Audio Editing

Audio Editing

© 2013 Aptech Limited

All rights reserved.

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mechanical, including photocopying, recording, taping, or storing in information retrieval system or sent
or transferred without the prior written permission of copyright owner Aptech Limited.

All trademarks acknowledged.

APTECH LIMITED

Contact E-mail: ov-support@onlinevarsity.com

Edition 1 – 2013

Disclaimer: Arena Multimedia is registered Brand of Aptech Ltd.

Preface

This book, Audio Editing, introduces you to the Adobe’s Audition CS6 software. The book explains the basic features of
Audition and describes how audio can be recorded, edited, and given various effects that are offered by the software.

The ARENA Design team has designed this course keeping in mind that motivation coupled with relevant training and
methodology can bring out the best. The team will be glad to receive your feedback, suggestions, and recommendations
for improvement of the book.

ARENA Design Team

 Table of Content

Introduction to Adobe Audition

Fundamentals of Sound
Understanding Adobe Audition Interface
Adjusting Audio Preferences
Creating and Opening Files
Summary
Exercise

Working with Audio Editing

Editing Audio Clips
Saving and Exporting Files
Batch Process
Converting Sample Type
Summary
Exercise

Working with Multitrack Editor and Recording Audio

Working with Multitrack Editor
Recording Audio
Working with Audio in Video Files
Summary
Exercise

Working with Audio Effects

Invert, Reverse, and Silence Effects
Amplitude and Compression
Delay and Echo
Diagnostics
Filter and EQ
Modulation
Noise Reduction/Restoration
Reverb Effects
Special Effect
Stereo Imaginary Effects
Time and Pitch
Summary
Exercise
 Iconography

: System Requirements



: Note

: Tip

: Quick Test

: Quick Answers

: Exercise Answers
 S e s s io n 1
Introduction to Adobe Audition
Learning Outcomes
In this session, you will learn to:
¾¾ Explain sound
¾¾ Differentiate between analog and digital audio
¾¾ List the technical terms used in audio
¾¾ Describe the elements available in the Adobe Audition interface
¾¾ Configure input and output devices using Adobe Audition
¾¾ Work with audio files in Adobe Audition

Adobe Audition CS6 is a post-production application. It comes with a wide range of professional tools that contribute in
effective production of sound files.

System Requirements
• Windows Operating System

• Intel® Pentium® 4 or AMD Athlon® 64 processor

• Microsoft Windows XP with Service Pack 3 or Windows 7 with Service Pack 1

• 2 GB of RAM (3 GB recommended)

• 3.5 GB of available hard-disk

• Java™ Runtime Environment 1.6

• Graphics card with at least 64 MB of VRAM

• QuickTime 7.6.6 software required for multimedia features

• 1024×768 display (1280×800 recommended)

1.1 Fundamentals of Sound

1.1.1 Sound
Imagine watching the sea closely where the waves that are produced follow a particular pattern. Sound also travels in
a similar manner. It travels through air in the form of longitudinal waves that are generated when air molecules vibrate
and propagate in a specific pattern from one medium to another. Refer to Figure 1.1.

Figure 1.1: Illustration depicting the sound waves

For better understanding, consider the sound produced while clapping hands. The sound waves that are produced travel
through the ears to the brain. The brain identifies the action by recognizing the sound.

Acoustics is the branch of science that deals with sound. Sound can be distinguished based on the following properties:
¾¾ Frequency
¾¾ Wavelength
¾¾ Period
¾¾ Amplitude
¾¾ Speed

The terms, sound and noise are often used interchangeably. However, there is a difference between the two. Sound
is a pleasant vibration sensed by the ear; whereas, noise is the inconsistency found in the sound. Refer to Figure 1.2.

Figure 1.2: Difference in waveforms between sound and noise

1.1.2 Analog Audio Versus Digital Audio

Analog and digital are the two formats in which sound is recorded and stored. Refer to Table 1.1 for differences between
the two formats.

Analog Format Digital Format

¾¾ In this format, sound is stored in the form of waves
that are produced when objects vibrate.
¾¾ Waves move in a specific pattern, known as the
waveform.
¾¾ Curves obtained in the waveform are used to plot
a graph of intensity or wave motion against time.
Refer to Figure 1.3.
¾¾ The air vibration of sound converted into electrical
signal is known as digital signal.
¾¾ Digital signals are a stream of 0’s and 1’s and are
used in computer applications. Refer to Figure
1.4.

