digital snapshots taken of an audio signal each second. This rate determines the frequency range of an audio file. The higher the sample rate, the closer the shape of the digital waveform is to that of the original analog waveform. Low sample rates limit the range of frequencies that can be recorded, which can result in a recording that poorly represents the original sound.

Two sample rates A. Low sample rate that distorts the original sound wave. B. High sample rate that perfectly reproduces the original sound wave.

To reproduce a given frequency, the sample rate must be at least twice that frequency. For example, CDs have a sample rate of 44,100 samples per second, so they can reproduce frequencies up to 22,050 Hz, which is just beyond the limit of human hearing, 20,000 Hz.

Here are the most common sample rates for digital audio:

Last updated 11/7/2011

USING ADOBE AUDITION

Digital audio fundamentals

Sample rate Quality level Frequency range

11,025 Hz Poor AM radio (low-end multimedia) 0–5,512 Hz

22,050 Hz Near FM radio (high-end multimedia) 0–11,025 Hz

32,000 Hz Better than FM radio (standard broadcast rate) 0–16,000 Hz

44,100 Hz CD 0–22,050 Hz

48,000 Hz Standard DVD 0–24,000 Hz

96,000 Hz Blu-ray DVD 0–48,000 Hz

Understanding bit depth

Bit depth determines dynamic range. When a sound wave is sampled, each sample is assigned the amplitude value closest to the original wave’s amplitude. Higher bit depth provides more possible amplitude values, producing greater dynamic range, a lower noise floor, and higher fidelity. For the best audio quality, remain at 32-bit resolution while transforming audio in Audition, and then convert to a lower bit depth for output.

Bit depth Quality level Amplitude values Dynamic range

8-bit Telephony 256 48 dB

16-bit Audio CD 65,536 96 dB

24-bit Audio DVD 16,777,216 144 dB

32-bit Best 4,294,967,296 192 dB

192 dB

144 dB

96 dB

48 dB

0 dB

Higher bit depths provide greater dynamic range.

8-bit

16-bit 24-bit 32-bit

Measuring amplitude in dBFS

In digital audio, amplitude is measured in decibels below full scale, or dBFS. The maximum possible amplitude is 0 dBFS; all amplitudes below that are expressed as negative numbers. As bit depth increases, the amplitude ruler in Audition increases the decibel range below full scale, reflecting greater dynamic range.

Note: A given dBFS value does not directly correspond to the original sound pressure level measured in acoustic dB.

Audio file contents and size

An audio file on your hard drive, such as a WAV file, consists of a small header indicating sample rate and bit depth, and then a long series of numbers, one for each sample. These files can be very large. For example, at 44,100 samples per second and 16 bits per sample, a mono file requires 86 KB per second—about 5 MB per minute. That figure doubles to 10 MB per minute for a stereo file, which has two channels.

Last updated 11/7/2011

USING ADOBE AUDITION

Digital audio fundamentals

How Adobe Audition digitizes audio

When you record audio in Adobe Audition, the sound card starts the recording process and specifies what sample rate and bit depth to use. Through Line In or Microphone In ports, the sound card receives analog audio and digitally samples it at the specified rate. Adobe Audition stores each sample in sequence until you stop recording.

When you play a file in Adobe Audition, the process happens in reverse. Adobe Audition sends a series of digital samples to the sound card. The card reconstructs the original waveform and sends it as an analog signal through Line Out ports to your speakers.

To sum up, the process of digitizing audio starts with a pressure wave in the air. A microphone converts this pressure wave into voltage changes. A sound card converts these voltage changes into digital samples. After analog sound becomes digital audio, Adobe Audition can record, edit, process, and mix it—the possibilities are limited only by your imagination.

Last updated 11/7/2011

Chapter 3: Workspace and setup

Viewing, zooming, and navigating audio

Comparing the Waveform and Multitrack editors

Adobe Audition provides different views for editing audio files and creating multitrack mixes. To edit individual files, use the Waveform Editor. To mix multiple files and integrate them with video, use the Multitrack Editor.

The Waveform and Multitrack editors use different editing methods, and each has unique advantages. The Waveform Editor uses a destructive method, which changes audio data, permanently altering saved files. Such permanent changes are preferable when converting sample rate and bit depth, mastering, or batch processing. The Multitrack Editor uses a nondestructive method, which is impermanent and instantaneous, requiring more processing power, but increasing flexibility. This flexibility is preferable when gradually building and reevaluating a multilayered musical composition or video soundtrack.

You can combine destructive and nondestructive editing to suit the needs of a project. If a multitrack clip requires destructive editing, for example, simply double-click it to enter the Waveform Editor. Likewise, if an edited waveform contains recent changes that you dislike, use the Undo command to revert to previous states—destructive edits aren’t applied until you save a file.

For more information about the Waveform Editor, see “Editing audio files” on page 35; for more information about the Multitrack Editor, see “Mixing multitrack sessions” on page 103.

Basic components of the editors

Though available options differ in the Waveform and Multitrack editors, both views share basic components, such as the tool and status bars, and the Editor panel.

Last updated 11/7/2011

USING ADOBE AUDITION

Workspace and setup

Basic components of Waveform and Multitrack editors (Waveform shown) A. View buttons and toolbar B. Editor panel with zoom navigator at top C. Various other panels D. Status bar

Switch editors

❖ Do one of the following:

• From the View menu, choose Waveform or Multitrack Editor.

• In the toolbar, click the Waveform or Multitrack Editor button.

• In the Multitrack Editor, double-click an audio clip to open it in the Waveform Editor. Alternatively, double-click

a file in the Files panel.

• In the Waveform Editor, choose Edit > Edit Original to open the multitrack session that created a mixdown file.

(This command requires embedded metadata in the file. See

“Embed edit-original data in exported mixdown files”

on page 127.)

Zoom audio in the Editor panel

To zoom into a specific time range, right-click and drag. A. Zoom navigator B. Timeline ruler

Zoom into a specific time range

In either the zoom navigator or the timeline ruler, right-click and drag. The magnifying glass icon creates a selection showing the range that will fill the Editor panel.

Last updated 11/7/2011

USING ADOBE AUDITION

Workspace and setup

Zoom into a specific frequency range

In the vertical ruler for the spectral display, right-click and drag. (See “View audio waveforms and spectrums” on page 35.)

Extend or shorten the displayed range

Place the pointer over the left or right edge of the highlighted area in the zoom navigator, and then drag the magnifying glass icon

Gradually zoom in or out

In the lower right of the Editor panel, click the Zoom In or Zoom Out button.

You can set the Zoom Factor in the General section of the Preferences dialog box. (See page 19.)

Zoom with the mouse wheel or Mac trackpad

Place the pointer over the zoom navigator or ruler, and either roll the wheel or drag up or down with two fingers. (In the Waveform Editor, this zoom method also works when the pointer is over the waveform.)

Roll or drag over the spectral display, and press Shift to switch between logarithmic and linear frequency scales. (Logarithmic better reflects human hearing; linear makes individual frequencies more visually distinct.)

Magnify selected audio

In the lower right of the Editor panel, click the Zoom In At In Point , Zoom In At Out Point , or Zoom To Selection buttons.

Display the entire audio file or multitrack session

In the lower right of the Editor panel, click the Zoom Out Full button .

To display zoom buttons in a separate panel, choose Window > Zoom.

“Customize preferences

” on

More Help topics

“Keys for playing and zooming audio” on page 143

Navigate through time

At higher zoom levels, you can navigate to different audio content in the Editor panel.

More Help topics

“Monitoring time” on page 27

“Position the current-time indicator” on page 27

“Dock, group, or float panels” on page 12

Last updated 11/7/2011

USING ADOBE AUDITION

Workspace and setup

Navigate by scrolling

Scrolling with the zoom navigator

• In the zoom navigator, drag left or right.

• To scroll through audio frequencies in the spectral display, drag up or down in the vertical ruler. (See “View audio

waveforms and spectrums” on page 35.)

Navigate with the Selection/View panel

The Selection/View panel shows the start and end of the current selection and view in the Editor panel. The panel displays this information in the current time format, such as Decimal or Bars And Beats. (See

format” on page 29.)

1 To display the Selection/View panel, choose Window > Selection/View Controls.

2 (Optional) Enter new values into the Begin, End, or Duration boxes to change the selection or view.

“Change the time display

Customizing workspaces

About workspaces

Adobe video and audio applications provide a consistent, customizable workspace. Although each application has its own set of panels (such as Project, Metadata, and Timeline), you move and group panels in the same way across products.

The main window of a program is the application window. Panels are organized in this window in an arrangement called a workspace. The default workspace contains groups of panels as well as panels that stand alone.

You customize a workspace by arranging panels in the layout that best suits your working style. As you rearrange panels, the other panels resize automatically to fit the window. You can create and save several custom workspaces for different tasks—for example, one for editing and one for previewing.

You can use floating windows to create a workspace more like workspaces in previous versions of Adobe applications, or to place panels on multiple monitors.

Last updated 11/7/2011

USING ADOBE AUDITION

Workspace and setup

Example workspace A. Application window B. Grouped panels C. Individual panel

Choose a workspace

Each Adobe video and audio application includes several predefined workspaces that optimize the layout of panels for specific tasks. When you choose one of these workspaces, or any custom workspaces you’ve saved, the current workspace is redrawn accordingly.

❖ Open the project you want to work on, choose Window > Workspace, and select the desired workspace.

Dock, group, or float panels

You can dock panels together, move them into or out of groups, and undock them so they float above the application window. As you drag a panel, drop zones—areas onto which you can move the panel—become highlighted. The drop zone you choose determines where the panel is inserted, and whether it docks or groups with other panels.

Docking zones

Docking zones exist along the edges of a panel, group, or window. Docking a panel places it adjacent to the existing group, resizing all groups to accommodate the new panel.

Last updated 11/7/2011

USING ADOBE AUDITION

Workspace and setup

Dragging panel (A) onto docking zone (B) to dock it (C)

Grouping zones

Grouping zones exist in the middle of a panel or group, and along the tab area of panels. Dropping a panel on a grouping zone stacks it with other panels.

Dragging panel (A) onto grouping zone (B) to group it with existing panels (C)

Dock or group panels

1 If the panel you want to dock or group is not visible, choose it from the Window menu.

2 Do one of the following:

•

To move an individual panel, drag the gripper area in the upper-left corner of a panel’s tab onto the desired drop zone.

Last updated 11/7/2011

USING ADOBE AUDITION

Workspace and setup

Drag panel gripper to move one panel

• To move an entire group, drag the group gripper in the upper-right corner onto the desired drop zone.

Drag group gripper to move entire group

The application docks or groups the panel, according to the type of drop zone.

Undock a panel in a floating window

When you undock a panel in a floating window, you can add panels to the window and modify it similarly to the application window. You can use floating windows to use a secondary monitor, or to create workspaces like the workspaces in earlier versions of Adobe applications.

❖ Select the panel you want to undock (if it’s not visible, choose it from the Window menu), and then do one of the

following:

• Choose Undock Panel or Undock Frame from the panel menu. Undock Frame undocks the panel group.

• Hold down Ctrl (Windows®) or Command (Mac OS®), and drag the panel or group from its current location.

When you release the mouse button, the panel or group appears in a new floating window.

• Drag the panel or group outside the application window. (If the application window is maximized, drag the

panel to the Windows taskbar.)

Resize panel groups

When you position the pointer over dividers between panel groups, resize icons appear. When you drag these icons, all groups that share the divider are resized. For example, suppose your workspace contains three panel groups stacked vertically. If you drag the divider between the bottom two groups, they are resized, but the topmost group doesn’t change.

Last updated 11/7/2011

USING ADOBE AUDITION

Workspace and setup

To quickly maximize a panel beneath the pointer, press the accent key. (Do not press Shift.) Press the accent key again to return the panel to its original size.

1 Do either of the following:

• To resize either horizontally or vertically, position the pointer between two panel groups. The pointer becomes a

double-arrow

• To resize in both directions at once, position the pointer at the intersection between three or more panel groups.

The pointer becomes a four-way arrow

2 Hold down the mouse button, and drag to resize the panel groups.

Dragging divider between panel groups to resize them horizontally A. Original group with resize icon B. Resized groups

Open, close, and scroll to panels

When you close a panel group in the application window, the other groups resize to use the newly available space. When you close a floating window, the panels within it close, too.

• To open or close a panel, choose it from the Window menu.

• To close a panel or window, click its Close button .

• To see all the panel tabs in a narrow panel group, drag the horizontal scroll bar.

• To bring a panel to the front of a group of panels, do one of the following:

• Click the tab of the panel you want in front.

• Hover the cursor above the tab area, and turn the mouse scroll wheel. Scrolling brings each panel to the front, one

after another.

• Drag tabs horizontally to change their order.

• To reveal panels hidden in a narrow panel group, drag the scroll bar above the panel group.

Last updated 11/7/2011

USING ADOBE AUDITION

Workspace and setup

Drag horizontal scroll bar to see all panels in narrow group

Working with multiple monitors

To increase the available screen space, use multiple monitors. When you work with multiple monitors, the application window appears on one monitor, and you place floating windows on the second monitor. Monitor configurations are stored in the workspace.

More Help topics

“Dock, group, or float panels” on page 12

Display the toolbar

The toolbar provides quick access to tools, the Workspace menu, and buttons that toggle between the Waveform and Multitrack editors. Some tools are unique to each view. Likewise, some Waveform Editor tools are available only in the spectral display.

By default, the toolbar is docked immediately below the menu bar. However, you can undock the toolbar, converting it to the Tools panel, which you can manipulate like any other panel.

• To show or hide the toolbar, choose Window > Tools. A check mark by the Tools command indicates that it is

shown.

• To undock the toolbar from its default location, drag the handle at the left edge to another location in the work area.

• To redock the Tools panel in its default location, drag the Tools panel tab to the drop zone that spans the entire

width of the Adobe Audition window, just under the menu bar.

Available tools differ in each view. A. Waveform Editor tools for spectral display B. Multitrack Editor tools

More Help topics

“Dock, group, or float panels” on page 12

“Comparing the Waveform and Multitrack editors” on page 8

Display the status bar

The status bar runs across the bottom of the Adobe Audition work area. The far left of the status bar indicates the time required to open, save, or process a file, as well as the current transport status (Playing, Recording, or Stopped). The far right of the bar displays various information that you can customize.

Last updated 11/7/2011

USING ADOBE AUDITION

Workspace and setup

A B C D

E G HF

Status bar A. Time to open, save, or process file B. Video Frame Rate C. File Status D. Sample Type E. Uncompressed Audio Size F. Duration G. Free Space H. Detect Dropped Samples

• To show or hide the status bar, choose View > Status Bar > Show. A check mark indicates that the status bar is

visible.

• To change the information displayed at the far right of the bar, choose View > Status Bar, or right-click the bar.

Then select from the following options:

Video Frame Rate Displays the current and target frame rate of open video files in the Multitrack Editor.

File Status Indicates when processing is occurring for effects and amplitude adjustments.

Sample Type Displays sample information about the currently opened waveform (Waveform Editor) or session file

(Multitrack Editor). For example, a 44,100 Hz, 16-bit stereo file is displayed as 44100 Hz • 16-bit • Stereo.

Uncompressed Audio Size Indicates either how large the active audio file would be if saved to an uncompressed format

such as WAV and AIFF, or the total size of a multitrack session.

Duration Shows you the length of the current waveform or session. For example, 0:01:247 means the waveform or

session is 1.247 seconds long.

Free Space Shows how much space is available on your hard drive.

Free Space (Time) Displays the time remaining for recording, based upon the currently selected sample rate. This

value is shown as minutes, seconds, and thousandths of seconds. For example, if Adobe Audition is set to record 8-bit mono audio at 11,025 Hz, the time remaining might read 4399:15.527 free. Change the recording options to 16-bit stereo at 44,100 Hz, and the time remaining becomes 680:44.736 free.

By default, Free Space (Time) information is hidden. To show it, right-click the status bar, and select Free Space (Time) from the pop-up menu.

Detect Dropped Samples Indicates that samples were missing during recording or playback. If this indicator appears,

consider rerecording the file to avoid audible dropouts.

More Help topics

“Basic components of the editors” on page 8

Change interface colors, brightness, and performance

1 Choose Edit > Preferences > Appearance (Windows) or Audition > Preferences > Appearance (Mac OS).

2 Adjust any of the following options, and then click OK:

Presets Applies, saves, or deletes a combination of Colors and Brightness settings.

Colors Click a swatch to change the color of waveforms, selections, or the current-time indicator.

Brightness Brightens or darkens panels, windows, and dialog boxes.

Use Gradients When deselected, removes shadows and highlights from panels, buttons, and meters.

Last updated 11/7/2011

USING ADOBE AUDITION

Workspace and setup

Save, reset, or delete workspaces

Save a custom workspace

As you customize a workspace, the application tracks your changes, storing the most recent layout. To store a specific layout more permanently, save a custom workspace. Saved custom workspaces appear in the Workspace menu, where you can return to and reset them.

❖ Arrange the frames and panels as desired, and then choose Window > Workspace > New Workspace. Type a name

for the workspace, and click

Note: (After Effects, Premiere Pro, Encore) If a project saved with a custom workspace is opened on another system, the application looks for a workspace with a matching name. If it can’t find a match (or the monitor configuration doesn’t match), it uses the current local workspace.

Reset a workspace

Reset the current workspace to return to its original, saved layout of panels.

❖ Choose Window > Workspace > Reset workspace name.

Delete a workspace

1 Choose Window > Workspace > Delete Workspace.

2 Choose the workspace you want to delete, and then click OK.

Note: You cannot delete the currently active workspace.

OK.

Connecting to audio hardware

You can use a wide range of hardware inputs and outputs with Adobe Audition. Sound card inputs let you bring in audio from sources such as microphones, tape decks, and digital effects units. Sound card outputs let you monitor audio through sources such as speakers and headphones.

A. Sound card inputs connect to sources such as microphones and tape decks. B. Sound card outputs connect to speakers and headphones.

Configure audio inputs and outputs

When you configure inputs and outputs for recording and playback, Adobe Audition can use these kinds of sound card drivers:

• In Windows, ASIO drivers support professional cards and MME drivers typically support standard cards.