Figure 1.3: Analog signal Figure 1.4: Digital signal

Table 1.1: Difference between analog format and digital format

1.1.3 Technical Terms Used in Audio
Following are some commonly used technical terms that users come across while creating audio:
¾¾ Mono: Monophonic, commonly referred to as mono, is a sound system in which all the audio signals are
combined and routed through a single channel. The advantage of this system is that sound is transmitted to all
the listeners at the same signal and sound level.
¾¾ Stereo: Stereophonic sound, commonly known as stereo, is a sound system comprising two channels, namely,
left and right.
¾¾ Decibel: Decibel, represented as dB, is a unit of measurement that measures sound based on human hearing.
It is a logarithmic unit. Since ears perceive loudness in a logarithmic scale, this unit is used to measure sound.
¾¾ Cycle: Phenomena that occur at regular intervals are referred to as cycles. The life cycle of a plant is one of the
natural phenomena that occur in a cycle. Coming to sounds, certain objects vibrate with loud noises. When an
object vibrates, the movement from its starting position to a point of maximum displacement in one direction,
followed by the maximum displacement in the opposite direction, and back to its starting position is called one
cycle.
¾¾ Frequency: Frequency is measured in hertz (Hz). It is the number of cycles in a sound wave per second, where
1 cycle is 1 Hz. Higher the frequency, higher the musical pitch. Refer to Figure 1.5.

Figure 1.5: Frequency

¾¾ Sample Rate: The process of measuring audio signals at specific intervals is referred to as sampling. A sample
rate is an amplitude value of a waveform measured against time. Higher the sampling rate, better the quality
of the audio signal and vice versa. For example, the sample rate for CD quality audio is 44,100 samples per
second. Refer to Figure 1.6 to view differences between the low, medium, and high sampling rates.
Click here to know more about sampling rates.

Figure 1.6: Low, medium, and high sampling rates

 ¾¾ Amplitude: In simple terms, amplitude or the height of the sound wave is the measure of the amount of energy
in the wave. The amplitude is the height between the topmost and bottommost part of the wave. The topmost
part of the wave is called the peak and the bottommost part of the wave is called the trough.
¾¾ Pitch: It is the response of the ears to the frequency of the sound. It helps listeners distinguish between high and
low sounds and depends on the frequency of the sound wave. A high-pitch sound is equivalent to high frequency
and a low-pitch sound is equivalent to low-frequency.
¾¾ MIDI: Musical Instrument Digital Interface, the expanded for of MIDI, is a technical standard that is used to
communicate between musical instruments, computers, and other related devices. Using this interface, more
than one instrument can be played from a single controller.

1.2 Understanding Adobe Audition Interface

Like any other application, the Adobe Audition interface is loaded with a lot of features. Following elements are present
in the Adobe Audition interface:
¾¾ Toolbar: This provides quick access to the tools.
¾¾ Editor: This panel displays the currently selected editor - Waveform or Multitrack.
●● Waveform: Using this editor, users can edit only one file at a time.

●● Multitrack: Using this editor, users can integrate multiple files at a time.
¾¾ Effect Rack: This panel displays the currently selected tracks.
¾¾ Files: This panel displays the currently loaded files.
¾¾ History: This panel displays a history of operations that have been applied to the currently opened file.
¾¾ Levels: This panel displays the sound levels used in the files open either in the Waveform editor or Multitrack
editor.
¾¾ Markers: This panel displays the file name, start point, duration, type, and other such information pertaining to
the file that is being edited.
¾¾ Media Browser: This panel is like an explorer that helps users look for drives, folders, and files on the computer.
¾¾ Mixer: This tab is an alternative way of mixing sound track while working with multitrack projects.
¾¾ Properties: This panel displays the properties of the selected file.
¾¾ Selection/View: This panel displays the start, end, and duration of the current selection.
¾¾ Transport: This panel is used to record the audio and playback control.
¾¾ Zoom: This toolbar consists of eight buttons that can be used to zoom in or zoom out the tracks while editing.
¾¾ Video: This panel is used to insert and preview a video file.
¾¾ Workspace: A space in the application window where panels are organized.

Refer to Figure 1.7 to view the elements available in the Adobe Audition interface.