• In Mac OS, CoreAudio drivers support both professional and standard cards.

Last updated 11/7/2011

USING ADOBE AUDITION

Workspace and setup

ASIO and CoreAudio drivers are preferable because they provide better performance and lower latency. You can also monitor audio as you record it and instantly hear volume, pan, and effects changes during playback.

1 Choose Edit > Preferences > Audio Hardware (Windows) or Audition > Preferences > Audio Hardware (Mac OS).

2 From the Device Class menu, choose the driver for the sound card you want to use.

3 Choose a Default Input and Output from the card.

In the Multitrack Editor, you can override the defaults for specific tracks. See “Assign audio inputs and outputs to

tracks” on page 110.

4 (MME and CoreAudio) For Master Clock, choose the input or output to which you want other digital audio

hardware to synchronize (ensuring accurate alignment of samples).

5 For I/O Buffer Size (ASIO and CoreAudio) or Latency (MME), specify the lowest setting possible without audio

dropouts. The ideal setting depends on the speed of your system, so some experimentation may be necessary.

6 Choose a Sample Rate for the audio hardware. (For common rates for different output mediums, see

“Understanding sample rate” on page 5.)

7 (Optional) To optimize the performance of ASIO and CoreAudio cards, click Settings. For more information,

consult the documentation for the sound card.

Note: By default, Adobe Audition controls ASIO sound cards while playing or monitoring audio. If you want to access the card in another application, select Release ASIO Driver In Background. (Audition still controls the card while recording to avoid having recordings suddenly stop.)

Assign file channels to inputs and outputs

1 Choose Edit > Preferences > Audio Channel Mapping (Windows) or Audition > Preferences > Audio Channel

Mapping (Mac OS).

2 To the far right of items in the Input and Output lists, click the triangles to choose a hardware port for each file

channel.

This procedure also sets default outputs for the Master track in the Multitrack Editor. To override the defaults, see “Assign audio inputs and outputs to tracks” on page 110.

More Help topics

“Monitoring 5.1 surround sound” on page 129

Customizing and saving application settings

Customize preferences

The Preferences dialog box lets you customize Adobe Audition’s display, editing behavior, use of hard disk space, and other settings.

❖ Choose Edit > Preferences (Windows) or Audition > Preferences (Mac OS). Then choose the area you want to

customize.

For information about a particular option, hover the mouse over it until a tooltip appears.

Last updated 11/7/2011

USING ADOBE AUDITION

Workspace and setup

In the Media & Disk Cache preferences, choose your fastest drive for the Primary Temp folder, and a separate drive for the Secondary Temp folder. Select Save Peak Files to store information about how to display WAV files. (Without

peak files, larger WAV files reopen more slowly.)

More Help topics

“Change interface colors, brightness, and performance” on page 17

“Configure audio inputs and outputs” on page 18

“Applying effects” on page 57

“Working with markers” on page 45

“Mixing multitrack sessions” on page 103

“Customize the spectral display” on page 37

“Change the time display format” on page 29

“Navigating time and playing audio” on page 27

Restore preferences to default settings

Unexpected behavior may indicate damaged preferences files. To re-create preferences files, do the following.

❖ Hold down the Shift key, and start Adobe Audition.

Export and import customized application settings

Application settings files store all current preferences, effect settings, and workspaces. Export and import these files to store groups of customized settings for specific workflows, or transfer favorite settings to another machine.

1 Choose File > Export > Application Settings. Then specify a filename and location.

2 To reapply the settings at a later time, choose File > Import > Application Settings.

To import preferences from Audition 2.0 or 3.0, search your system for the audition_settings.xml file. You can import that file into both the Mac and Windows versions of Audition CS.

Last updated 11/7/2011

Chapter 4: Importing, recording, and playing audio

Creating and opening files

Note: Some features in the Windows version of Adobe Audition 3.0 are not available in CS5.5. Examples include some file formats and effects, CD burning, MIDI, the metronome, clip grouping and time stretching, and control surface support. For a complete list, see

Create a new, blank audio file

New, blank audio files are perfect for recording new audio or combining pasted audio.

1 Choose File > New > Audio File.

To quickly create a file from selected audio in an open file, choose Edit > Copy To New. (See “Copy or cut audio data” on page 42.)

Adobe Audition 3.0 features replaced or not implemented in CS5.5.

2 Enter a filename, and set the following options:

Sample Rate Determines the frequency range of the file. To reproduce a given frequency, the sample rate must be at

least twice that frequency. (See

Channels Determines if the waveform is mono, stereo, 5.1 surround.

For voice-only recordings, the mono option is a good choice that results in quicker processing and smaller files.

Bit Depth Determines the amplitude range of the file. The 32-bit level provides maximum processing flexibility in

Adobe Audition. For compatibility with common applications, however, convert to a lower bit depth when editing is complete. (See

“Understanding bit depth” on page 6 and “Change the bit depth of a file” on page 54.)

“Understanding sample rate” on page 5.)

Create a new multitrack session

Session (*.sesx) files contain no audio data themselves. Instead, they are small XML-based files that point to other audio files on the hard drive. A session file keeps track of which files are a part of the session, where they are inserted, which envelopes and effects are applied, and so on.

To examine settings in detail, SESX files can be opened in text editors or stored in version control systems (such as Perforce, which is popular in the gaming industry).

1 Choose File > New > Multitrack Session.

2 Enter a filename and location, and set the following options:

Sample Rate Determines the frequency range of the session. To reproduce a given frequency, the sample rate must be

at least twice that frequency. (See

“Understanding sample rate” on page 5.)

Note: All files added to a session must share the sample rate. If you attempt to import files with different sample rates, Adobe Audition prompts you to resample them, which may lower audio quality. To change resampling quality, adjust the Sample Rate Conversion settings in the Data preferences.

Last updated 11/7/2011

USING ADOBE AUDITION

Importing, recording, and playing audio

Bit Depth Determines the amplitude range of the session, including recordings and files created with the Multitrack >

Mixdown To New File command. (See

“Understanding bit depth” on page 6.)

Important: Choose a bit depth carefully, because it cannot be changed after you create a session. Ideally, you should work at the 32-bit level with CoreAudio and ASIO sound cards, and the 16-bit level with MME cards. If your system performs slowly, try a lower bit depth.

Master Determines whether tracks are mixed down to a mono, stereo, or 5.1 Master track. (See “Routing audio to

buses, sends, and the Master track” on page 110.)

More Help topics

“About multitrack sessions” on page 103

Open existing audio files and multitrack mixes

The following file types open in the Multitrack Editor: Audition Session, Adobe Premiere Pro Sequence XML, Final Cut Pro XML Interchange, and OMF.

All other supported file types open in the Waveform Editor, including the audio portion of video files.

Important: SES session files from previous Audition versions are unsupported. If you have Audition 3.0, save sessions to XML format to open them in CS5.5. Note, however, that effects and time-stretching are excluded.

If you open multiple files, Editor panel menu lets you choose which file to display

1 Choose File > Open.

2 Select an audio or video file. (See “Supported import formats” on page 24.)

If you don’t see the file you want, choose All Supported Media from the menu at the bottom of the dialog box.

Import a file as raw data

If you can’t open a particular file, it may lack necessary header information that describes the sample type. To manually specify this information, import the file as raw data.

1 Choose File > Import > Raw Data.

2 Select the file, and click Open.

3 Set the following options:

Sample Rate Should match the known rate of the file, if possible. For examples of common settings, see

“Understanding sample rate” on page 5. Adobe Audition can import raw data with rates ranging from 1 to 10,000,000 Hz, but playback and recording are supported only between 6000 Hz and 192,000 Hz.

Channels Enter a number from 1 to 32.

Last updated 11/7/2011

USING ADOBE AUDITION

Importing, recording, and playing audio

Encoding Specifies the data storage scheme for the file. If you are unsure what encoding the file uses, consult the

supplier of the file, or the documentation for the application that created it. In many cases, trial and error might be necessary.

Byte Order Specifies the numerical sequence for bytes of data. The Little-Endian method is common to WAV files,

while the Big-Endian method is common to AIFF files. The Default Byte Order automatically applies the default for your system processor and is typically the best option.

Insert an audio file into a multitrack session

When you insert an audio file in the Multitrack Editor, the file becomes an audio clip on the selected track. If you insert several files at once, or a single file that’s longer than the space available on the selected track, Adobe Audition inserts new clips on the nearest empty tracks.

1 In the Multitrack Editor, select a track, and then place the current-time indicator at the desired time position.

2 Choose Multitrack > Insert File.

3 Select an audio or video file. (See “Supported import formats” on page 24.)

More Help topics

“Insert a video file into a multitrack session” on page 128

“Importing with the Files panel” on page 23

“Arranging and editing clips” on page 114

Spot-insert a Broadcast Wave file into a session

When you insert a Broadcast Wave (BWF) file into a multitrack session, Adobe Audition can use the embedded timestamp to insert the file at a specific time. This is commonly called spot-inserting.

1 Choose Edit > Preferences > Multitrack (Windows) or Audition > Preferences > Multitrack (Mac OS).

2 Select Use Embedded Timecode When Inserting Clips Into Multitrack.

3 In the Multitrack Editor, select a track.

4 Choose Multitrack > Insert File, and select a BWF file.

Adobe Audition inserts an audio clip at the designated start time.

To view or edit the timestamp for a BWF clip, open the clip in the Waveform Editor, and then choose Window > Metadata. On the BWF tab, the timestamp value appears as the Time Reference.

More Help topics

“Viewing and editing XMP metadata” on page 136

Importing with the Files panel

The Files panel displays a list of open audio and video files for easy access.

Double-click an empty area of the file list to quickly access the Open File dialog box.

Last updated 11/7/2011

USING ADOBE AUDITION

Importing, recording, and playing audio

Import files into the Files panel

Import files into the Files panel if you want to retain the currently open file in the Editor panel. This technique is particularly helpful when assembling files for a multitrack session.

1 Do either of the following:

• In the Files panel, click the Import File button .

• Choose File > Import > File.

2 Select an audio or video file. (See “Supported import formats” on page 24.)

Insert from the Files panel into a multitrack session

1 In the Files panel, select the files you want to insert.

To select multiple adjacent files, click the first file in the desired range, and then Shift-click the last. To select nonadjacent files, Ctrl-click (Windows) or Command-click (Mac OS).

2 At the top of the Files panel, click the Insert Into Multitrack button . Then choose either New Multitrack

Session (see “Create a new multitrack session” on page 21) or an open session.

The files are inserted on separate tracks at the current time position.

Change displayed metadata in the Files panel

1 In the upper right of the Files panel, click the menu icon , and choose Metadata Display.

2 Select the metadata you want to display, and click OK

3 To move metadata columns left or right, drag column headers such as Name or Duration.

To change the sort order of files, click column headers.

More Help topics

“Viewing and editing XMP metadata” on page 136

Supported import formats

Audio file formats

Adobe Audition can open audio files in the following formats:

• AAC

• AIF, AIFF, AIFC (including files with up to 32 channels)

There are many different variations of AIFF format. Audition can open all uncompressed AIFF files and most common compressed versions.

Note: To see Author metadata in AIFF files, view the Dublin Core: Creator field on the XMP tab of the Metadata panel. (See

“Viewing and editing XMP metadata” on page 136.)

• AU

• AVR

Last updated 11/7/2011

USING ADOBE AUDITION

Importing, recording, and playing audio

• BWF

• CAF

• FLAC

• HTK

• IFF

• M4A

• MAT

• MPC

• mp3 (including mp3-surround files)

• OGA, OGG

• PAF

• PCM

• PVF

• RAW

• RF64

• SD2

• SDS

• SF

• SND

• VOC

• VOX

• W64

• WAV (including files with up to 32 channels)

There are many different variations of WAV format. Adobe Audition can open all uncompressed WAV files and most common compressed versions.

• WVE

• XI

More Help topics

“Saving and exporting files” on page 132

Video file formats

The Waveform Editor lets you open the audio portion of video files in the formats below. The Multitrack Editor lets you insert the same file types and provides a preview in the Video panel.

To access these video formats, QuickTime must be installed. To import additional formats, extend QuickTime support. For more information, see this article on the Apple website.

• AVI

• DV

Last updated 11/7/2011

USING ADOBE AUDITION

Importing, recording, and playing audio

• MOV (including files with up to 32 audio channels)

• MPEG-1

• MPEG-4

• 3GPP and 3GPP2

More Help topics

“Insert a video file into a multitrack session” on page 128

“Export a multitrack mix to Premiere Pro CS5.5” on page 126

“Export sessions to OMF or Final Cut Pro Interchange format” on page 133

Extracting audio from CDs

Extract CD tracks with the Extract Audio From CD command

The Extract Audio From CD command is faster and provides more control, including the ability to optimize drive speed and rename tracks.

1 Place an audio CD in the computer’s CD-ROM drive.

2 Choose File > Extract Audio From CD.

3 For Drive, choose the drive that contains the audio CD.

4 For Speed, choose from all the extraction speeds that the selected drive supports. The Maximum Speed option

usually produces satisfactory results, but if it produces errors, specify a slower speed.

5 Do any of the following:

• To preview a track, click its Play button.

• To include or exclude tracks, click the checkboxes to the left of track numbers, or click Toggle All.

• To rename a track, double-click it.

Extract CD tracks with the Open command (Mac OS)

The Open command lets you extract tracks in AIFF format but requires Audition to continue reading audio data from CD, slowing importing and editing.

1 Place an audio CD in the computer’s CD-ROM drive.

2 Choose File > Open.

3 Choose QuickTime as the file type, and navigate to the CD-ROM drive.

4 Select the tracks you want to extract, and click Open.

Last updated 11/7/2011

USING ADOBE AUDITION

Importing, recording, and playing audio

Navigating time and playing audio

Monitoring time

In the Editor panel, the following features help you monitor time:

• In the timeline near the top of the panel, the current-time indicator lets you start playback or recording at a

specific point.

• In the lower left of the panel, the time display shows the current time in numerical format. The default time format

is Decimal, but you can easily change it. (See used by the timeline.

To show the time display in separate panel, choose Window > Time.

Adobe Audition 3.0 features replaced or not implemented in CS5.5.

“Change the time display format” on page 29.) The same format is

Features that help you monitor time A. Current-time indicator B. Timeline C. Time display

Position the current-time indicator

❖ In the Editor panel, do any of the following:

• In the timeline, drag the indicator or click a specific time point.

• In the time display at lower left, drag across the numbers, or click to enter a specific time.

• At the bottom of the panel, click one of the following buttons:

To display these buttons in a separate panel, choose Window > Transport.

Pause Temporarily stops the current-time indicator. Click the Pause button again to resume playback or

recording.

Move CTI to Previous Places the current-time indicator at the beginning of the next marker. If there are no

markers, the current-time indicator moves to the beginning of the waveform or session.

Rewind Shuttles the current-time indicator backward in time.

Last updated 11/7/2011

USING ADOBE AUDITION

Importing, recording, and playing audio

Right-click the Rewind button to set the rate at which the cursor moves.

Fast Forward Shuttles the current-time indicator forward in time.

Right-click the Fast Forward button to set the rate at which the cursor moves.

Move CTI to Next Moves the current-time indicator to the next marker. If there are no markers, the current-

time indicator moves to the end of the waveform or session.

Preview audio by scrubbing

To scrub audio (producing an audible preview as you shuttle across a file), do any of the following:

• Drag the current-time indicator .

• Press the Rewind or Fast Forward buttons.

• Press the J, K, and L keys to shuttle backward, stop, or shuttle forward. Repeatedly pressing the J or L key gradually

increases shuttle speed. (To change the default, set JKL Shuttle Speed in the Playback preferences.)

Play audio linearly or in a loop

To quickly start and stop playback, press the spacebar.

1 In the Editor panel, position the current-time indicator, or select a range.

2 (Optional) At the bottom of the panel, right-click the Play button , and select one of the following:

Return CTI to Start Position on Stop Reflects the behavior of Audition 3.0 and earlier. (Press Shift+X to toggle this

option on and off.)

Play Spectral Selection Only Plays only frequencies you’ve selected with the Marquee , Lasso , or Paintbrush

Selection tool.

3 (Optional) Click the Loop Playback button if you want to fine-tune a selected range or experiment with

different effects processing.

4 To start playback, click the Play button.

Note: By default, the Editor panel scrolls when playback extends beyond the visible section of a waveform. In the Playback area of the Preferences dialog box, you can disable auto-scrolling.

Synchronize the current-time indicator across files or views

In the Waveform Editor, you can maintain the position of the current-time indicator when you switch between files— a useful technique when editing different versions of the same waveform. In the Multitrack Editor, you can maintain the position of the current-time indicator when you switch to the Waveform Editor—a useful technique when applying edits and effects in both views.

Synchronize the current-time indicator between files in the Waveform Editor

1 Choose Edit > Preferences > General (Windows) or Adobe Audition Preferences > General (Mac OS).

2 Select Synchronize Selection, Zoom Level, and CTI Across Files In The Waveform Editor.

Last updated 11/7/2011

USING ADOBE AUDITION

Importing, recording, and playing audio

Synchronize the current-time indicator between the Multitrack and Waveform Editors

1 Choose Edit > Preferences > Multitrack (Windows) or Adobe Audition Preferences > Multitrack (Mac OS).

2 Select Synchronize Clips With Waveform Editor.

Change the time display format

By default, all audio files and multitrack sessions use the same time display format. To customize the format for an open file or session, choose Window > Properties, expand the Advanced settings, and deselect Synchronize With Time

Display Preferences.

❖ Choose View > Display Time Format, and choose the desired option:

Decimal (mm:ss.ddd) Displays time in minutes, seconds, and thousandths of a second.

Compact Disc 75 fps Displays time in the same format used by audio compact discs, where each second equals 75

frames.

SMPTE 30 fps Displays time in the SMPTE format, where each second equals 30 frames.

SMPTE Drop (29.97 fps) Displays time in the SMPTE drop-frame format, where each second equals 29.97 frames.

SMPTE 29.97 fps Displays time in the SMPTE non-drop-frame format, where each second equals 29.97 frames.

SMPTE 25 fps (EBU) Displays time using the European PAL television frame rate, where each second equals 25 frames.