Using Adobe Audition, it is possible to open a Multitrack editor audio file in the Waveform editor. To do so,
double-click the clip in the Multitrack editor. Alternatively, right-click the clip in the Multitrack editor and choose
Edit Source File. In addition to the two methods, users can also press 0 and 1 to switch between the Waveform
and Multitrack editor panels respectively.
 Figure 1.7: Adobe Audition interface

Quick Test 1.1

1. Sound travels in the form of latitudinal waves.

True False
2. Waveform editor is best for integrating more than one file at a time.
True False

1.3 Adjusting Audio Preferences

With Adobe Audition, users can make use of a wide range of hardware input and output devices. For example, using
sound card inputs, audio can be imported from microphones, tape decks, and digital effect units. On the other hand,
using sound card outputs, users can monitor audio through speakers and headphones. Refer to Figure 1.8.

Figure 1.8: Preferences dialog box in Adobe Audition

■■ Configuring audio inputs and outputs

Following sound card drivers can be used for recording and playback in Adobe Audition:
¾¾ ASIO drivers
¾¾ MME drivers
¾¾ CoreAudio drivers
ASIO drivers support professional cards; whereas, MME drivers support standard cards. ASIO and CoreAudio
drivers provide better performance and lower latency. Using these drivers, it is possible to monitor audio while
recording it. Users can edit the volume, pan, and other effects during playback.

Steps to configure audio are as follows:

1. From the Edit menu, choose Preferences. Then, choose Audio Hardware.

2. From the Device Class menu, select the required driver.

3. Select the default input and output from the card.

4. For the Master Clock, select the input or output to which the other digital audio hardware has to synchronize.

5. For the Latency, specify the lowest setting possible without any audio dropouts.

6. Choose a sample rate for the audio hardware.

1.4 Creating and Opening Files

1.4.1 Create a New Audio File
To create a new audio file, from the File menu, click New. Then, click Audio File. A New Audio File dialog box appears
where the user can enter a file name, select the channels, sample rate, and bit depth. Description of the fields in the
New Audio File dialog box is as follows:
¾¾ Sample Rate: This option determines the frequency range of the file.
¾¾ Channels: This option determines if the waveform is mono, stereo, or 5.1 surround.
¾¾ Bit Depth: This option determines the amplitude range of the file. Maximum processing flexibility is achieved
when the amplitude is set to 32 bit.

Refer to Figure 1.9.

Figure 1.9: New Audio File dialog box

1.4.2 Append Audio Files in Audition
Files with CD track markers are appended to audio files to quickly assemble audio and apply consistent processing.
While in the Waveform editor, users can either append CD track markers to the active file or to the new file.
¾¾ To append to the active file, from the File menu, click Open Append, and then click To Current.
¾¾ To append to a new file, from the File menu, click Open Append, and then click To New.
In either case, the Open Append dialog box appears. Select the desired files, and then click Open.

If the files that are appended to the CD track marker have a different sample rate, bit depth, or channel type as
compared to the open file, Adobe Audition converts the specifications of the appended file to match with that
of the open file.

1.4.3 Audio Import Formats

Adobe Audition can open or import audio files in any of the following formats:
¾¾ AAC, including HE-AAC in Audition CS6
¾¾ AIF, AIFF, AIFC, including uncompressed files, compressed files, and files with up to 32 channels
¾¾ WAV, including uncompressed files, compressed files, and files with up to 32 channels
1.4.4 Import File as Raw Data
To import file as raw data, from the File menu, click Import, and then click Raw Data. Select the desired file, and click
Open. A dialog box appears with the following options:
¾¾ Sample Rate: In Adobe Audition, raw data can be imported with sample rates ranging from 1 to 10,000,000 Hz.
However, playback and recording are supported only between 6000 Hz and 1,92,000 Hz.
¾¾ Channels: Channels can be anything between 1 to 32.
¾¾ Encoding: This option specifies the data storage scheme for the file.
¾¾ Byte Order: This option specifies the numerical sequence for bytes of data.
¾¾ Start Byte Offset: This option specifies the data file at which the import process should begin.
Refer to Figure 1.10.

Figure 1.10: Importing file as raw data

1.4.5 Video Import Formats
The Waveform editor allows users to open the audio portion of video files in any of the following formats:
¾¾ AVI
¾¾ DV
¾¾ MOV
¾¾ MPEG-1
¾¾ MPEG-4
¾¾ 3GPP
¾¾ 3GPP2

To enable the following formats, users have to select Enable DLMS Format Support in the Media & Disk Cache
preferences.
¾¾ AVI (Windows only)
¾¾ FLV
¾¾ R3D
¾¾ SWF
¾¾ WMV

1.4.6 Close Files

Following are the various ways in which users can close
files in Adobe Audition:
¾¾ To close the current file in the Editor panel, from
the File menu, click Close.
¾¾ To close all open audio, video, and session files,
from the File menu, click Close All.
¾¾ To close files that are not referenced by an open
multitrack session, from the File menu, click
Close Unused Media.
¾¾ To close the current session and related audio
clips in the Multitrack Editor, from the File
menu, click Close Session and Its Media.

1.4.7 Extract Audio from CD

Tracks on audio CDs are in a specific format and cannot
be dragged directly into the editor without processing. To
do so, from the File menu, click Extract Audio from CD.
Refer to Figure 1.11. Figure 1.11: Extract Audio from CD dialog box

1.5 Summary
In this session, Introduction to Adobe Audition, you learned that:
¾¾ Sound travels through air in the form of longitudinal waves that are generated when air molecules vibrate and
circulate in a specific pattern from one medium to another.
¾¾ Noise is the inconsistency found in the sound.
¾¾ Acoustics is the branch of science that deals with sound.
¾¾ Sound can be distinguished based on its frequency, wavelength, period, amplitude, and speed.
¾¾ Analog and digital are the two formats in which sound is recorded and stored. In the analog format, sound is
stored in the form of waves; whereas, in the digital format, sound is converted into electrical signals.
¾¾ Adobe audition allows users to make use of a wide range of input and output devices. AISO drivers, MME
drivers, and CoreAudio drivers can be used for audio recording and playback.
 ¾¾ In addition to creating and opening new files, Adobe Audition gives the option of appending files either to the
active file or to the new file using the Open Append command.
¾¾ Using Adobe Audition, it is possible to import audio and video files. However, the format of the files to be
imported has to be looked for. Users can make use of the Enable DLMS Format Support command to enable
the AVI (Windows only), FLV, R3D, SWF, and WMV formats in Adobe Audition.

1.6 Exercise
1. Bit depth determines the amplitude range of the file.

True

False

2. Since the terms, sound and noise have the same meaning, they can be used interchangeable.

True

False

3. The number of cycles in a sound wave per second that is measured in hertz is known as _______.

Decibel
Stereo

Sample rate

Frequency

4. In the analog format, sound is stored in the form of waves and converted into electrical signals.
True

False

5. The currently selected tracks are displayed in the __________.

Effect Rack

Markers panel

Mixer tab

Transport panel

6. MME drivers support standard cards.

True

False

7. While importing files, the _________ option specifies the data storage scheme for the files.

Byte order

Encoding

Sample rate

Channels
 S e s s io n 2
Working with Audio Editing
Learning Outcomes
In this session, you will learn to:
¾¾ Edit audio clips
¾¾ Save and export audio files
¾¾ Process files using the Batch Process command
¾¾ Manipulate the sample rate and bit depth of audio files

2.1 Editing Audio Clips

On inserting an audio file in the Multitrack Editor, it becomes a clip on the selected track. In Adobe Audition, it is possible
to move, edit, and perform other processes on audio files with ease.

2.1.1 Selecting and Moving Clips

Following are the ways in which you can select audio clips:
¾¾ Click at the beginning of the audio clip. Drag to select the section of the clip that has to be moved. The highlighted
area will appear white.
¾¾ Alternatively, keeping the Time Selection tool selected, click at the point on the timeline where you want the
selection to start. This will position the playhead. Then, type I to set an in. Drag the playhead where you want
the selection to end and type O. Refer to Figure 2.1.

Figure 2.1: Selecting audio clips

To move the selected clips, click the Move tool and then, drag the selection to the desired location - either to a different
track or to a location on the timeline.

It is possible to extend or shorten a selection by dragging the selection edges.

2.1.2 Cropping and Deleting Clips

It is possible to trim (crop) or delete audio clips to suit the needs of a mix.
¾¾ To delete an audio clip, from the Edit menu, click Delete.
 ¾¾ To crop an audio clip, from the Edit menu, click Crop.
2.1.3 Copying and Pasting Clips
■■ Copying audio clips

It is possible to create two types of copied audio clips - reference copies and unique copies. The type of copy to be
created depends on the available disk space and the nature of editing that has to be performed.