SMPTE 24 fps (Film) Displays time in a format where each second equals 24 frames, suitable for film.

Samples Displays time numerically, using as a reference the actual number of samples that have passed since the

beginning of the edited file.

Bars and Beats Displays time in a musical measures format of bars:beats:subdivisions. To customize settings, choose

Edit Tempo, and set the following options in the Properties panel:

• Tempo Specifies beats per minute.

• Time Signature Specifies the number of beats per measure, and the note that represents full beats. For example,

with a signature of 3/8, there are three notes per measure, and eighth-notes represent full beats.

• Subdivisions Specifies the number of sections each beat is divided into, or the value after the decimal point. For

example, if you enter 32 subdivisions per beat, a time setting of 4:2:16 represents an eighth note halfway between beats 2 and 3 in 4/4 time.

Custom (X frames per second) Displays time in a custom format. To modify a custom format, choose Edit Custom

Frame Rate, and enter a number of frames per second. Valid values are whole numbers from 2 to 1000.

More Help topics

“Customize start offset and time display for multitrack sessions” on page 105

Recording audio

Note: Some features in the Windows version of Adobe Audition 3.0 are not available in CS5.5. Examples include timed recording, CD burning, MIDI, the metronome, some file formats and effects, and control surface support. For a complete

Adobe Audition 3.0 features replaced or not implemented in CS5.5.

list, see

Last updated 11/7/2011

USING ADOBE AUDITION

Importing, recording, and playing audio

Record audio in the Waveform Editor

You can record audio from a microphone or any device you can plug into the Line In port of a sound card. Before recording, you may need to adjust the input signal to optimize signal-to-noise levels. (See either

levels for standard sound cards” on page 33 or the documentation for a professional card.)

1 Set audio inputs. (See “Configure audio inputs and outputs” on page 18.)

2 Do one of the following:

• Create a new file.

• Open an existing file to overwrite or add new audio, and place the current-time indicator where you want to

start recording.

3 At the bottom of the Editor panel, click the Record button to start and stop recording.

More Help topics

“Create a new, blank audio file” on page 21

“Position the current-time indicator” on page 27

“Monitoring recording and playback levels” on page 32

“Adjust recording

Correct DC offset

Some sound cards record with a slight DC offset, in which direct current is introduced into the signal, causing the center of the waveform to be offset from the zero point (the center line in the waveform display). DC offset can cause a click or pop at the beginning and end of a file.

❖ In the Waveform Editor, choose Favorites > Repair DC Offset.

To measure DC offset, see “Analyze amplitude” on page 51.

Direct-to-file recording in the Multitrack Editor

In the Multitrack Editor, Adobe Audition automatically saves each recorded clip directly to a WAV file. Direct-to-file recording lets you quickly record and save multiple clips, providing tremendous flexibility.

Inside the session folder, you’ll find each recorded clip in the [session name]_Recorded folder. Clip file names begin with the track name, followed by the take number (for example, Track 1_003.wav).

After recording, you can edit takes to produce a polished final mix. For example, if you create multiple takes of a guitar solo, you can combine the best sections of each solo. (See use one version of the solo for a video soundtrack, and another version for an audio CD.

“Trimming and extending clips” on page 115.) Or, you can

Record audio clips in the Multitrack Editor

In the Multitrack Editor, you can record audio on multiple tracks by overdubbing. When you overdub tracks, you listen to previously recorded tracks and play along with them to create sophisticated, layered compositions. Each recording becomes a new audio clip on a track.

1 In the Inputs/Outputs area of the Editor panel, choose a source from a track’s Input menu.

Note: To change the available inputs, choose Audio Hardware, and then click Settings.

2 Click the Arm For Record button for the track.

Last updated 11/7/2011

USING ADOBE AUDITION

Importing, recording, and playing audio

The track meters display the input, helping you optimize levels. (To disable this default and display levels only while recording, deselect Enable Input Metering When Arming Tracks in the Multitrack preferences.)

3 To hear hardware inputs routed through any track effects and sends, click the Monitor Input button .

Note: Routing inputs through effects and sends requires significant processing. To reduce latency (an audible delay) that disrupts timing for performers, see

4 To simultaneously record on multiple tracks, repeat steps 1-3.

5 In the Editor panel, position the current-time indicator at the desired starting point, or select a range for the

“Configure audio inputs and outputs” on page 18.

new clip.

6 At the bottom of the panel, click the Record button to start and stop recording.

More Help topics

“Monitoring recording and playback levels” on page 32

Punch into a selected range in the Multitrack Editor

If you’re dissatisfied with a time range of a recorded clip, you can select that range and punch in a new recording, leaving the original clip intact. Though you can record into a specific range without punching in, punching in lets you hear audio immediately before and after a range; that audio provides vital context that helps you create natural transitions.

For particularly important or difficult sections, you can punch in multiple takes, and then select or edit takes to create the best performance.

A take created by punching in

1 In the Editor panel, drag the Time Selection tool in the appropriate track to select a time range for the clip.

2 Select the correct track input. (See “Assign audio inputs and outputs to tracks” on page 110.)

3 Click the Arm For Record button for the track.

4 Position the current-time indicator a few seconds before the selected range.

5 At the bottom of the Editor panel, click the Record button .

Audition plays the audio preceding the selection, records for the duration of the selected range, and then resumes playback.

Last updated 11/7/2011

USING ADOBE AUDITION

Importing, recording, and playing audio

Punch in during playback in the Multitrack Editor

If you don’t need to punch into a specific range, you can quickly punch into a general area during playback.

1 Enable one or more tracks for recording. (See “Record audio clips in the Multitrack Editor” on page 30.)

2 At the bottom of the Editor panel, click the Play button .

3 When you reach an area where you want to begin recording, click the Record button . When you finish

recording, click the button again.

Choose punch-in takes

If you punch in multiple takes, Audition layers the takes over each other in the Editor panel. To choose between takes, do the following:

1 With the Time Selection tool , select a range that snaps to the start and end of the punch-in takes. (See “Snap to

clip endpoints” on page 114.)

2 In the track, position the mouse over the clip header. (The header displays the track name, followed by take

number.)

3 Drag the topmost take to a different location (typically the end of the session to avoid unwanted playback).

4 Play the session. If you prefer a take you previously moved, drag it back to the selected range.

To mute the original clip for the duration of the punch-in range, adjust the volume envelope. (See “Automating clip

settings” on page 121.)

Monitoring recording and playback levels

Level meters overview

To monitor the amplitude of incoming and outgoing signals during recording and playback, you use level meters. The Waveform Editor provides these meters only in the Levels panel. The Multitrack Editor provides them in both the Levels panel, which shows the amplitude of the Master output, and track meters, which show the amplitude of individual tracks.

You can dock the Levels panel horizontally or vertically. When the panel is docked horizontally, the upper meter represents the left channel, and the lower meter represents the right channel.

To show or hide the panel, choose Window > Level Meters.

A B

Levels panel, docked horizontally A. Left channel B. Right channel C. Peak indicators D. Clip indicators

The meters show signal levels in dBFS (decibels below full scale), where a level of 0 dB is the maximum amplitude possible before clipping occurs. Yellow peak indicators remain for 1.5 seconds so you can easily determine peak amplitude.

C D

Last updated 11/7/2011

USING ADOBE AUDITION

Importing, recording, and playing audio

If amplitude is too low, sound quality is reduced; if amplitude is too high, clipping occurs and produces distortion. The red clip-indicator to the right of the meters lights up when levels exceed the maximum of 0 dB.

To clear clip indicators, either click them individually, or right-click the meters and choose Reset Indicators.

Customize level meters

Right-click the meters and select any of the following options:

Meter Input Signal In the Waveform Editor, displays the level of the default hardware input. (See “Configure audio

inputs and outputs” on page 18.) To quickly enable or disable this option, double-click the meters.

Range options Change the displayed decibel range.

Show Valleys Shows valley indicators at low-amplitude points.

If valley indicators are close to peak indicators, dynamic range (the difference between the quietest and loudest sounds) is low. If the indicators are spread far apart, dynamic range is high.

Show Color Gradient Gradually transitions the meters from green, to yellow, to red. Deselect this option to display

abrupt color shifts to yellow at -18 dBFS, and red at -6.

Show LED Meters Displays a separate bar for each whole decibel level.

Dynamic or Static Peaks Change the mode of peak indicators. Dynamic Peaks resets the yellow peak level indicators

to a new peak level after 1.5 seconds, letting you easily see recent peak amplitude. As the audio gets quieter, the peak indicators recede. Static Peaks retains peak indicators, letting you determine the maximum amplitude of the signal since monitoring, playback, or recording began. However, you can manually reset peak indicators by clicking clip indicators.

To find out how loud audio will get before you record it, choose Static Peaks. Then monitor input levels; the peak indicators show the level of the loudest part.

Adjust recording levels for standard sound cards

Adjust levels if recordings are too quiet (causing unwanted noise) or too loud (causing distortion). To get the best sounding results, record audio as loud as possible without clipping. When setting recording levels, watch the meters, and try to keep the loudest peaks in the yellow range below -3

Adobe Audition doesn’t directly control a sound card’s recording levels. For a professional sound card, you adjust these levels with the mixer application provided with the card (see the card’s documentation for instructions). For a standard sound card, you use the mixer provided by Windows or Mac

Adjust sound card levels in Windows 7 and Vista

1 Right-click the speaker icon in the taskbar, and choose Recording Devices.

2 Double-click the input source you want to use.

3 Click the Levels tab, and adjust the slider as needed.

OS.

Adjust sound card levels in Windows XP

1 Double-click the speaker icon in the taskbar.

2 Choose Options > Properties.

3 Select Recording, and then click OK.

Last updated 11/7/2011

USING ADOBE AUDITION

Importing, recording, and playing audio

4 Select the input source you want to use, and adjust the Volume slider as needed.

Adjust sound card levels in Mac OS

1 Choose System Preferences from the Apple menu.

2 Click Sound, and then click the Input tab.

3 Select the device you want to use, and adjust the Input Volume slider as needed.

Last updated 11/7/2011

Chapter 5: Editing audio files

Displaying audio in the Waveform Editor

View audio waveforms and spectrums

In the Waveform Editor, the Editor panel provides a visual representation of sound waves. Below the panel’s default waveform display, which is ideal for evaluating audio amplitude, you can view audio in the spectral display, which reveals audio frequency (low bass to high treble).

❖ To view the spectral display, do either of the following:.

• In the toolbar, click the Spectral Display button.

• In the Editor panel, drag the divider between the waveform and spectral displays to change the proportion of each.

To instantly show or hide the spectral display, double-click the handle or click the triangle to its right.

Viewing the waveform and spectral displays A. Drag the divider to change the proportion of each. B. Click the triangle to show or hide the spectral display.

To identify specific channels in stereo and 5.1 surround files, note the indicators in the vertical ruler.

More Help topics

“Sound waves” on page 3

“Comparing the Waveform and Multitrack editors” on page 8

About the waveform display

The waveform display shows a waveform as a series of positive and negative peaks. The x-axis (horizontal ruler) measures time, and the y-axis (vertical ruler) measures amplitude—the loudness of the audio signal. Quiet audio has both lower peaks and lower valleys (near the center line) than loud audio. You can customize the waveform display by changing the vertical scale and colors.

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

With its clear indication of amplitude changes, the waveform display is perfect for identifying percussive changes in vocals, drums, and more. To find a particular spoken word, for example, simply look for the peak at the first syllable and the valley after the last.

Stereo file in waveform display

More Help topics

“Change the vertical scale” on page 38

“Change interface colors, brightness, and performance” on page 17

About the spectral display

The spectral display shows a waveform by its frequency components, where the x-axis (horizontal ruler) measures time and the y-axis (vertical ruler) measures frequency. This view lets you analyze audio data to see which frequencies are most prevalent. Brighter colors represent greater amplitude components. Colors range from dark blue (low-amplitude frequencies) to bright yellow (high-amplitude frequencies).

The spectral display is perfect for removing unwanted sounds, such as coughs and other artifacts.

Spectral display, with high frequencies selected

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

More Help topics

“Select spectral ranges” on page 39

“Customize the spectral display” on page 37

“Techniques for restoring audio” on page 84

View layered or uniquely colored waveform channels

For stereo and 5.1 surround files, you can view layered or uniquely colored channels. Layered channels better reveal overall volume changes. Uniquely colored channels help you visually distinguish them.

❖ Choose View > Waveform Channels, and then select Layered or Uniquely Colored.

Channel View options A. Uniquely Colored B. Layered (with Uniquely Colored still selected)

More Help topics

“Change interface colors, brightness, and performance” on page 17

Customize the spectral display

The Spectral Display preferences help you enhance different details and better isolate artifacts.

1 Choose Edit > Preferences > Spectral Display (Windows) or Audition > Preferences > Spectral Display (Mac OS).

2 Set the following options

Windowing Function Determines the Fast Fourier transform shape. These functions are listed in order from

narrowest to widest. Narrower functions include fewer surrounding 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.

Spectral Resolution Specifies the number of vertical bands used to draw frequencies. As you increase resolution,

frequency accuracy increases, but time accuracy decreases. Experiment to find the right balance for your audio content. Highly percussive audio, for example, may be better reflected by low resolution.

To adjust resolution directly in the Editor panel, right-click the vertical ruler next to the spectral display, and choose Increase or Decrease Spectral Resolution.

Decibel Range Changes the amplitude range over which frequencies are displayed. Increasing the range intensifies

colors, helping you see more detail in quieter audio. This value simply adjusts the spectral display; it does not change audio amplitude.

Play Only Selected Frequencies When A Spectral Selection Exists Deselect this option to hear all frequencies in the

same time range as a selection.

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

More Help topics

“About the spectral display” on page 36

“Techniques for restoring audio” on page 84

Change the vertical scale

In the Waveform Editor, you can change the amplitude or frequency scale of the vertical ruler.

Change the amplitude scale of the waveform display

❖ In the waveform display, right-click the vertical ruler and select one of the following:

Decibels Indicates amplitude on a decibel scale that ranges from –Infinity to zero dBFS.

Percentage Indicates amplitude on a percentage scale that ranges from –100% to 100%.

Sample Values Indicates amplitude on a scale that shows the range of data values supported by the current bit

depth. (See

Normalized Values Indicates amplitude on a normalized scale that ranges from –1 to 1.

Change the frequency scale of the spectral display

❖ In the spectral display, right-click the vertical ruler and select one of the following:

More Logarithmic or Linear Gradually displays frequencies in a more logarithmic scale (reflecting human hearing)

or a more linear scale (making high frequencies more visually distinct).

“Understanding bit depth” on page 6.) 32-bit float values reflect the normalized scale below.

Hold down Shift and roll the mouse wheel over the spectral display to show frequencies more logarithmically (up) or linearly (down).

Full Logarithmic or Linear Displays frequencies completely logarithmically or linearly.

More Help topics

“About the waveform display” on page 35

“About the spectral display” on page 36

Selecting audio

Select time ranges

1 In the toolbar, select the Time Selection tool .

2 Do any of the following:

• To select a range, drag in the Editor panel.

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

Dragging to select time ranges

• To extend or shorten a selection, drag the selection edges. (Shift-click beyond the edges to quickly extend a selection

to a specific location.)

Note: If you prefer, you can right-click to extend or shorten a selection. To enable this feature, select Extend Selection in the General section of the Preferences dialog box.

Select spectral ranges

When working in a spectral display, you can use the Marquee, Lasso, or Paintbrush Selection tool to select audio data within specific spectral ranges. The Marquee Selection tool lets you select a rectangular area. The Lasso Selection and Paintbrush Selection tools let you make free-form selections. All three tools allow for detailed editing and processing, including incredible flexibility in audio restoration work. For example, if you find audio artifacts, you can select and edit just the affected frequencies, producing superior results with faster processing.

The Paintbrush Selection tool creates unique selections that determine the intensity of applied effects. To adjust intensity, either layer brush strokes or change the Opacity setting in the toolbar. The more opaque the white, selected area is, the more intense applied effects will be.

AB C

Types of spectral selections A. Marquee B. Lasso C. Paintbrush

1 In the toolbar, select the Marquee , Lasso , or Paintbrush Selection .

2 In the Editor panel, drag in the spectral display to select the desired audio data.

Note: When you make a selection in a stereo waveform, the selection is applied to all channels by default. To select audio data in specific channels, choose them from the Edit

> Enable Channels menu.

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

3 To adjust the selection, do any of the following:

• To move the selection, position the pointer in the selection, and drag it to the desired location.

• To resize the selection, position the pointer on the corner or edge of the selection, and drag it to the desired size.

(For paintbrush selections, you can also adjust the brush Size setting in the toolbar.)

• To add to a lasso or paintbrush selection, Shift-drag. To subtract from the selection, Alt-drag.

• To determine the intensity of effects applied to paintbrush selections, adjust the Opacity setting in the toolbar.

By default, Adobe Audition plays only audio the spectral selection. To hear all audio in the same time range, rightclick the Play button , and deselect Play Spectral Selection Only.

More Help topics

“Techniques for restoring audio” on page 84

“About the spectral display” on page 36

Select artifacts and repair them automatically

For the quickest repair of small, individual audio artifacts like isolated clicks or pops, use the Spot Healing Brush. When you select audio with this tool, it automatically applies the Favorites

> Auto Heal command.

Note: Auto-healing is optimized for small audio artifacts and thus limited to selections of four seconds or less.

1 In the toolbar, select the Spot Healing Brush .

2 To change the pixel diameter, adjust the Size setting. Or press the square bracket keys.

3 In the Editor panel, either click and hold or drag across an audio artifact in the spectral display.

Note: If you click without holding down the mouse button, Audition moves the current-time indicator so you can preview audio, but doesn’t repair it. To repair audio by clicking, select Create A Circular Selection On Mouse Down in the General preferences.

Instantly removing an artifact with the Spot Healing Brush A. Before B. After

More Help topics

“Customize the spectral display” on page 37

“Techniques for restoring audio” on page 84

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

Select all of a waveform

❖ Do either of the following:

• To select the visible range of a waveform, double-click in the Editor panel.

• To select all of a waveform, triple-click in the Editor panel.

Specify which channels you want to edit

By default, Adobe Audition applies selections and edits to all channels of a stereo or surround waveform. However, you can easily select and edit specific channels.