Reference copies do not require additional disk space. Here, changes made to the original source file reflect in all
its instances.

Unique copies have a separate audio file on the disk. In other words, these files require additional disk space.
Instances of such files can be edited separately.

To copy a file, right-click the clip using the Move tool or the Time Selection tool and choose Copy Here or Copy
Unique Here.

Copy Here creates a reference copy; whereas, Copy Unique Here creates a unique copy.

Pressing the Ctrl+C keys creates a reference copy of the selected audio clip.

■■ Pasting audio clips

To paste an audio clip, begin by selecting the current time indicator or by selecting the existing audio that has to be
replaced.
¾¾ To replace the existing audio, from the Edit menu, choose Paste.
¾¾ To paste audio into a new file, from the Edit menu, choose Paste to New. The new file automatically opens in
a new window.
■■ Mixing audio clips

In the Editor panel, it is possible to adjust the levels of copied audio and existing audio. To do so, from the Edit
menu, choose Mix Paste. In the dialog box that appears, select the Copied and Existing Audio, Invert Copied
Audio, and Crossfade options and click OK.

Description of the options are as follows:

¾¾ Copied and Existing Audio: This option is used to adjust the percentage volume of both - copied audio and
existing audio.
¾¾ Invert Copied Audio: This option reverses the phase of copied audio.
¾¾ Crossfade: This option applies a crossfade to the beginning and end of the copied audio.
2.1.4 Snapping
Snapping aligns clips with each other. On enabling snapping, dragged clips and the current time indicator snap to the
selected items, like Markers, Rulers, Zero-Crossing, and Frames.

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

2.1.5 Markers
On inserting markers, the name, duration, start and end points, and other such relevant information is displayed in the
Markers panel.
¾¾ Clicking the Add Cue Marker icon in the Marker area creates a Range Marker in the timeline. Alternatively,
markers can be inserted by pressing the M key.
¾¾ Double-clicking a marker lets users navigate with ease.
¾¾ To delete a marker, select the desired marker and click the Delete Selected Markers icon. Alternatively, users
can right-click a marker and click Delete from the shortcut menu.
¾¾ The shortcut menu that appears on right-click has the following options:
●● Delete
 ●● Rename

●● Show in Markers Panel

●● Convert to Range

●● Change Marker Type

¾¾ To save a marker, from the File menu, click Save As. Using the Save As command creates an alternate version
of the original .wav file. Refer to Figure 2.2.

Figure 2.2: Markers

2.1.6 Navigation Zooming

The Zoom toolbar consists of eight buttons that control the zoom levels. Refer to Figure 2.3.

Figure 2.3: Options available in the Zoom toolbar

2.1.7 Zero Crossing

On defining a region, when the boundary occurs at a place where the waveform transitions from positive to negative or
there is no rapid level change, it is called zero-crossing.

In case if a region does not fall on a zero-crossing, Audition automatically optimizes the region such that it falls on a
zero-crossing. To apply a zero-crossing, from the Edit menu, click Zero Crossing. The following options are available:
¾¾ Adjust Selection Inward: This option brings the region boundaries closer so that each boundary falls on the
nearest zero-crossing.
¾¾ Adjust Selection Outward: This option moves the region boundaries apart so that each boundary falls on the
nearest zero-crossing.
¾¾ Adjust Left Side to Left: This option moves the left region boundary to the nearest zero-crossing to the left.
¾¾ Adjust Left Side to Right: This option moves the left region boundary to the nearest zero-crossing to the right.
¾¾ Adjust Right Side to Left: This option moves the right region boundary to the nearest zero-crossing to the left.
¾¾ Adjust Right Side to Right: This option moves the left region boundary to the nearest zero-crossing to the
right.
Refer to Figure 2.4.

Figure 2.4: Zero Crossings

2.1.8 Fading and Gain Clips

To make transitions smoother in sound bites, it is a good idea to apply a slight fade to every audio file in a track. To apply
a fade, click and drag the Fade handles. These handles appear only in a selected audio track.

Visual fade and gain clips make the task quick and intuitive. A preview is displayed on dragging these controls. This
helps in adjusting the audio with precision. Refer to Figure 2.5.

Figure 2.5: Fade controls

Dragging the Fade icon inward determines the fade length and dragging it up or down adjusts the fade curve. Following
three types of visual fades are available in Adobe Audition:
¾¾ Linear fade: Dragging the Fade handles perfectly horizontally creates a linear fade.
¾¾ Logarithmic fade: Dragging the Fade handles up or down creates a logarithmic fade.
¾¾ Cosine fade: This curve is also known as the S curve. Keeping the Ctrl key pressed while dragging the handle
creates a cosine curve.
To fade the selected audio with ease, from the Favorite menu, click Fade In and Fade Out.

■■ Gain clips

To access the gain clips, from the Editor panel, make the desired selection to adjust the entire file. Drag the knob
or numbers that appear in the gain control. The numbers indicate the comparison between the new and existing
amplitudes. On releasing the mouse button, the numbers return to 0 dB. This helps in making further adjustments.
Refer to Figure 2.6.

Figure 2.6: Gain controls

2.1.9 Spectral Frequency and Spectral Pitch Display

■■ Spectral display

The spectral display shows a waveform by its frequency components. The x‑axis measures time and the y‑axis
measures frequency. This display allows users analyze audio data to see the most prevalent frequencies. In the
spectral display, brighter colors represent greater amplitude components. Colors range from dark blue to bright
yellow, where dark blue signifies low amplitude frequencies and bright yellow signifies high-amplitude frequencies.
Clicking the Spectral Display button displays the spectral display. Refer to Figure 2.7.
■■ Select spectral range

While working in a spectral display, it is possible to use the Marquee tool, Lasso tool, or Paintbrush Selection
tool to select audio data within a specific spectral range. While the Marquee Selection tool is used to select a
rectangular area, the Lasso Selection and Paintbrush Selection tools lets users make free‑form selections. The
Spot Healing Brush tool is used to repair small, individual audio artefacts, like isolated clicks or pops. Refer to
Figure 2.8.
 Figure 2.7: Spectral display

Figure 2.8: Tools available while working with the spectral display

On making a selection on a stereo waveform, the selection is applied to all the channels by default. Inorder to
select audio data in specific channels, users can choose the Enable Channels command from the Edit menu.

■■ Spectral pitch display

Most sounds elicit spectral pitch. The fundamental pitch is displayed as a bright blue line; whereas, the overtones
appear in yellow to red hues. On correcting the pitch, the display turns to a bright green line. Refer to Figure 2.9.
 Figure 2.9: Spectral pitch display

Quick Test 2.1

1. Markers can be inserted using the N key.

True False

2. The Spot Healing Brush tool is used for selecting audio data.

True False

2.2 Saving and Exporting Files

2.2.1 Saving Audio Files
In the Waveform editor, audio files can be saved in a variety of formats. The usage of the file defines the format in which
the file is saved.
¾¾ From the File menu, click Save, to save changes to the current file.
¾¾ From the File menu, click Save As, to save changes under a different file name.
¾¾ From the File menu, click Save Selection As, to save the currently selected audio as a new file.
¾¾ From the File menu, click Save All, to save all open files in their current formats.
2.2.2 Exporting Audio Files
Once the audio file is ready after all edits, the next step is to export the project. To do so, from the File menu choose
Export. In the Export File dialog box that appears, specify the file name and location. Also, choose the format (WAV or
Mp3) in which the audio has to be saved from the Format drop-down list. Then, click OK.

The audio file is not only available in the specified path but also available in the Files area of Adobe Audition. Refer to
Figure 2.10.
 Figure 2.10: Export File dialog box

2.2.3 Extract Audio Channels to Mono Files

To be able to edit individual channels of a stereo or surround sound file, it is necessary to convert them to mono files.
On conversion, Adobe Audition appends the channel name to each extracted file name and automatically opens each
extracted file in the Editor panel.

To extract audio channels in the Waveform editor, from the Edit menu, choose Extract Channels to Mono Files.

2.2.4 Export Session Templates

Session templates comprise multitrack properties and clips that can be used across various projects. These templates
help in quickly starting projects that require similar settings and tasks.

To export session templates, from the File menu, point to Export, and then click Session As Template. At this juncture,
you can specify a name and location for the file.

There could be instances when you have to apply a template to a new session. To do so, from the File menu, point to
New, and then click Multitrack Session. Then, choose an option from the Template menu.