• At the right of the Editor panel, click channel buttons in the amplitude ruler. For a stereo file, for example, click the

left channel

To select one stereo channel simply by dragging across the very top or bottom of the Editor panel, select Allow ContextSensitive Channel Editing in the General section of the Preferences dialog box.

or right channel button.

Selecting specific channels of a 5.1 surround file

Adjust a selection to zero-crossing points

For many editing tasks such as deleting or inserting audio, zero-crossings (points where amplitude is zero) are the best places to make selections. Selections that begin and end at zero-crossings reduce the chance that edits will create audible pops or clicks.

❖ To adjust a selection to the closest zero-crossing points, choose Edit > Zero Crossings. Then select an option such

as Adjust Selection Inward (which moves both edges inward to the next zero crossing).

To further reduce the chance of pops or clicks, all edits are crossfaded. You can change crossfade durations in the Data section of the Preferences dialog box.

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

Snap to markers, rulers, frames, and zero crossings

Snapping causes selection boundaries, as well as the start-time indicator, to move to items such as markers, ruler ticks, zero-crossing points, and frames. Enabling snapping helps you make accurate selections; however, if you prefer, you can disable snapping for specific items.

1 To enable snapping for selected items, click the Toggle Snapping icon at the top of the Editor panel.

2 To specify items to snap to, choose Edit > Snapping, and select any of the following:

Snap To Markers Snaps to a marker point. For information on defining markers, see “Working with markers” on

page 45.

Snap To Ruler (Coarse) Snaps only to the major numeric divisions (such as minutes and seconds) in the timeline.

Note: You can enable only one Snap To Ruler command at a time.

Snap To Ruler (Fine) Snaps to subdivisions (such as milliseconds) in the timeline. Zoom in (right-click and drag across

the timeline) to display more accurate subdivisions and place the cursor more precisely.

Snap To Zero Crossings Snaps to the nearest place where audio crosses the center line (the zero amplitude point).

Snap To Frames Snaps to a frame boundary if the time format is measured in frames (such as Compact Disc and

SMPTE).

You can access snapping commands by right-clicking the timeline.

More Help topics

“Snap to clip endpoints” on page 114

“Snap to loop beats” on page 118

Copying, cutting, pasting, and deleting audio

Note: Some features in the Windows version of Adobe Audition 3.0 are not available in CS5.5. Examples include Audition-specific clipboards, CD burning, MIDI, the metronome, some file formats and effects, and control surface support. For a complete list, see

Copy or cut audio data

1 In the Waveform Editor, select the audio data you want to copy or cut. Or, to copy or cut the entire waveform,

deselect all audio data.

2 Choose one of the following:

• Edit > Copy to copy audio data to the clipboard.

• Edit > Copy To New to copy and paste the audio data into a newly created file.

• Edit > Cut to remove audio data from the current waveform and copy it to the clipboard.

Adobe Audition 3.0 features replaced or not implemented in CS5.5.

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

Paste audio data

❖ Do either of the following:

• To paste audio into the current file, place the current-time indicator where you want to insert the audio or select

existing audio you want to replace. Then choose Edit > Paste.

• To paste audio data into a new file, choose Edit > Paste To New. The new file automatically inherits the sample type

(rate and bit depth) from the original clipboard material.

Mix audio data when pasting

The Mix Paste command mixes audio data from the clipboard with the current waveform.

1 In the Editor panel, place the current-time indicator where you want to start mixing the audio data. Alternately,

select the audio data you want to replace.

2 Choose Edit > Mix Paste.

3 Set the following options:

Copied and Existing Audio Adjust the percentage of each.

Invert Copied Audio Reverses the phase of copied audio, either exaggerating or reducing phase cancellation if the

existing audio contains similar content. (To understand phase cancellation, see page 4.)

“How sound waves interact” on

Modulate Modulates the amount of copied and existing audio, producing more audible variation.

Crossfade Applies a crossfade to the beginning and end of the pasted audio, producing smoother transitions. Specify

the fade length in milliseconds.

More Help topics

“Convert a waveform between surround, stereo, and mono” on page 53

“Channel Mixer effect” on page 65

Delete or crop audio

❖ Do one of the following:

• Select audio you want to delete, and choose Edit > Delete.

• Select audio you want to keep, and choose Edit > Crop. (Unwanted audio at the beginning and end of the file is

removed.)

Visually fading and changing amplitude

Though various effects can change amplitude or produce fades, visual fade and gain controls make the task quick and intuitive. As you drag these controls in the Editor panel, a preview helps you precisely adjust audio.

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

Visual controls in the Editor panel A. Fade controls B. Gain control (heads-up display)

To quickly fade selected audio, choose Favorites > Fade In or Fade Out.

Visually fade in or out

Adobe Audition offers three types of visual fades:

• Linear fades produce an even volume change that works well for much material. If this fade sounds too abrupt,

however, try one of the other options.

• Logarithmic fades smoothly change volume slowly and then rapidly, or vice versa.

• Cosine fades are shaped like an S-curve, changing volume slowly at first, rapidly through the bulk of the fade, and

slowly at the finish.

Note: In the Waveform Editor, fades permanently change audio data. To apply fades you can readjust in the Multitrack Editor, see

“Fade or crossfade multitrack clips” on page 119.

Fade types A. Linear B. Logarithmic C. Cosine

❖ In the upper left or right of the waveform, drag the Fade In or Fade Out handle inward, and do any of the

following:

• For a linear fade, drag perfectly horizontally.

• For a logarithmic fade, drag up or down.

• For a cosine (S-curve) fade, hold down Ctrl (Windows) or Command (Mac OS).

To create cosine fades by default and hold the keys above to create linear or logarithmic fades, change the Default Fade setting in the General preferences.

More Help topics

“Volume Envelope effect (Waveform Editor only)” on page 73

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

Visually raise or lower amplitude

1 In the Editor panel, select specific audio, or select nothing to adjust the entire file.

2 In the gain control that floats above the panel, drag the knob or numbers.

The numbers indicate how new amplitude compares with existing amplitude. When you release the mouse button, the numbers return to 0 dB, so you can make further adjustments.

Changing the volume of selected area

More Help topics

“Amplify effect” on page 65

Pin or hide the visual amplitude control

By default, the visual amplitude control appears in a heads-up display (HUD) that floats over all waveforms. If you find the HUD distracting, do any of the following:

• To lock the HUD in one location, click the Pin button .

• To show the HUD only over highlighted selections, select Show HUD for Selection Ranges Only in the General

preferences.

• To totally hide the HUD, deselect View > Show HUD.

Working with markers

Markers (sometimes called cues) are locations that you define in a waveform. Markers make it easy to navigate within a waveform to make a selection, perform edits, or play back audio.

In Adobe Audition, a marker can be either a point or a range. A point refers to a specific time position within a waveform (for instance, 1:08.566 from the start of the file). A range has both a start time and an end time (for example, all of the waveform from 1:08.566 to 3:07.379). You can drag start and end markers for a range to different times.

In the timeline at the top of the Editor panel, markers have white handles you can select, drag, or right-click to access additional commands.

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

Examples of markers

A. Marker point B. Marker range

Note: To preserve markers when you save a file, select Include Markers and Other Metadata.

Add, select, and rename markers

Though you can add markers directly in the Editor panel, you use the Markers panel (Windows > Markers) to define and select markers.

To hide or show information such as Duration and Type, choose Markers Display from the panel menu .

Add a marker

1 Do one of the following:

• Play audio.

• Place the current-time indicator where you want a marker point to be.

• Select the audio data you want to define as a marker range.

2 Either press the M key, or click the Add Marker button in the Markers panel.

To automatically create markers where silence occurs, see “Delete Silence and Mark Audio options” on page 78.

Select markers

• Click a marker in the Editor or Markers panel. Or double-click to move the current-time indicator to that

location and select the area for range markers.

• To select adjacent markers, click the first marker you want to select in the Markers panel, and then Shift-click the

last.

• To select nonadjacent markers, Ctrl-click (Windows) or Command-click (Mac OS) them in the Markers panel.

• To move the current-time indicator to the nearest marker, choose Edit > Marker > Move CTI to Next or Previous.

Rename a marker

1 In the Markers panel, select the marker.

2 Click the marker name, and enter a new name.

Adjust, merge, convert, or delete markers

After creating markers, you can fine-tune them to best address the needs of an audio project.

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

Reposition markers

• In the Editor panel, drag marker handles to a new location.

• In the Markers panel, select the marker, and enter new Start values for point markers, or Start, End, and Duration

values for range markers.

Merge individual markers

❖ In the Markers panel, select the markers you want to merge, and click the Merge button .

The new merged marker inherits its name from the first marker. Merged point markers become range markers.

Convert a point marker to a range marker

❖ Right-click the marker handle, and choose Convert to Range.

The marker handle splits into two handles.

Convert a range marker to a point marker

❖ Right-click a marker handle, and choose Convert to Point.

The two parts of the range marker handle merge into a single handle, with the start time of the range becoming the time for the point marker.

Delete markers

• Select one or more markers, and click the Delete button in the Markers panel.

• Right-click the marker handle in the Editor panel, and choose Delete Marker.

Save audio between markers to new files

1 In the Waveform Editor, choose Window > Markers.

2 Select one or more marker ranges. (See “Working with markers” on page 45 .)

3 Click the Export Audio button in the Markers panel.

4 Set the following options:

Use Marker Names In Filename Uses the marker name as the prefix for the filename.

Prefix Specifies a filename prefix for the new files.

Postfix Starting # Specifies the number to begin with when adding numbers to the filename prefix. Adobe Audition

automatically adds numbers after the prefix (for example, prefix02, prefix03) to distinguish saved files.

Location Specifies the destination folder for saved files. Click Browse to specify a different folder.

Format Specifies the file format. The Format Settings area below indicates and data compression and storage

modes; to adjust these, click Change. (See

Sample Type Indicates the sample rate and bit depth. To adjust these options, click Change. (See “Convert the

sample rate of a file” on page 53.)

“Audio format settings” on page 135.)

Include Markers and Other Metadata Includes audio markers and information from the Metadata panel in

processed files. (See

“Viewing and editing XMP metadata” on page 136.)

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

More Help topics

“Delete Silence and Mark Audio options” on page 78

“Batch process files” on page 141

Creating playlists

A playlist is an arrangement of marker ranges that you can play back in any order and loop a specified number of times. A playlist lets you try different versions of an arrangement before you commit to edits. You create playlists in the Playlist panel (Window

Important: To store a playlist with a file, you must save in WAV format. (See “Save audio files” on page 132.)

Create a playlist

1 In the Playlist panel, click the Open Markers Panel button .

2 In the Markers panel, select marker ranges you want to add to the playlist. Then click the Insert Selected Range

Markers Into Playlist button

Change the order of items in a playlist

❖ Drag the item up or down.

> Playlist).

, or drag the range markers to the Playlist panel.

Play items in a playlist

• To play all or part of the list, select the first item you want to play. Then click the Play button at the top of the

panel.

• To play a specific item, click the Play button to the left of the item name.

Loop an item in a playlist

❖ Select an item, and enter a number in the Loops column. Each item can loop a different number of times.

Delete items from a playlist

❖ Select the items, and click the Remove button .

More Help topics

“Working with markers” on page 45

Inverting, reversing, and silencing audio

Invert a waveform

The Invert effect inverts audio phase by 180 degrees. (To understand phase degrees, see “Waveform measurements” on page 3.)

Inverting doesn’t produce an audible change on an individual waveform, but you can hear a difference when combining waveforms. For example, you might invert pasted audio to better align it with existing audio. Or, you could invert one channel of a stereo file to correct an out-of-phase recording.

1 If you want to invert part of a waveform, select the desired range. Or, deselect all audio data to invert the entire

waveform.

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

2 Choose Effects > Invert.

More Help topics

“Waveform measurements” on page 3

Reverse a waveform

The Reverse effect reverses a waveform from right to left so it plays backwards. Reversing is useful for creating special effects.

1 If you want to reverse part of the waveform, select the desired range. Or, deselect all audio data to reverse the entire

waveform.

2 Choose Effects > Reverse.

Create silence

Creating silence is useful for inserting pauses and removing nonessential noise from an audio file. Adobe Audition provides two ways to create silence:

• To mute existing audio in the Waveform Editor, select the desired content, and choose Effects > Silence. Unlike

deleting or cutting a selection, which splices the surrounding material together, muting leaves the duration of the selection intact.

• To add silence in the Waveform or Multitrack Editor, either position the current-time indicator or select

existing audio. Then choose Edit > Insert >Silence, and enter the number of seconds. Any audio to the right is pushed out in time, lengthening duration. Multitrack clips are split if necessary.

More Help topics

“Delete Silence and Mark Audio options” on page 78

Analyzing phase, frequency, and amplitude

Adobe Audition provides several ways to analyze audio. To compare phase relationships between any two channels, use the Phase Meter panel. To analyze tonal and dynamic range, use the Frequency Analysis and Amplitude Statistics panels.

The Waveform Editor also provides Spectral Frequency Display, which you can use together with the analysis methods above. (See

Analyze phase

The Phase Meter panel reveals out-of-phase channels for stereo and surround waveforms, which you can address with the Effects > Invert command. (See channels that will sound similar if summed to mono. (See “Convert a waveform between surround, stereo, and mono” on page 53.)

To understand audio phase, see “How sound waves interact” on page 4.

1 Choose Window > Phase Meter.

“Displaying audio in the Waveform Editor” on page 35.)

“Invert a waveform” on page 48.) This panel also helps you identify highly in-phase

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

2 Right-click the Phase Meter panel, and choose channels from the Channel and Compare To menus.

3 In Editor panel, select a range if desired, and start playback.

In the Phase Meter, audio to the left is more out of phase, while audio to the right is more in phase. -1.0 reflects total phase cancellation, while 1.0 reflects identical audio content in each channel.

To customize meter appearance, right-click them, and select Show Color Gradient or Show LED Meters.

Analyze frequency range

You can use the Frequency Analysis panel to identify problematic frequency bands, which you can then correct with a filter effect.

1 Choose Window > Frequency Analysis.

2 In the Editor panel, click a time point, select a range, or start playback.

3 In the Frequency Analysis panel, view frequency along the horizontal axis, and amplitude along the vertical axis.

If you selected a range, Adobe Audition analyzes only the center point. To analyze the overall frequency of the range, click Scan Selection.

Frequency Analysis options

Scale Displays the frequency scale either logarithmically (reflecting human hearing) or linearly (providing more detail

for upper frequencies).

Copy All Graph Data Copies a text report of the frequency data to the system clipboard.

Hold buttons Let you take up to eight frequency snapshots as a waveform is playing. The frequency outline (which is

rendered in the same color as the button clicked) is frozen on the graph and overlaid on other frequency outlines. To clear a frozen frequency outline, click its corresponding Hold button again.

Display Changes the graph display. Choose one of the following styles:

• Lines Displays amplitude at each frequency with simple lines. By default, the left channel is green; the right is blue.

• Area Also displays lines for amplitude, but fills the area beneath the lines in a solid color, and smooths out

amplitude differences in the same area.

• Bars Shows the effect of analysis resolution by splitting the display into rectangular segments. The higher the FFT

size, the greater the analysis resolution, and the narrower the bars.

Top Channel Determines which channel of a stereo or surround file appears over others in the graph. To combine

displayed channels, choose Average.

Scan or Scan Selection Scans the entire file or selection, and displays average frequency data in the graph. (By default,

the graph displays data from the center point of files and selections.)

Advanced options

FFT Size Specifies the Fast Fourier Transform size. Higher FFT sizes report frequency data more accurately but they

require longer processing times.

Window Determines the Fast Fourier transform shape. These functions are listed in order from narrowest to widest.

Narrower functions include fewer surrounding 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.

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

0 dB Reference Determines the amplitude at which full scale, 0 dBFS audio data is displayed. For example, a value of

zero displays 0 dBFS audio at 0 dB. A value of 30 displays 0 dBFS audio at –30 dB. This value simply moves the graph up or down; it does not change the amplitude of audio data.

Adjust the 0 dB Reference to calibrate this display to another decibel reference, like sound pressure level (SPL).

Value at [x] Hz Reveals precise amplitude for specific frequencies when you position the mouse over the graph.

Overall Frequency For the start point of a selected range, indicates average frequency.

Overall Musical Note For the start point of a selected range, indicates keyboard position and variance from standard

tuning (A440). For example, A2 +7 equals the second-lowest A on a keyboard tuned 7% higher than normal.

Zoom frequency graphs

In the Frequency Analysis panel, you can zoom graphs to analyze frequency in more detail.

❖ Do any of the following:

• To zoom in on a graph, right-click and drag the magnifying glass icon in the vertical or horizontal ruler.

• To navigate a magnified graph, left-click and drag the hand icon in the vertical or horizontal ruler.

• To zoom out on a magnified graph, right-click in the vertical or horizontal ruler, and choose Zoom Out to return

to the previous magnification, or Zoom Out Full to zoom out completely.

Zooming and navigating a Frequency Analysis graph

Analyze amplitude

1 In the Waveform Editor, choose Window > Amplitude Statistics.

2 To calculate statistics from an entire file or selection, click Scan or Scan Selection. (By default, statistics are

calculated from the center point of files and selections.)

You can adjust a selection in the Editor panel. Click Scan Selection again to recalculate statistics.

3 Evaluate amplitude on the following tabs:

• The General tab displays numerical statistics that indicate dynamic range, identify clipped samples, and note any

DC offset.

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

• The RMS Histogram tab displays a graph that shows the relative prevalence of each amplitude. The horizontal ruler

measures amplitude in decibels, and the vertical ruler measures prevalence using the RMS formula. Choose a channel to display from the Show Channel menu.

Use the Histogram tab to identify prevalent amplitudes, and then compress, limit, or normalize them with an amplitude effect.

General options

Click the icons to the right of values to navigate to the corresponding location in the file.

Peak Amplitude Shows the sample with the highest amplitude in decibel form.

Maximum Sample Value Shows the sample with the highest amplitude.

Minimum Sample Value Shows the sample with the lowest amplitude.

Possibly Clipped Samples Shows the number of samples have likely exceeded 0 dBFS. Click the icon to the right of

this value to navigate to the first clipped sample in the audio file. (If necessary, click the icon again to view subsequent clipped samples.)

Total, Maximum, Minimum, and Average RMS Amplitude Show the root-mean-square values of the selection. RMS

values are based on the prevalence of specific amplitudes, often reflecting perceived loudness better than absolute or average amplitudes.

DC Offset Shows any direct current offset applied to the waveform during recording. Positive values are above the

center line, and negative values are below it. (See

Measured Bit Depth Reports the waveform’s bit depth. (32 indicates that the waveform uses the full 32-bit float range).

Dynamic Range Reflects the difference between the Maximum and Minimum RMS Amplitude.

Dynamic Range Used Shows the dynamic range minus unusually long periods of low RMS amplitude, such as silent

“Correct DC offset” on page 30.)

passages.

Loudness Shows the average amplitude.

Perceived Loudness Compensates for the human ear’s emphasis on middle frequencies.

Copy Copies all statistics on the General tab to the system clipboard.

RMS Settings options

To adjust how RMS statistics are calculated, set the following options:

0dB = FS Sine Wave Correspond the dB level to a full-scale sine wave, where peak amplitude is about 3.01 dB quieter

than a full-scale square wave.

0dB = FS Square Wave Corresponds the dB level to a full-scale square wave, where peak amplitude is about 3.01 dB

louder than a full-scale sine wave.

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

Account For DC Ignores any DC offset in the measurements.

Window Width Specifies the number of milliseconds in each RMS window. A selected range contains a series of such

windows, which Adobe Audition averages to calculate the Minimum RMS and Maximum RMS values. To achieve the most accurate RMS values, use wide windows for audio with a wide dynamic range, and narrow windows for audio with a narrow dynamic range.

Converting sample types

Hear a file in a different sample rate

The Interpret Sample Rate command lets you hear how an audio file sounds at a different sample rate. (See “Understanding sample rate” on page 5.) This command helps you identify files that specify an incorrect rate in the file header. To then permanently convert the sample rate, choose Edit > Convert Sample Type.

1 In the Waveform Editor, choose Edit > Interpret Sample Rate.

2 Enter a sample rate in the text box, or choose a common sample rate from the list.

Note: Although you can work with sample rates ranging from 6000 to 192,000 Hz in Adobe Audition, your sound card may not be capable of playing all rates properly. To determine supported sample rates, consult the documentation for the card.

Convert the sample rate of a file

The sample rate of a file determines the frequency range of the waveform. When converting the sample rate, keep in mind that most sound cards support only certain sample rates.

1 In the Waveform Editor, choose Edit > Convert Sample Type.

To quickly access the Convert Sample Type dialog box, double-click the Sample Type section of the status bar. (See “Display the status bar” on page 16.)

2 Select a rate from the Sample Rate list, or enter a custom rate in the text box.

3 In the Advanced section, drag the Quality slider to adjust the quality of the sampling conversion.

Higher values retain more high frequencies, but the conversion takes longer. Lower values require less processing time but reduce high frequencies.

Use higher Quality values whenever you downsample a high rate to a low rate. When upsampling, higher values have little effect.

4 For the best results, select Pre/Post Filter to prevent aliasing noise.

Convert a waveform between surround, stereo, and mono

The Convert Sample Type command is the quickest way to convert a waveform to a different number of channels.

1 In the Waveform Editor, choose Edit > Convert Sample Type.

2 From the Channels menu, select Mono, Stereo, or 5.1.

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

3 In the Advanced section, Enter percentages for Left Mix and Right Mix:

• When you convert from mono to stereo, the Left Mix and Right Mix options specify the relative amplitude with

which the original mono signal is placed into each side of the new stereo signal. For example, you can place the mono source on the left channel only, the right channel only, or any point in between.

• When you convert from stereo to mono, the Left Mix and Right Mix options control the amount of signal from the

respective channel that will be mixed into the final mono waveform. The most common mixing method uses 50% of both channels.

For other channel-conversion techniques, see the following topics:

• “Extract audio channels to mono files” on page 132

• “Mix audio data when pasting” on page 43

• “Channel Mixer effect” on page 65

Change the bit depth of a file

The bit depth of a file determines the dynamic range of the audio. (See “Understanding bit depth” on page 6.) Adobe Audition supports up to 32-bit resolution. You can raise the bit depth of a file to gain a greater dynamic range, or you can lower the bit depth to reduce the file size.

Note: Some common applications and media players require 16-bit or lower audio.

1 In the Waveform Editor, choose Edit > Convert Sample Type.

2 Select a Bit Depth from the menu, or enter a custom bit depth in the text box.

3 In the Advanced section, set the following options:

Dithering Enables or disables dithering when converting to lower bit depths. If dithering is disabled, bit depth is

abruptly truncated, producing a crackly effect on low-volume passages caused by quantization distortion.

Although dithering introduces a small amount of noise, the result is far preferable to the increased distortion that you would otherwise hear at low signal levels. Dithering also lets you hear sounds that would be masked by the noise and distortion limits of audio at lower bit depths.

Dither Type Controls how dithering noise is distributed relative to the original amplitude value. Usually, Triangular

provides the best tradeoff among signal-to-noise ratio, distortion, and noise modulation.

Dither type Reduces signal-to-noise ratio Noise modulation

Triangular 4.8 dB No

Gaussian 6.0 dB Negligible

Note: Triangular (Shaped) and Gaussian (Shaped) move slightly more noise to higher frequencies. For additional control, set Noise Shaping options.

Noise Shaping Determines which frequencies contain dithering noise. By introducing noise shaping, you may be able

to use lower dither depths without introducing audible artifacts. The best shaping depends on the source audio, final sample rate, and bit depth.

Note: Noise Shaping is disabled for sample rates below 32Khz because all noise would remain in audible frequencies.

• High Pass With a crossover set to 7.3 kHz, drops dithering noise to -180dB at 0 Hz and -162dB at100 Hz.

• Light Slope With a crossover set to 11 kHz, drops noise to -3dB at 0 Hz and -10dB at 5 kHz.

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

• Neutral Light is flat up to 14 kHz, ramps noise up to a maximum at 17kHz, and is again flat at higher frequencies.

Background noise sounds the same as it does without noise shaping but is about 11dB quieter.

Heavy is flat up to 15.5kHz, placing all dithering noise above 16kHz (or wherever you specify the crossover). Sensitive ears may hear a high pitched ringing if the crossover is too low. If converting 48 or 96 kHz audio, however, the crossover can be placed well above 20 kHz.

Choose a Neutral shape to avoid sonically coloring background hiss. Note, however, that hiss will sound louder than with other shapes.

• U-Shaped Shallow is mostly flat from 2 kHz up to 14 kHz but gets louder as audio approaches 0 Hz because low

frequencies are much less audible. Medium places a little more noise in the highs above 9 kHz, allowing for lower noise below that frequency. Deep increases noise above 9 kHz even more, but also lowers it much more in the 2-6 kHz range.

• Weighted Light attempts to match how the ear perceives low-level sounds by reducing noise more in the 2-6 kHz

range and raising it in the 10-14 kHz range. At high volumes, hiss may be more noticeable. Heavy more evenly reduces the most sensitive 2-6KHz range at the expense of more noise above 8kHz.

Crossover Specifies the frequency above which noise shaping will occur.

Strength Specify the maximum amplitude of noise added to any one frequency.

Adaptive Mode Varies the distribution of noise across frequencies.

Use sample rate conversion presets

If you need to make the same conversion on multiple files, you can save time by using a sample rate conversion preset.

1 Choose Edit > Convert Sample Type.

2 Adjust the settings as desired.

3 Click the New Preset button .

After you create a preset, it appears in the Presets list at the top of the dialog box. If you want to delete a preset, choose it from the list, and click the Delete button

Undo, redo, and history

Undo or redo changes

Each time you start Adobe Audition, it keeps track of the edits you perform. Edits aren’t permanently applied until you save and close a file, giving you unlimited undo and redo capability.

❖ To undo or redo changes, do any of the following:

• To undo a change, choose Edit > Undo [name of change].

• To redo a change, choose Edit > Redo [name of change].

• To repeat the last command in the Waveform Editor, choose Edit > Repeat Last Command. You can repeat most

commands; however, there are a few exceptions (such as Delete).

To repeat the last command without opening its dialog box, press Ctrl+R (Windows) or Command+R (Mac OS).

Last updated 11/7/2011

USING ADOBE AUDITION

Editing audio files

Compare history states

While the Undo and Redo commands restrict you to an incremental sequence of changes, the History panel lets you instantly revert back to any previous change. Use the panel to quickly compare processed and original audio or discard a series of changes that produced undesired results.

Note: History states disappear when you close a file.

Revert to states

• To revert to any history state, click it.

• To incrementally move through states, press the up and down arrows on the keyboard.

Delete states

When you work with very large audio files, delete unnecessary history states to clear disk space and improve performance.

• To delete all states, choose Clear History from the panel menu .

• To delete a specific state, select it, and then click the trash icon .

Note: Deleting history states also removes related Undo commands.

Last updated 11/7/2011

Chapter 6: Applying effects

Shared effects controls

Effects Rack overview

The Effects Rack lets you insert, edit, and reorder up to 16 effects, optimize mix levels, and store favorite presets. Most rack controls appear in both the Waveform and Multitrack editors.

Controls shared by the Waveform and Multitrack editors A. Rack Preset controls B. Effect slots C. Level controls D. Main Power button

More Help topics

“Apply groups of effects in the Waveform Editor” on page 60

“Apply effects to clips or tracks” on page 61

“Comparing the Waveform and Multitrack editors” on page 8

Controls unique to the Waveform Editor

In the Waveform Editor, the Effects Rack provides a Process menu that that lets you modify a selection or the entire file, and an Apply button that permanently applies effects.

Controls unique to the Waveform Editor A. Apply button permanently applies effects B. Process menu lets you modify selection or entire file

Controls unique to the Multitrack Editor

The Effects Rack provides Pre-render Track and FX Pre/Post-Fader buttons that you use to optimize and route effects. Each clip and track has its own Effects Rack, which is saved with the session.

Note: Buses and the Master track lack a Pre-render option because processing effects from all source tracks would reduce performance.

Last updated 11/7/2011

USING ADOBE AUDITION

Applying effects

A B

Controls unique to the Multitrack Editor A. FX Pre/Post-Fader B. Pre-render

Set input, output, and mix levels in racks

• To optimize volume, adjust Input and Output levels so their meters peak without clipping.

• To change the percentage of processed audio, drag the Mix slider. 100% (Wet) equals fully processed audio; 0%

(Dry) equals original, unprocessed audio.

Insert, bypass, reorder, or remove effects in racks

In the Effects Rack, you manage groups of effects by using individual effect slots.

In the Multitrack Editor, the fx section of the Editor panel or Mixer provides quick access to slots in the Effects Rack.

Reordering and inserting effects in racks: A. Reorder by dragging B. Insert with the slot menu

• To insert an effect, choose it from a slot’s pop-up menu. Then adjust effect settings as desired.

To later reaccess effect settings, double-click the effect name in the rack.

• To bypass an effect, click its Power button .

• To bypass all effects, click the main Power button in the lower left corner of a rack, or the fx power button in the

Editor panel or Mixer.

• To bypass a selected group of effects, choose Toggle Power State of Selected Effects from the panel menu .

Bypass effects to quickly compare processed and unprocessed audio.

• To remove a single effect, choose Remove Effect from a slot’s pop-up menu. Or select the slot, and press Delete.

• To remove all effects, choose Remove All Effects from the panel menu .

• To reorder effects, drag them to different slots.

Reordering effects produces different sonic results. (For an example, place Reverb prior to Phaser, and vice versa.)

Last updated 11/7/2011

USING ADOBE AUDITION

Applying effects

Use effect presets

Many effects provide presets that let you store and recall favorite settings. In addition to effect-specific presets, the Effects Rack provides rack presets that store groups of effects and settings.

• To apply a preset, choose it from the Presets menu.

• To save current settings as a preset, click the New Preset button .

• To delete a preset, select it, and click the Delete button .

To modify an existing preset, apply it, adjust settings as desired, and then save a new preset with the same name.

More Help topics

“Favorites” on page 139

Control effect settings with graphs

Many Adobe Audition effects provide graphs where you can adjust parameters. By adding and moving control points on the graph, you can precisely tailor effect settings.

Graph control points function together with related numerical settings. If you change or disable a numerical setting, the related graph control follows suit.

Moving a control point changes the related settings, and vice versa.

• To move a point on a graph, drag it to a new location.

Note: The following techniques don’t apply to the DeHummer, Mastering, Full Reverb, Parametric Equalizer, and Track EQ graphs.

• To add a control point to a graph, click in the grid at the location where you want to place the point.

• To enter numeric values for a control point, right-click it, and choose Edit Point.

• To remove a point from a graph, drag it off the graph.

• To return a graph to its default state, click the Reset button .

About spline curves for graphs

By default, graphs display straight lines between control points. However, some graphs provide a Spline Curves option that creates a curve between control points for smoother transitions.

Last updated 11/7/2011

USING ADOBE AUDITION

Applying effects

When you use spline curves, lines don’t travel directly through control points. Instead, the points control the shape of the curve. To move the curve closer to a control point, click near it to create a cluster of control points.

Graph with straight lines compared to graph with spline curves

Applying effects in the Waveform Editor

While previewing effects, you can adjust selections and the current-time indicator in the Editor panel. (The Normalize and Stretch effects are exceptions.)

Apply groups of effects in the Waveform Editor

In the Waveform Editor, the Effects Rack lets you apply groups of effects. (It doesn’t provide process effects such as Noise Reduction, which must be applied individually.)

1 Choose Window > Effects Rack.

2 In the numbered list, choose effects for up to 16 slots. (See “Insert, bypass, reorder, or remove effects in racks” on

page 58.)

3 Start playback to preview the changes, and then edit, mix, and reorder effects as needed.

To compare processed audio to original audio, select and deselect the main Power button in the lower left corner of the rack, or the Power buttons for individual effects.

4 To apply the changes to the audio data, click Apply.

To store settings, save a rack preset. (See “Use effect presets” on page 59.)

More Help topics

“Effects Rack overview” on page 57

Apply individual effects in the Waveform Editor

1 From any submenu in the Effects menu, choose an effect.

2 Click the Preview button , and then edit settings as needed.

Last updated 11/7/2011

USING ADOBE AUDITION

Applying effects

As you edit settings, watch the Levels panel to optimize amplitude.

3 To compare original audio to processed audio, select and deselect the Power button .

4 To apply the changes to the audio data, click Apply.

More Help topics

“Use effect presets” on page 59

“Control effect settings with graphs” on page 59

About process effects

You can identify process effects by the word process in menu commands. These processing-intensive effects are available only offline in the Waveform Editor. Unlike real-time effects, process effects can be applied only individually, so they aren’t accessible in the Effects Rack.

Applying effects in the Multitrack Editor

Apply effects to clips or tracks

In the Multitrack Editor, you can apply up to 16 effects to each clip, track, and bus and adjust them while a mix plays. (Apply clip effects if a track contains multiple clips that you want to process independently.)

You can insert, reorder, and remove effects in the Editor, Mixer, or Effects Rack panel. Only in the Effects Rack, however, can you save favorite settings as presets, which you can apply to multiple tracks.

In the Multitrack Editor, effects are nondestructive, so you can change them at any time. To readapt a session for different projects, for example, simply reopen it and change effects to create new sonic textures.

Last updated 11/7/2011

USING ADOBE AUDITION

Applying effects

Revealing effect slots in the Editor panel

1 Do any of the following:

• Select a clip, and click Clip Effects at the top of the Effects Rack.

• Select a track, and click Track Effects at the top of the Effects Rack.

• Display the fx section of the Editor or Mixer. (In the Editor panel, click the button in the upper-left corner.)

2 Choose effects for up to 16 slots in the list. (See “Insert, bypass, reorder, or remove effects in racks” on page 58.)

3 Press the spacebar to play the session, and then edit, reorder, or remove effects as needed.

To change effect settings over time, use envelopes. (See “Automating mixes with envelopes” on page 121.)

More Help topics

“Effects Rack overview” on page 57

Pre-render track effects to improve performance

In the Multitrack Editor, pre-render track effects to address heavy CPU usage, improving performance for complex mixes or low-latency recording. (Latency measures the delay between user input and sound output from a computer. If latency is high, it produces an audible echo during recording, disrupting timing for musicians.)

You can continue to edit track settings normally; pre-rendering processes audio when pauses occur in playback or editing.

❖ In the Editor panel, Effects Rack, or Mixer, click the Pre-Render Track button .

Insert effects before or after sends and EQ

On each track, you can insert effects either pre- or post-fader. Pre-fader effects process audio before sends and EQ. Post-fader effects process audio after sends and EQ. For most mixes, the default, pre-fader setting works well. The post-fader setting offers signal-routing flexibility for particularly complex mixes.

❖ In the fx section of the Editor panel or Mixer, click the Pre-Fader/Post-Fader button to insert effects either before

sends and EQ

, or after .

Last updated 11/7/2011

USING ADOBE AUDITION

Applying effects

If you’re editing effect settings in the Effects Rack, click the Pre-Fader/Post-Fader button in the lower-left corner.

ABCD

Pre- and post-fader effect and send routing for each track: A. Input B. EQ C. Volume D. Mute E. Send F. Effects Rack

More Help topics

“Routing audio to buses, sends, and the Master track” on page 110

“Set up a send” on page 112

Adding third-party plug-ins

Third-party plug-ins let you extend the already powerful effects provided with Adobe Audition. The application supports VST plug-ins on both platforms and Audio Units plug-ins on Mac OS.

Applying plug-in effects is identical to applying built-in effects. For information about plug-in features, consult the documentation provided by the plug-in manufacturer.

Enable VST and Audio Units plug-ins

To access third-party plug-ins in Adobe Audition, you must first enable them. By default, all third-party plug-ins are disabled. To optimize performance, enable only the plug-ins you plan to use in Adobe Audition.

Note: If effects are being used in a multitrack session, close the session.

1 Choose Effects > Audio Plug-in Manager.

2 In the VST Plug-in Folders section, click Add to specify custom folders you want to scan for plug-ins. Click Default

to specify the standard VST folder for your operating system.

3 In the Available Plug-ins section, click Scan For Plug-ins.

If you’ve recently updated a plug-in, select Rescan Existing Plug-ins.

4 Select the plug-ins you want to access in Adobe Audition, and then click OK.

Last updated 11/7/2011

USING ADOBE AUDITION

Applying effects

Note: If a third-party effect is incompatible, Adobe Audition adds it to an Unsupported submenu in effects menus.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Automating track settings” on page 123

Last updated 11/7/2011

Chapter 7: Effects reference

Amplitude and compression effects

Amplify effect

The Amplitude And Compression > Amplify effect boosts or attenuates an audio signal. Because 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.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

Channel Mixer effect

The Amplitude and Compression > Channel Mixer effect alters the balance of stereo or surround channels, letting you change the apparent position of sounds, correct mismatched levels, or address phasing issues.

Channel tabs Select the output channel.

Input channel sliders Determine the percentage of the current channels to mix into the output channel. For a stereo

file, for example, an L value of 50 and an R value of 50 results in an output channel that contains equal audio from the current left and right channels.

Invert Inverts a channel’s phase. (To understand this key audio concept, see “How sound waves interact” on page 4.)

Inverting all channels causes no perceived difference in sound. Inverting only one channel, however, can greatly change the sound.

More Help topics

“Mix audio data when pasting” on page 43

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

DeEsser effect

The Amplitude and Compression > DeEsser effect removes sibilance, “ess” sounds heard in speech and singing that can distort high frequencies.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

The graph reveals the processed frequencies. Click the Preview button to see how much audio content exists in the processed range.

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.

To visually adjust Center Frequency and Bandwidth, drag the edges of the selection in the graph.

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.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

Dynamics Processing effect

The Amplitude And Compression > 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.)

The Dynamics Processing effect can produce subtle changes that you notice only after repeated listening. When applying this effect in the Waveform Editor, use a copy of the original file so you can return to the original audio if necessary.

Use the Broadcast Limiter preset to simulate the processed sound of a contemporary radio station.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Control effect settings with graphs” on page 59

“Use effect presets” on page 59

Dynamics tab

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.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

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 will dramatically boost 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.

To numerically adjust an existing control point, right-click it, and choose Edit Point.

Delete point Removes selected point from the graph.

Invert Flips the graph, converting compression into expansion, or vice versa.

Note: You can invert a graph only if it has points in the two default corners (-100, -100 and 0, 0) and if its output level increases from left to right (that is, each control point must be higher than the one to its left).

Reset Resets the graph to its default state.

Spline Curves creates smoother, curved transitions between control points, rather than more abrupt, linear

transitions. (See

Make-Up Gain Boosts the processed signal.

“About spline curves for graphs” on page 59.)

Settings tab

General Provides overall settings.

• Look-Ahead Time Addresses transient spikes that can occur at the onset of extremely 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 may be 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 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.

Use fast attack and release settings for audio with fast transients, and slower settings for less percussive audio.

• 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.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

• 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.

Note: If the sum of Attack and Release times is too short (less than about 30 milliseconds), audible artifacts can be heard. To see good attack and release times for different types of audio content, choose various options from the Presets menu.

• Link Channels Processes all channels equally, preserving the stereo or surround balance. For example, a

compressed drum beat on the left channel will reduce 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

The Amplitude And Compression > 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.

To avoid clipping when working with 16-bit audio, set this value to no more than -0.3 dB. If you set it even lower, to

-3 dB, you’ll have a little more clearance for any future edits.

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) generally needed to attenuate the audio before the loudest

peak is hit.

Note: Make sure that the value is at least 5 milliseconds. If this value is too small, audible distortion effects may occur.

Release Time Sets the time (in milliseconds) needed for the attenuation to rebound back 12 dB (or roughly the time

needed for audio to resume normal volume if an extremely loud peak is encountered). In general, a setting of around 100 (the default) works well and preserves very low bass frequencies.

Note: If this value is too large, audio may remain very quiet and not resume normal levels for a while.

Link Channels Links the loudness of all channels together, preserving the stereo or surround balance.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

Multiband Compressor effect

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

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

Controls in the Multiband Compressor let you precisely define crossover frequencies and apply band-specific compression settings. Click Solo buttons to preview bands in isolation, or Bypass buttons to pass bands through without processing. After you fine-tune individual bands, select Link Band Controls to adjust them globally, and then optimize overall volume with the Output Gain slider and Limiter settings.

To change compression settings over time, use automation lanes in the Multitrack Editor. (See “Automating track

settings” on page 123.)

A B C D

Adjusting a crossover frequency in the Multiband Compressor A. Frequency bands B. Crossover markers C. Bypassed band (no processing applied) D. Amplitude scale E. Frequency scale

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.

A B

Band-specific controls in the Multiband Compressor A. Solo B. Bypass C. Threshold slider D. Input Level meters E. Gain Reduction meters

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.

Alt-click (Windows) or Option-click (Mac OS) Solo or Bypass buttons to quickly apply a unique setting to one band.

Threshold sliders Set the input level at which compression begins. Possible values range from -60 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.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

Input Level meters Measure input amplitude. Double-click the meters to reset peak and clip indicators.

Gain Reduction meters Measure amplitude reduction with red meters that extend from top (minimal reduction) to

bottom (maximum reduction).

Gain Boosts or cuts amplitude after compression. Possible values range from -18 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 extremely compressed sound often heard in pop music.

Attack Determines how quickly compression is applied when audio exceeds the threshold. Possible values range from

0 to 500 milliseconds. The default, 10 milliseconds, works well for a wide range of audio. Faster settings may 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 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 to +18 dB, where 0

is unity gain. Double-click the meters to reset peak and clip indicators.

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 agressive than similar band-specific settings. Then specify a Margin setting to determine the absolute ceiling relative to 0 dBFS.

To create extremely compressed audio, enable the Limiter, and then experiment with very high Output Gain settings.

Spectrum On Input Displays 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 may 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.

To temporarily link band controls, hold down Alt+Shift (Windows) or Option+Shift (Mac OS). To reset a control in all bands, hold down Ctrl+Alt+Shift (Windows) or Command+Option+Shift (Mac OS), and click the control.

More Help topics

“Mastering effect” on page 98

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

Normalize effect (Waveform Editor only)

Note: This effect requires offline processing. While it is open, you cannot edit the waveform, adjust selections, or move the current-time indicator.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

The Amplitude And Compression > Normalize effect lets you set a peak level for a file or selection. When you normalize audio to 100%, you achieve the maximum amplitude that digital audio allows—0 dBFS. If you’re sending audio to a mastering engineer, however, normalize audio to between –3 and –6 dBFS, providing a cushion for further processing.

The Normalize effect amplifies the entire file or selection equally. For example, if the original audio reaches a loud peak of 80% and a quiet low of 20%, normalizing to 100% amplifies the loud peak to 100% and the quiet low to 40%.

To apply RMS normalization, choose Effects > Match Volume. If desired, you can apply that command to only one file. (See “Match volume across multiple files” on page 139.)

Normalize To Sets the percentage of the highest peak relative to the maximum possible amplitude.

Select dB to enter the Normalize value in decibels instead of a percentage.

Normalize All Channels Equally Uses all channels of a stereo or surround waveform to calculate the amplification

amount. If this option is deselected, the amount is calculated separately for each channel, potentially amplifying one considerably more than others.

DC Bias Adjust Lets you adjust the position of the waveform in the wave display. Some recording hardware may

introduce a DC bias, causing the recorded waveform to appear to be above or below the normal center line in the wave display. To center the waveform, set the percentage to zero. To skew the entire selected waveform above or below the center line, specify a positive or negative percentage.

More Help topics

“About process effects” on page 61

“Apply individual effects in the Waveform Editor” on page 60

Single-band Compressor effect

The Amplitude And Compression > Single-band Compressor effect reduces dynamic range, producing consistent volume levels and increasing perceived loudness. Single-band compression is particularly effective for voice-overs, because it helps the speaker stand out over musical soundtracks and background audio.

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 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 3 outputs 1 dB for every 3-dB

increase above the threshold. Typical settings range from 2 to 5; higher settings produce the extremely 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.

dB below the peak input level.

Output Gain

Boosts or cuts amplitude after compression. Possible values range from -30 dB to +30 dB, where 0 is unity gain.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

More Help topics

“Applying effects” on page 57

Speech Volume Leveler effect

The Amplitude and Compression > Speech Volume Leveler is a compression effect that optimizes dialogue, evening out levels and removing background noise.

For the best results, do the following:

1 Select audio with the lowest level. Set Target Volume Level and Leveling Amount to the left. Start playback, and

gradually increase the Leveling Amount until speech becomes nicely audible without increasing background noise.

2 Select audio with the highest level, and start playback. Adjust the Target Volume Level until the volume matches

the loudness of the quiet passage you adjusted previously.

3 If necessary, readjust the Leveling Amount to avoid an over-compressed sound.

Here are additional details about each option:

Target Volume Level Sets the desired output level relative to zero dBFS. (See “Measuring amplitude in dBFS” on

page 6.)

Leveling Amount At low settings, amplifies speech slightly without boosting the noise floor. At high settings, amplifies

the entire signal more as the signal drops closer to the noise floor.

Boost Low Signals Interprets shorter, low-volume passages as speech that should be amplified. For most audio

content, deselect this option to produce smoother sound.

Advanced settings Click the triangle to access the following options:

• Compressor Maintains a strong level if the processed signal falls below a threshold relative to zero dBFS.

• Noise Gate Eliminates background noise by dramatically reducing output level when the signal drops by an offset

you specify.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

Tube-modeled Compressor effect

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

Threshold slider Sets the input level at which compression begins. Possible values range from -60 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 settings around 15 dB below the peak input level.

Input Level meters To the left of the slider, these meters measure input amplitude. Double-click the meters to reset

peak and clip indicators.

dB below the peak input level; to highly compress audio and greatly reduce dynamic range, try

Gain Reduction meters To the right of the slider, these meters measure amplitude reduction with red bars that extend

from top (minimal reduction) to bottom (maximum reduction).

Gain Boosts or cuts amplitude after compression. Possible values range from -18 to +18 dB, where 0 is unity gain.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

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 extremely compressed sound often heard in pop music.

Attack Determines how quickly compression is applied when audio exceeds the threshold. Possible values range from

Release Determines how quickly compression stops after audio drops below the threshold. Possible values range from

0 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.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

Volume Envelope effect (Waveform Editor only)

The Amplitude And Compression > Volume Envelope effect lets you change volume over time with boosts and fades. In the Editor panel, simply drag the yellow line. The top of the panel represents 100% (normal) amplification; the bottom represents 100% attenuation (silence).

Though the Volume Envelope effect isn’t available in the Multitrack Editor, you can use automation lanes to accomplish the same task. (See “Automating track settings” on page 123.)

Dragging an anchor point in the Editor panel

Yellow envelope line in Editor panel Drag to adjust amplitude percentage, and click to add keyframes for additional

boosts and fades. To quickly select, reposition, or delete multiple keyframes, see

“Adjust automation with keyframes”

on page 124.

Spline curves Applies smoother, curved transitions between keyframes, rather than linear transitions. See “About

spline curves for graphs” on page 59.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

More Help topics

“About process effects” on page 61

“Apply individual effects in the Waveform Editor” on page 60

“Control effect settings with graphs” on page 59

“Use effect presets” on page 59

Delay and echo effects

Delays are separate copies of an original signal that reoccur within milliseconds of each other. Echoes are sounds that are delayed far enough in time so that you hear each as a distinct copy of the original sound. When reverb or chorus might muddy the mix, both delays and echoes are a great way to add ambience to a track.

To access familiar options from hardware delays, use the Echo effect in Adobe Audition.

More Help topics

“Reverb effects” on page 92

Analog Delay effect

The 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.

Note: When experimenting with extremely high Feedback settings, turn down your system volume.

Trash Increases distortion and boosts low frequencies, adding warmth.

Spread Determines the stereo width of the delayed signal.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

Delay effect

The Delay And Echo > Delay effect can be used to create single echoes, as well as a number of other effects. Delays of milliseconds or more create discrete echoes, while those between 15-34 milliseconds can create a simple chorus or

35 flanging effect. (These results won’t be as effective as the Chorus or Flanger effects in Adobe Audition, because the delay settings don’t change over time.)

By further reducing a delay to between 1 and 14 milliseconds, you can spatially locate a mono sound so that the sound seems to be coming from the left or the right side, even though the actual volume levels for left and right are identical.

Delay Time Adjusts the delay for both the left and right channels from -500 milliseconds to +500 milliseconds.

Entering a negative number means that you can move a channel ahead in time instead of delaying it. For instance, if you enter 200 milliseconds for the left channel, the delayed portion of the affected waveform is heard before the original part.

Mix Sets the ratio of processed, Wet signal to original, Dry signal to be mixed into the final output. A value of 50 mixes

the two evenly.

Invert Inverts the phase of the delayed signal, creating phase-cancellation effects similar to comb filters. (To

understand phase cancellation, see

More Help topics

“Applying effects in the Waveform Editor” on page 60

“How sound waves interact” on page 4.)

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

Echo effect

The Delay And Echo > Echo effect adds a series of repeated, decaying echoes to a sound. (For a single echo, use the Delay effect instead.) You can create effects ranging from a Grand Canyon-type “Hello-ello-llo-lo-o” to metallic, clanging drainpipe sounds by varying the delay amount. By equalizing the delays, you can change a room’s characteristic sound from one with reflective surfaces (creating echoes that sound brighter) to one that is almost totally absorptive (creating echoes that sound darker).

Note: Make sure the audio file is long enough for the echo to end. If the echo is cut off abruptly before it fully decays, undo the Echo effect, add several seconds of silence by choosing Generate

Delay Time Specifies the number of milliseconds, beats, or samples between each echo. For example, a setting of 100

milliseconds results in a 1/10th-second delay between successive echoes.

Feedback Determines the falloff ratio of an echo. Each successive echo tails off at a certain percentage less than the

previous one. A decay setting of 0% results in no echo at all, while a decay of 100% produces an echo that never gets quieter.

Echo Level Sets the percentage of echoed (wet) signal to mix with the original (dry) signal in the final output.

You can create striking stereo echo effects by setting different left and right values for the Delay Time, Feedback, and Echo Level controls.

Lock Left & Right Links the sliders for Decay, Delay, and Initial Echo Volume, maintaining the same settings for each

channel.

> Silence, and then reapply the effect.

Echo Bounce Makes the echoes bounce back and forth between the left and right channels. If you want to create one

echo that bounces back and forth, select an initial echo volume of 100% for one channel and 0% for the other. Otherwise, the settings for each channel will bounce to the other, creating two sets of echoes on each channel.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

Successive Echo Equalization Passes each successive echo through an eight-band equalizer, letting you simulate the

natural sound absorption of a room. A setting of 0 leaves the frequency band unchanged, while a maximum setting of

-15 decreases that frequency by 15 dB. And, because -15 dB is the difference of each successive echo, some frequencies will die out much faster than others.

Delay Time Units Specifies milliseconds, beats, or samples for the Delay Time setting.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

Diagnostics effects (Waveform Editor only)

Diagnostics are available either via the Effects menu or directly from the Diagnostics panel (Window > Diagnostics). These tools let you quickly remove clicks, distortion, or silence from audio, as well as add markers where silence occurs.

For maximum audio restoration control, use diagnostics together with Spectral Display tools and Noise Reduction effects. See “Techniques for restoring audio” on page 84.

More Help topics

“About process effects” on page 61

“Apply individual effects in the Waveform Editor” on page 60

“Use effect presets” on page 59

Diagnose and repair, delete, or mark audio

Unlike conventional noise reduction effects, which process all selected audio, diagnostics scan for problematic or silent areas, and then let you choose which to address.

1 In the Diagnostics panel, choose an option from the Effect menu.

2 Click Scan.

3 At the bottom of the panel, do any of the following:

• Select one or more detected items in the list, and click Repair, Delete, or Mark. (The available options depend

upon the chosen diagnostic effect.)

To mark detected clicks or clipping, right-click selected items in the list, and choose Create Markers from the pop-up menu. (See “Working with markers” on page 45.)

• Click Repair All, Delete All, or Mark All to address all detected items.

• Click the magnifying glass to zoom in on a selected problem in the Editor panel. Click the icon again to zoom out.

• Click Clear Repaired, Deleted, or Marked to remove previously addressed items from the list.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

DeClicker options

The Diagnostics > DeClicker effect detects and removes clicks and pops from wireless microphones, vinyl records, and other sources.

DeClicker options match those for the Automatic Click Remover, which you can combine with other effects in the Effects Rack and apply in the Multitrack Editor. (See applies multiple scan and repair passes automatically; to achieve the same level of click reduction with the DeClicker, you must manually apply it multiple times. However, the DeClicker lets you evaluate detected clicks and choose which to address.

In the Diagnostics panel, click Settings to access these options:

Threshold Determines sensitivity to noise. Lower settings detect more clicks and pops but may include audio you wish

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.

To visually identify clicks, zoom in and use Spectral Frequency Display with a resolution of 256 bands. (You can access this setting in the Spectral Display tab of the Preferences dialog box.) Most clicks appear as bright vertical bars that

extend from the top to the bottom of the display.

“Automatic Click Remover effect” on page 88.) That effect also

DeClipper options

The Diagnostics > DeClipper effect repairs clipped waveforms by filling in clipped sections with new audio data. Clipping occurs when audio amplitude exceeds the maximum level for the current bit depth. Commonly, clipping results from recording levels that are too high. You can monitor clipping during recording or playback by watching the Level Meters; when clipping occurs, the boxes on the far right of the meters turn red.

Visually, clipped audio appears as broad flat areas at the top of a waveform. Sonically, clipped audio is a static-like distortion.

Note: If you need to adjust the DC offset of clipped audio, first use the DeClipper effect. If you instead adjust DC offset first, the DeClipper won’t identify clipped areas that fall below 0 dBFS.

In the Diagnostics panel, click Settings to access these options:

Gain Specifies the amount of attenuation that occurs before processing. Click Auto to base the gain setting on average

input amplitude.

Tolerance Specifies the amplitude variation in clipped regions. A value of 0% detects clipping only in perfectly

horizontal lines at maximum amplitude; 1% detects clipping beginning at 1% below maximum amplitude, and so on. (A value of 1% detects most clipping.)

Min. Clip Size Specifies the length of the shortest run of clipped samples to repair. Lower values repair a higher

percentage of clipped samples; higher values repair clipped samples only if they’re preceded or followed other clipped samples.

Interpolation The Cubic option uses spline curves to re-create the frequency content of clipped audio. This approach

is faster for most situations but can introduce spurious new frequencies. The FFT option uses Fast Fourier transforms to re-create clipped audio. This approach is typically slower but best for severe clipping. From the FFT Size menu, choose the number of frequency bands to evaluate and replace. (More bands result in greater accuracy but longer processing.)

To retain amplitude when restoring clipped audio, apply the DeClipper effect with a Gain setting of zero, followed by the Hard Limiting effect with a Boost value of zero and a Limit value of -0.2 dB.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

Delete Silence and Mark Audio options

The Diagnostics > Delete Silence and Mark Audio effects identify silent passages of audio and either remove or mark them. (See affecting foreground audio. Automatically marking silence helps you quickly navigate to audio cues for editing.

In the Diagnostics panel, click Settings to access these options:

Define Silence As Specifies the amplitude and duration identified as silence.

Define Audio As Specifies the amplitude and duration identified as audio content.

Find Levels Automatically calculates the signal levels of silence and audio based on content in the file.

Fix By (Delete Silence only) Choose Shortening Silence to reduce silent passages to the specified number of

milliseconds. Choose Deleting Silence to mute silent passages but retain file length. (Deleting silence helps maintain video synchronization with audio clips in video editing applications.)

More Help topics

“Create silence” on page 49

“Working with markers” on page 45.) Automatically deleting silence helps you tighten up tracks without

To divide sections of sound or speech separated by silence into different files, apply the Mark Audio effect, and click Mark All. Then see “Save audio between markers to new files” on page 47.

Filter and equalizer effects

FFT Filter effect

The graphic nature of the Filter And EQ > 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), or 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. (See

Reset Reverts the graph to the default state, removing filtering.

Advanced options 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 between 1024 and 8192 work well for most material.

“About spline curves for graphs” on page 59.)

• Window Determines the Fast Fourier transform shape, with each option resulting in a different frequency response

curve.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

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.

More Help topics

“About process effects” on page 61

“Apply individual effects in the Waveform Editor” on page 60

“Control effect settings with graphs” on page 59

“Use effect presets” on page 59

Graphic Equalizer effect

The Filter And EQ > 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 and 120 dB. (By comparison, standard

hardware equalizers have a range of about 12 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.

If you equalize extremely low frequencies, set Accuracy to between 500 and 5000 points.

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.

Note: The Graphic Equalizer is an FIR (Finite Impulse Response) filter. FIR filters better maintain phase accuracy but have slightly less frequency accuracy than IIR (Infinite Impulse Response) filters like the Parametric Equalizer.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

Parametric Equalizer effect

The Filter And EQ > Parametric Equalizer effect provides maximum control over tonal equalization. Unlike the Graphic Equalizer, which provides a fixed number of frequencies and Q bandwidths, the Parametric Equalizer gives you total control over frequency, Q, and gain settings. For example, you can simultaneously reduce a small range of frequencies centered around 1000 notch filter.

The Parametric Equalizer uses second-order IIR (Infinite Impulse Response) filters, which are very fast and provide very accurate frequency resolution. For example, you can precisely boost a range of 40 to 45 Hz. FIR (Finite Impulse Response) filters like the Graphic Equalizer provide slightly improved phase accuracy, however.

Master Gain Compensates for an overall volume level that’s too loud or too soft after you adjust the EQ settings.

Graph Shows frequency along the horizontal ruler (x-axis) and amplitude along the vertical ruler (y-axis). Frequencies

in the graph range from lowest to highest in a logarithmic fashion (evenly spaced by octaves).

Identifying band-pass and shelving filters in the Parametric Equalizer: A. High- and low-pass filters B. High and low shelving filters

Hz, boost a broad low-frequency shelf centered around 80 Hz, and insert a 60 Hz

A B

Frequency Sets the center frequency for bands 1-5, and the corner frequencies for the band-pass and shelving filters.

Use the low shelving filter to reduce low-end rumble, hum, or other unwanted low-frequency sounds. Use the high shelving filter to reduce hiss, amplifier noise, and the like.

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. Very

high Q values (close to 100) affect a very narrow band and are ideal for notch filters removing particular frequencies, like 60 Hz hum.

When a very narrow band is boosted, audio tends to ring or resonate at that frequency. Q values of 1-10 are best for general equalization.

Band Enables up to five intermediate bands, as well as high-pass, low-pass, and shelving filters, giving you very fine

control over the equalization curve. Click the band button to activate the corresponding settings above.

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.

To visually adjust enabled bands in the graph, drag the related control points.

Constant Q, Constant Width 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.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

Modulation effects

Chorus effect

The Modulation > Chorus effect 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. You can use Chorus to enhance a vocal track or add stereo spaciousness to mono audio.

Adobe Audition uses a direct-simulation method to achieve a chorus effect, making each voice sound distinct from the original by slightly varying timing, intonation, and vibrato. The Feedback setting lets you add extra detail to the result.

To achieve the best results with mono files, convert them to stereo before applying the Chorus effect.

Characteristics Represent the characteristics of each voice in the chorus.

• Voices Determines the number of simulated voices.

Note: As you add more voices, the sound becomes richer and richer —but processing time also increases.

• Delay Time Specifies the maximum amount of delay allowed. Chorusing introduces short delays (often in the 15-35

millisecond range) that vary in duration over time. If the setting is very small, all the voices start merging into the original, and an unnatural flanging effect might occur. If the setting is too high, a warbled effect might occur, like a tape being eaten by a cassette deck.

• Delay Rate Determines how quickly the delay cycles from zero to the maximum delay setting. Because the delay

varies over time, the pitch of the sample increases or decreases over time, giving the effect of separate, slightly out of tune voices. For example, a rate of 2 Hz would vary the delay from zero to the maximum and back twice per second (simulating a pitch vibrato at twice per second). If this setting is too low, the individual voices don’t vary much in pitch. If it is set too high, the voices may vary so quickly that a warbled effect might occur.

• Feedback Adds a percentage of processed voices back into the effect input. Feedback can give a waveform an extra

echo or reverb effect. A little feedback (less than 10%) can provide extra richness, depending on the delay and vibrato settings. Higher settings produce more traditional feedback, a loud ringing which can get loud enough to clip the signal.

• Spread Gives an added delay to each voice, separating them in time by as much as 200 milliseconds (1/5th of a

second). High values cause the separate voices to start at different times—the higher the value, the farther apart the onset of each voice may be. In contrast, low values cause all voices to be in unison. Depending on other settings, low values can also produce flanging effects, which may be undesirable if your goal is a realistic chorus effect.

• Modulation Depth Determines the maximum variation in amplitude that occurs. For example, you can alter the

amplitude of a chorused voice so that it is 5 dB louder or quieter than the original. At extremely high settings, the sound may cut in and out, creating an objectionable warble. At extremely low settings (less than 1 dB), the depth may be unnoticeable unless the Modulation Rate is set extremely high. Natural vibratos occur around 2 dB to 5 dB.

Note that this setting is a maximum only; the vibrato volume might not always go as low as the setting indicates. This limitation is intentional, as it creates a more natural sound.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

• Modulation Rate Determines the maximum rate at which amplitude changes occur. With very low values, the

resulting voice slowly gets louder and quieter, like a singer that cannot keep his or her breath steady. With very high settings, the result can be jittery and unnatural.

• Highest Quality Ensures the best quality results. Increasing the quality, however, increases the processing time for

previewing and applying the effect.

Stereo Width Determines where the individual voices are placed in the stereo field and how the original stereo signal

is interpreted. These options are active only when you work with stereo files:

• Average Left & Right Channel Input Combines the original left and right channels. If deselected, the channels are

kept separate to preserve the stereo image. Leave this option deselected if the stereo source audio was originally monophonic—it won’t have any effect other than increasing processing time.

• Add Binaural Cues Adds separate delays to the left and right outputs of each voice. This delay can make each voice

seem to come from a different direction when you listen through headphones. For greater stereo separation, deselect this option for audio that will be played through standard speakers .

• Stereo Field Specifies where chorused voices are placed across the left and right stereo image. At lower settings,

voices are closer to the center of the stereo image. At a setting of 50%, voices are spaced evenly from left to right. At higher settings, voices move to the outer edges. If you use an odd number of voices, one is always directly in the center.

Output Level Sets the ratio of original (Dry) signal to chorused (Wet) signal. Extremely high settings may cause

clipping.

In the Multitrack Editor, you can vary the Wet level over time with automation lanes. (See “Automating track

settings” on page 123.)This technique is handy for emphasizing vocal or instrumental solos.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

Chorus/Flanger effect

The 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.

The Flanger option creates a psychedelic, phase-shifted sound by mixing a varying, short delay with the original signal. This effect was originally created by sending an identical audio signal to two reel-to-reel tape recorders, and periodically pressing the flange of one reel to slow it down.

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.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

Flanger effect

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 Modulation 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.

> Flanger effect lets you create a similar

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, cancelling 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.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

Phaser effect

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 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 degrees 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.

> Phaser effect sweeps a series of phase-shifting filters to and from an upper frequency. Phasing can

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.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

Noise reduction / restoration effects

Techniques for restoring audio

You can fix a wide array of audio problems by combining two powerful features. First, use Spectral Display to visually identify and select ranges of noise or individual artifacts. (See

and repair them automatically” on page 40.) Then, use either Diagnostic or Noise Reduction effects to fix problems

like the following:

• Crackle from wireless microphones or old vinyl records. (See “Automatic Click Remover effect” on page 88.)

• Background noise like wind rumble, tape hiss, or power-line hum. (See “Adaptive Noise Reduction effect” on

page 88 and “DeHummer effect” on page 89.)

• Phase cancellation from poorly placed stereo microphones or misaligned tape machines. (See “Automatic Phase

Correction effect” on page 89.)

“Select spectral ranges” on page 39 and “Select artifacts

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

The real-time restoration effects above, which are available in both the Waveform and Multitrack editors, quickly address common audio problems. For unusually noisy audio, however, consider using offline, process effects unique

to the Waveform Editor, such as Hiss Reduction and Noise Reduction.

AB C

Selecting various types of noise in Spectral Display A. Hiss B. Crackle C. Rumble

More Help topics

“About the spectral display” on page 36

“Edit audio clips from Adobe Premiere Pro CS5.5 or After Effects” on page 126

Noise Reduction effect (Waveform Editor only)

The Noise Reduction/Restoration > Noise Reduction effect dramatically reduces background and broadband noise with a minimal reduction in signal quality. This effect can remove a combination of noise, including tape hiss, microphone background noise, power-line hum, or any noise that is constant throughout a waveform.

The proper amount of noise reduction depends upon the type of background noise and the acceptable loss in quality for the remaining signal. In general, you can increase the signal-to-noise ratio by 5 to 20 dB and retain high audio quality.

To achieve the best results with the Noise Reduction effect, apply it to audio with no DC offset. With a DC offset, this effect may introduce clicks in quiet passages. (To remove a DC offset, choose Favorites > Repair DC Offset.)

Evaluating and adjusting noise with the Noise Reduction graph: A. Drag control points to vary reduction in different frequency ranges B. Low amplitude noise. C. High amplitude noise D. Threshold below which noise reduction occurs.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

More Help topics

“About process effects” on page 61

“Control effect settings with graphs” on page 59

Apply the Noise Reduction effect

1 In the Waveform Editor, select a range that contains only noise and is at least half a second long.

To select noise in a specific frequency range, use the Marquee Selection tool. (See “Select spectral ranges” on page 39.)

2 Choose Effects > Noise Reduction/Restoration > Capture Noise Print.

3 In the Editor panel, select the range from which you want to remove noise.

4 Choose Effects > Noise Reduction/Restoration > Noise Reduction.

5 Set the desired options.

When recording in noisy environments, record a few seconds of representative background noise that can be used as a noise print later on.

Noise Reduction options

Capture Noise Print Extracts a noise profile from a selected range, indicating only background noise. Adobe Audition

gathers statistical information about the background noise so it can remove it from the remainder of the waveform.

If the selected range is too short, Capture Noise Print is disabled. Reduce the FFT Size or select a longer range of noise. If you can’t find a longer range, copy and paste the currently selected range to create one. (You can later remove the

pasted noise by using the Edit

Save the Current Noise Print Saves the noise print as an .fft file, which contains information about sample type,

> Delete command.)

FFT (Fast Fourier Transform) size, and three sets of FFT coefficients: one for the lowest amount of noise found, one for the highest amount, and one for the power average.

Load a Noise Print from Disk Opens any noise print previously saved from Adobe Audition in FFT format.

However, you can apply noise prints only to identical sample types. (For example, you can’t apply a 22 kHz mono profile to 44kHz stereo samples.)

Note: Because noise prints are so specific, a print for one type of noise won’t produce good results with other types. If you regularly remove similar noise, however, a saved profile can greatly increase efficiency.

Graph Depicts frequency along the x-axis (horizontal) and the amount of noise reduction along the y-axis (vertical).

The blue control curve sets the amount of noise reduction in different frequency ranges. For example, if you need noise reduction only in the higher frequencies, adjust the control curve downward to the right of the graph.

If you click the Reset button to flatten the control curve, the amount of noise reduction is based entirely on the noise print.

To better focus on the noise floor, click the menu button to the upper right of the graph, and deselect Show Control Curve and Show Tooltip Over Graph.

Noise Floor High shows the highest amplitude of detected noise at each frequency; Low shows the lowest amplitude.

Threshold shows the amplitude below which noise reduction occurs.

The three elements of the noise floor can overlap in the graph. To better distinguish them, click the menu button , and select options from the Show Noise Floor menu.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

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.

Channel Displays the selected channel in the graph. The amount of noise reduction is always the same for all channels.

Select Entire File Lets you apply a captured noise print to the entire file.

Noise Reduction Controls the percentage of noise reduction in the output signal. Fine-tune this setting while

previewing audio to achieve maximum noise reduction with minimum artifacts. (Excessively high noise reduction levels can sometimes cause audio to sound flanged or out-of-phase.)

Reduce By Determines the amplitude reduction of detected noise. Values between 6 and 30 dB work well. To reduce

bubbly artifacts, enter lower values.

Output Noise Only Previews only noise so you determine if the effect is removing any desirable audio.

Advanced settings Click the triangle to display the following options:

• Spectral Decay Rate Specifies the percentage of frequencies processed when audio falls below the noise floor.

Fine-tuning this percentage allows greater noise reduction with fewer artifacts. Values of 40% to 75% work best. Below those values, bubbly-sounding artifacts are often heard; above those values, excessive noise typically remains.

• Smoothing Takes into account the variance of the noise signal in each frequency band. Bands that vary greatly

when analyzed (such as white noise) will be smoothed differently than constant bands (like 60-Hz hum). In general, increasing the smoothing amount (up to 2 or so) reduces burbly background artifacts at the expense of raising the overall background broadband noise level.

• Precision Factor Controls changes in amplitude. Values of 5-10 work best, and odd numbers are ideal for

symmetrical processing. With values of 3 or less, the Fast Fourier transform is performed in giant blocks, and between them drops or spikes in volume can occur. Values beyond 10 cause no noticeable change in quality, but they increase processing time.

• Transition Width Determines the amplitude range between noise and desirable audio. For example, a width of zero

applies a sharp, noise gate to each frequency band. Audio just above the threshold remains; audio just below is truncated to silence. Alternatively, you can specify a range over which the audio fades to silence based upon the input level. For example, if the transition width is 10 dB, and the noise level for the band is -60 dB, audio at -60 dB stays the same, audio at -62 dB is reduced slightly, and audio at -70 dB is removed entirely.

• FFT Size Determines how many individual frequency bands are analyzed. This option causes the most drastic

changes in quality. The noise in each frequency band is treated separately, so with more bands, noise is removed with finer frequency detail. Good settings range from 4096 to 8192.

Fast Fourier Transform size determines the tradeoff between frequency- and time-accuracy. Higher FFT sizes might cause swooshing or reverberant artifacts, but they very accurately remove noise frequencies. Lower FFT sizes result in better time response (less swooshing before cymbal hits, for example), but they can produce poorer frequency resolution, creating hollow or flanged sounds.

• Noise Print Snapshots Determines how many snapshots of noise to include in the captured profile. A value of 4000

is optimal for producing accurate data.

Very small values greatly affect the quality of the various noise reduction levels. With more snapshots, a noise reduction level of 100 will likely cut out more noise, but also cut out more original signal. However, a low noise reduction level with more snapshots will also cut out more noise, but likely retain the intended signal.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

Adaptive Noise Reduction effect

The Noise Reduction/Restoration > Adaptive Noise Reduction effect quickly removes variable broadband noise such as background sounds, rumble, and wind. Because this effect operates in real time, you can combine it with other effects in the Effects Rack and apply it in the Multitrack Editor. By contrast, the standard Noise Reduction effect is available only as an offline process in the Waveform Editor. That effect, however, is sometimes more effective at removing constant noise, such as hiss or hum.

For best results, apply Adaptive Noise Reduction to selections that begin with noise followed by desirable audio. The effect identifies noise based on the first few seconds of audio.

Important: This effect requires significant processing. If your system performs slowly, lower FFT Size and turn off High Quality Mode.

Reduce Noise By Determines the level of noise reduction. Values between 6 and 30 dB work well. To reduce bubbly

background effects, enter lower values.

Noisiness Indicates the percentage of original audio that contains noise.

Fine Tune Noise Floor Manually adjusts the noise floor above or below the automatically calculated floor.

Signal Threshold Manually adjusts the threshold of desirable audio above or below the automatically calculated

threshold.

Spectral Decay Rate Determines how quickly noise processing drops by 60 decibels. Fine-tuning this setting allows

greater noise reduction with fewer artifacts. Values that are too short create bubbly sounds; values that are too long create a reverb effect.

Broadband Preservation Retains desirable audio in specified frequency bands between found artifacts. A setting of

100 Hz, for example, ensures that no audio is removed 100 Hz above or below found artifacts. Lower settings remove more noise but may introduce audible processing.

FFT Size Determines how many individual frequency bands are analyzed. Choose a high setting to increase frequency

resolution; choose a low setting to increase time resolution. High settings work well for artifacts of long duration (like squeaks or power-line hum), while low settings better address transient artifacts (like clicks and pops).

High Quality Mode Performs slower processing but achieves superior results.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

Automatic Click Remover effect

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

This effect provides the same options as the DeClicker effect, which lets you choose which detected clicks to address

“DeClicker options” on page 77). However, because the Automatic Click Remover operates in real time, you can

(see combine it with other effects in the Effects Rack and apply it in the Multitrack Editor. The Automatic Click Remover effect also applies multiple scan and repair passes automatically; to achieve the same level of click reduction with the DeClicker, you must manually apply it multiple times.

Threshold Determines sensitivity to noise. Lower settings detect more clicks and pops but may include audio you wish

to retain. Settings range from 1 to 100; the default is 30.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

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.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

Automatic Phase Correction effect

The Noise Reduction/Restoration > Automatic Phase Correction effect addresses azimuth errors from misaligned tape heads, stereo smearing from incorrect microphone placement, and many other phase-related problems.

Global Time Shift Activates the Left and Right Channel Shift sliders, which let you apply a uniform phase shift to all

selected audio.

Auto Align Channels and Auto Center Panning Align phase and panning for a series of discrete time intervals, which

you specify using the following options:

• Time Resolution Specifies the number of milliseconds in each processed interval. Smaller values increase accuracy;

larger ones increase performance.

• Responsiveness Determines overall processing speed. Slow settings increase accuracy; fast settings increase

performance.

• Channel Specifies the channels phase correction will be applied to.

• Analysis Size Specifies the number of samples in each analyzed unit of audio.

For the most precise, effective phase correction, use the Auto Align Channels option. Enable the Global Time Shift sliders only if you are confident that a uniform adjustment is necessary, or if you want to manually animate phase

correction in the Multitrack Editor.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

DeHummer effect

The Noise Reduction/Restoration > 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.

To quickly address typical audio problems, choose an option from the Presets menu.

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.

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

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

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.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

Hiss Reduction effect (Waveform Editor only)

The Noise Reduction/Restoration > Hiss Reduction effect reduces hiss from sources such as audio cassettes, vinyl records, or microphone preamps. This effect greatly lowers the amplitude of a frequency range if it falls below an amplitude threshold called the noise floor. Audio in frequency ranges that are louder than the threshold remain untouched. If audio has a consistent level of background hiss, that hiss can be removed completely.

To reduce other types of noise that have a wide frequency range, try the Noise Reduction effect. (See “Noise Reduction

effect (Waveform Editor only)” on page 85.)

Using the Hiss Reduction graph to adjust the noise floor

Capture Noise Floor Graphs an estimate of the noise floor. The estimate is used by the Hiss Reduction effect to more

effectively remove only hiss while leaving regular audio untouched. This option is the most powerful feature of Hiss Reduction.

To create a graph that most accurately reflects the noise floor, click Get Noise Floor with a selection of audio that contains only hiss. Or, select an area that has the least amount of desirable audio, in addition to the least amount of high frequency information. (In the spectral display, look for an area without any activity in the top 75% of the display.)

After you capture the noise floor, you might need to lower the control points on the left (representing the lower frequencies) to make the graph as flat as possible. If music is present at any frequency, the control points around that frequency will be higher than they should be.

Graph Represents the estimated noise floor for each frequency in the source audio, with frequency along the

horizontal ruler (x-axis) and the amplitude of the noise floor along the vertical ruler (y-axis). This information helps you distinguish hiss from desirable audio data.

The actual value used to perform hiss reduction is a combination of the graph and the Noise Floor slider, which shifts the estimated noise floor reading up or down for fine tuning.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

To disable tooltips for frequency and amplitude, click the menu button to the upper right of the graph, and deselect Show Tooltip Over Graph.

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.

Channel Displays the selected audio channel in the graph.

Reset Resets the estimated noise floor. To reset the floor higher or lower, click the menu button to the upper

right of the graph, and choose an option from the Reset Control Curve menu.

For quick, general-purpose hiss reduction, a complete noise floor graph isn’t always necessary. In many cases, you can simply reset the graph to an even level and manipulate the Noise Floor slider.

Noise Floor Fine-tunes the noise floor until the appropriate level of hiss reduction and quality is achieved.

Reduce By Sets the level of hiss reduction for audio below the noise floor. With higher values (especially above 20 dB)

dramatic hiss reduction can be achieved, but the remaining audio might become distorted. With lower values, not as much noise is removed, and the original audio signal stays relatively undisturbed.

Output Hiss Only Lets you preview only hiss to determine if the effect is removing any desirable audio.

Advanced settings Click the triangle to display these options:

• Spectral Decay Rate When audio is encountered above the estimated noise floor, determines how much audio in

surrounding frequencies is assumed to follow. With low values, less audio is assumed to follow, and hiss reduction will cut more closely to the frequencies being kept.

Values of 40% to 75% work best. If the value is too high (above 90%), unnaturally long tails and reverbs might be heard. If the value is too low, background bubbly effects might be heard, and music might sound artificial.

• Precision Factor Determines the time-accuracy of hiss reduction. Typical values range from 7 to 14. Lower values

might result in a few milliseconds of hiss before and after louder parts of audio. Larger values generally produce better results and slower processing speeds. Values over 20 don’t ordinarily improve quality any further.

• Transition Width Produces a slow transition in hiss reduction instead of an abrupt change. Values from 5 to 10

usually achieve good results. If the value is too high, some hiss may remain after processing. If the value is too low, background artifacts might be heard.

• FFT Size Specifies a Fast Fourier Transform size, which determines the tradeoff between frequency- and time-

accuracy. In general, sizes from 2048 to 8192 work best.

Lower FFT sizes (2048 and below) result in better time response (less swooshing before cymbal hits, for example), but they can produce poorer frequency resolution, creating hollow or flanged sounds.

Higher FFT sizes (8192 and above) might cause swooshing, reverb, and drawn out background tones, but they produce very accurate frequency resolution.

• Control Points Specifies the number of points added to the graph when you click Capture Noise Floor.

More Help topics

“About process effects” on page 61

“Apply individual effects in the Waveform Editor” on page 60

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

“Use effect presets” on page 59

“Control effect settings with graphs” on page 59

Reverb effects

In a room, sound bounces off the walls, ceiling, and floor on the way to your ears. All these reflected sounds reach your ears so closely together that you don’t perceive them as separate echoes, but as a sonic ambience that creates an impression of space. This reflected sound is called reverberation, or reverb for short. With Adobe Audition, you can use reverb effects to simulate a variety of room environments.

For the most flexible, efficient use of reverb in the Multitrack Editor, add reverb effects to buses, and set reverb output levels to 100% Wet. Then, route tracks to these buses, and use sends to control the ratio of dry to reverberant sound.

More Help topics

“Delay and echo effects” on page 74

“Routing audio to buses, sends, and the Master track” on page 110

Convolution Reverb effect

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

Sources of impulse files include audio you’ve recorded of an ambient space, or impulse collections available online. For best results, impulse files should be uncompressed, 16- or 32-bit files matching the sample rate of the current audio file. Impulse length should be no more than 30 seconds. For sound design, try a variety of source audio to produce unique, convolution-based effects.

Note: 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 predelay 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.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

Full Reverb effect

The Reverb > Full Reverb effect is convolution-based, avoiding ringing, metallic, and other artificial sounding artifacts.

This effect offers some unique options, such as Perception, which simulates room irregularities, Left/Right Location, which places the source off-center, and Room Size and Dimension, which help you realistically simulate rooms that you can customize. To simulate wall surfaces and resonance, you can change the reverb’s frequency absorption by using a three-band, parametric EQ in the Coloration section.

When you change reverb settings, this effect creates a temporary impulse file, which simulates the acoustic environment you specify. This file can be several megabytes in size, requiring a few seconds to process, so you might have to wait before hearing a preview. The results, however, are incredibly realistic and easy to tailor.

Important: The Full Reverb effect demands significant processing; for real-time multitrack use, either pre-render this effect or replace it with Studio Reverb. (See

“Pre-render track effects to improve performance” on page 62.)

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

“Control effect settings with graphs” on page 59

Reverb Settings

Decay Time Specifies how many milliseconds the reverb takes to decay 60 dB. However, depending on the Coloration

parameters, certain frequencies may take longer to decay to 60 dB, while other frequencies may decay much faster. Longer values give longer reverb tails, but they also require more processing. The effective limit is about 6000 milliseconds (a 6-second tail), but the actual tail generated is much longer to allow for decaying into the background noise level.

Pre-Delay Time Specifies how many milliseconds reverb takes to build to its maximum amplitude. Generally, reverbs

build up quickly, and then decay at a much slower rate. Interesting effects can be heard with extremely long pre-delay times of 400 milliseconds or more.

Diffusion Controls the rate of echo buildup. High diffusion values (above 900 milliseconds) give very smooth reverbs,

without distinct echoes. Lower values produce more distinct echoes because the initial echo density is lighter, but the density builds over the life of the reverb tail.

Bouncy echo effects can be obtained by using low Diffusion values and high Perception values. With long reverb tails, using low Diffusion values and somewhat low Perception values gives the effect of a football stadium or similar arena.

Perception Simulates irregularities in the environment (objects, walls, connecting rooms, and so on). Low values

create a smoothly decaying reverb without any frills. Larger values give more distinct echoes (coming from different locations).

If a reverb is too smooth, it may not sound natural. Perception values up to about 40 give simulate typical room variations.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

Room Size Sets the volume of the virtual room, as measured in cubic meters. The larger the room, the longer the

reverb. Use this control to create virtual rooms of only a few square meters to giant coliseums.

Dimension Specifies the ratio between the room’s width (left to right) and depth (front to back). A sonically

appropriate height is calculated and reported as Actual Room Dimensions at the bottom of the dialog box. Generally, rooms with width-to-depth ratios between 0.25 and 4 provide the best sounding reverbs.

Left/Right Location (stereo audio only) Lets you place early reflections off-center. Select Include Direct in the Output

Level section to place the original signal in the same location. Very nice effects are possible with singers slightly off center, 5-10% to the left or right.

High Pass Cutoff Prevents the loss of low-frequency (100 Hz or less) sounds, such as bass or drums. These sounds can

get phased out when using small rooms if the early reflections mix with the original signal. Specify a frequency above that of the sound you wish to keep. Good settings are generally between 80 Hz and 150 Hz. If the cutoff setting is too high, you may not get a realistic image of the room size.

Set Reverb Based On Room Size Sets Decay and Pre-delay times to match the specified room size, producing a more

convincing reverb. If desired, you can then fine-tune the Decay and Pre-Delay times.

Coloration options

To visually adjust Coloration options, drag directly in the graph.

Frequency Specifies the corner frequency for the low and high shelves or the center frequency for the middle band.

For example, to increase reverb warmth, lower the high shelf frequency while also reducing its gain.

Gain Boosts or attenuates reverb in different frequency ranges.

To subtly enhance audio, boost reverb frequencies around the natural frequency of a key sonic element. For a singer’s voice, for example, boost frequencies from 200 Hz to 800 Hz to enhance resonance in that range.

Q Sets the width of the middle band. Higher values affect a narrower range of frequencies, and lower values affect a

wider range.

For distinct resonance, use values of 10 or higher. To boost or cut a wide range of frequencies, use lower values like 2 or 3.

Decay Specifies how many milliseconds the reverb decays before the Coloration curve is applied. Values up to 700

work fine. For more colored reverbs, use lower settings (such as 100 to 250).

Output Level options

Dry Controls the level of original signal included with reverb. Use a low level to create a distant sound. Use a high level

(near 100%) along with low levels of reverberation and reflections to create a sense of close proximity to the source.

Reverberation Controls the level of the dense layer of reverberant sound. The balance between the dry and reverberant

sounds changes perception of distance.

Early Reflections Controls the level of the first echoes to 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 remove audible cues for the room’s size. Half the volume of the Dry signal is a good starting point.

Include Direct Slightly phase-shifts the original signal’s left and right channels to match the location of early reflections

(set by Left/Right Location on the Early Reflections tab).

Sum Inputs Combines the channels of a stereo or surround waveform before processing occurs. Select this option for

faster processing, but deselect it for a fuller, richer reverb.

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

Reverb effect

The Reverb > Reverb effect simulates acoustic spaces with convolution-based processing. It can reproduce acoustic or ambient environments such as a coat closet, a tiled bathroom shower, a concert hall, or a grand amphitheater. The echoes can be spaced so closely together that a signal’s reverberated tail decays smoothly over time, creating a warm and natural sound. Alternatively, Pre-Delay Time can be adjusted to give a sense of room size.

Relative to the Reverb effect, the Full Reverb effect provides more options and better audio rendering. For quick adjustments, however, you may prefer the reduced options set of the Reverb effect.

Important: The Reverb effect demands significant processing; for real-time, multitrack use, either pre-render this effect or replace it with Studio Reverb. (See

Decay Time Sets how many milliseconds it takes for reverb to tail off to infinity (about -96 dB). Use values below 400

for small rooms, values between 400 and 800 for medium-sized rooms, and values above 800 for very large rooms, such as concert halls. For example, enter 3000 milliseconds to create reverb tails for a giant amphitheater.

To simulate rooms that have both echoes and reverb, first use the Echo effect to establish the size of the room, and then use the Reverb effect to make the sound more natural. A Decay Time as little as 300 milliseconds can add perceived

spaciousness to dry sound.

Pre-Delay Time Specifies how many milliseconds reverb takes to build to its maximum amplitude. For a short Decay

Time, the Pre-Delay Time time should also be smaller. In general, a value about 10% as long as the Decay Time sounds most realistic. However, you can create interesting effects by using a longer Pre-Delay Time with a shorter Decay Time.

“Pre-render track effects to improve performance” on page 62.)

Diffusion Simulates natural absorption, reducing high frequencies as the reverb decays. Faster absorption times

simulate rooms full of people, furniture, and carpeting, such as nightclubs and theaters. Slower times (over 1000 milliseconds) simulate empty rooms such as auditoriums, where high frequency reflections are more prevalent.

Perception Changes the characteristics of reflections within a room. Lower values create smoother reverb without as

many distinct echoes. Higher values simulate larger rooms, cause more variation in reverb amplitude, and add spaciousness by creating distinct reflections over time.

A Perception setting of 100 and a Decay Time of 2000 milliseconds or more creates interesting canyon effects.

Dry Sets the percentage of source audio to output. In most cases, 90% works well. To add subtle spaciousness, set the

Dry percentage higher; to achieve a special effect, set the Dry percentage lower.

Wet Sets the percentage of reverb to output. To add subtle spaciousness to a track, keep the Wet percentage lower than

the Dry percentage. Increase the Wet percentage to simulate greater distance from the audio source.

Sum Inputs Combines the channels of a stereo or surround waveform before processing occurs. Select this option for

faster processing, but deselect it for fuller, richer reverb.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

Last updated 11/7/2011

USING ADOBE AUDITION

Effects reference

Studio Reverb effect

Like the other reverb effects, the Reverb > 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.

Room Size Sets the room size.

Decay Adjusts the amount of reverberation decay in milliseconds.

Early Reflections 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.

Stereo 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 of surfaces, such as carpeting and

drapes. Lower settings create more echoes, while higher settings produce a smoother reverberation with fewer echoes.

Dry Sets the percentage of source audio to output with the effect.

Wet Sets the percentage of reverb to output.

More Help topics

“Applying effects in the Waveform Editor” on page 60

“Applying effects in the Multitrack Editor” on page 61

“Use effect presets” on page 59

Surround Reverb effect

The Reverb > 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.)

Note: The effect always inputs 100% of the Left, Right, and rear surround channels.

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.

Last updated 11/7/2011

Macromedia Audition - CS5.5 Instruction Manual

Specifications and Main Features

Frequently Asked Questions

User Manual

Legal notices

Contents

Chapter 1: What’s new

Chapter 2: Digital audio fundamentals

Understanding sound

Sound waves

Waveform measurements

How sound waves interact

Digitizing audio

Comparing analog and digital audio

Understanding sample rate

Understanding bit depth

Measuring amplitude in dBFS

Audio file contents and size

How Adobe Audition digitizes audio

Chapter 3: Workspace and setup

Viewing, zooming, and navigating audio

Comparing the Waveform and Multitrack editors

Basic components of the editors

Switch editors

Zoom audio in the Editor panel

Navigate through time

Navigate by scrolling

Navigate with the Selection/View panel

Customizing workspaces

About workspaces

Choose a workspace

Dock, group, or float panels

Resize panel groups

Open, close, and scroll to panels

Working with multiple monitors

Display the toolbar

Display the status bar

Change interface colors, brightness, and performance

Save, reset, or delete workspaces

Connecting to audio hardware

Configure audio inputs and outputs

Assign file channels to inputs and outputs

Customizing and saving application settings

Customize preferences

Restore preferences to default settings

Export and import customized application settings

Chapter 4: Importing, recording, and playing audio

Creating and opening files

Create a new, blank audio file

Create a new multitrack session

Open existing audio files and multitrack mixes

Import a file as raw data

Insert an audio file into a multitrack session

Spot-insert a Broadcast Wave file into a session

Importing with the Files panel

Import files into the Files panel

Insert from the Files panel into a multitrack session

Change displayed metadata in the Files panel

Supported import formats

Audio file formats

Video file formats

Extracting audio from CDs

Extract CD tracks with the Extract Audio From CD command

Extract CD tracks with the Open command (Mac OS)

Navigating time and playing audio

Monitoring time

Position the current-time indicator

Preview audio by scrubbing

Play audio linearly or in a loop

Synchronize the current-time indicator across files or views

Change the time display format

Recording audio

Record audio in the Waveform Editor

Correct DC offset

Direct-to-file recording in the Multitrack Editor

Record audio clips in the Multitrack Editor

Punch into a selected range in the Multitrack Editor

Punch in during playback in the Multitrack Editor

Choose punch-in takes

Monitoring recording and playback levels