2.2.5 Export Multitrack Mixdown

Usually, after mixing a session, all of it or a part of it, has to be exported in a variety of common formats that users
can access. On exporting, the resultant file reflects the current volume, pan, and effects settings. These settings are
automatically applied to the Master track.

Steps to export an entire session or a part of it are almost the same. To export a part of a session, begin by selecting the
desired range with the Time Selection tool.

From the File menu, point to Export, and then click Multitrack Mixdown. To complete the export, specify the name of
the file, location to save the file, and file format.
2.2.6 Export Sessions to OMF or Final Cut Pro Interchange Format
Open Media Format (OMF) is a common multitrack exchange format that supports many audio mixing applications. Final
Cut Pro Interchange format is based on human-readable XML files that can be edited offline to revise text references,
mix settings, and such other parameters.

These formats are mostly preferred to transfer complete mixes to other applications in the workflow.

2.3 Batch Process

Batch processing is a method to save time and effort by automating repetitive tasks. It groups files to expedite processing,
resampling, or any other tasks.

In a batch process, to convert and export files to a specific format, drag the files from the desktop, media browser, or
Files panel into the Batch Process panel. Refer to Figures 2.11 and 2.12.
 Figure 2.11: Batch Process panel

Figure 2.12: Export Settings dialog box

To batch process files, from the Window menu, click Batch Process. In the Batch Process panel that appears, click
the Add Files button to browse files on the computer.
To process new files, select the Export check box and click Run. In the Export Settings dialog box that appears, define
values for the following options as per the requirement:
¾¾ Filename Prefix and Postfix: This option aids in identifying the batched files.
¾¾ Template: This option sets the naming convention for the processed files.
¾¾ Location: This option defines the destination folder of the processed files.
¾¾ Format: This option specifies the file format.
¾¾ Sample Type: This option indicates the sample rate and bit depth.
¾¾ New Sample Type: This option indicates the sample rate and bit depth after export.

To process existing files, uncheck the Export check box and click Run.

2.4 Converting Sample Type

The sample rate of a file determines the frequency range of the waveform. While converting sample rates, only certain
sample rates are supported by sound cards. Refer to Figure 2.13.
■■ Convert the sample rate of a file

1. To convert the sample rate, in the Waveform editor, from the Edit menu, click Convert Sample Rate.

2. Select a rate from the Sample Rate list. Alternatively, you can enter a value in the Sample Rate text box.

3. To adjust the quality of the rate, in the Advanced section, drag the Quality slider to desired percentage.
4. To prevent any aliasing noise, select the Pre/Post Filter check box.
■■ Convert a waveform between surround, stereo, and mono

Using the Convert Sample Type command, users can convert a waveform to a number of channels. To do so, in
the Waveform editor, from the Edit menu, click Sample Type. Select the channel in the Channels section. Channels
can be either Mono, or Stereo, or 5.1. In the Advanced section, enter percentages for Left Mix and Right Mix.
■■ Change the bit depth of a file

The dynamic range of any audio is determined by the bit depth of a file. Adobe audition supports up to 23-bit
resolution. Raising the bit depth of a file provides a greater dynamic range; however, the file size increases. Lowering
the bit depth of the file reduces the file size.

1. To change the bit depth of a file, in the Waveform editor, from the Edit menu, choose Convert Sample Type.

2. Select a bit depth from the Bit Depth drop-down list. Alternatively, users can enter a value in the Bit Depth text box.

3. In the Advanced section, define the values for the following options:
¾¾ Dithering: This option enables or disables dithering on reducing the bit depths.
¾¾ Dither Type: This option controls the distribution of the dithering noise, in relation to the original amplitude
values.
¾¾ Noise Shaping: This option defines the frequency that contains dithering noise.
¾¾ Crossover: This option defines the frequency above which noise frequency will occur.
¾¾ Strength: This option defines the maximum amplitude of noise added to a particular frequency.
¾¾ Adaptive Mode: This option varies the distribution of noise across frequencies.

Figure 2.13: Convert Sample Type dialog box

2.5 Summary
In this session, Working with Audio Editing, you learned that:
¾¾ It is possible to move, edit, and perform other processes on audio files with ease.
¾¾ On copying an audio clip, it can be saved either as a reference copy or as a unique copy. Reference copies do
not require additional disk space. Changes made to an instance of a reference copy, reflects across all files.
Unique copies require additional disk space and each instance can be edited separately without affecting other
files.
¾¾ Snapping aligns clips with each other.
¾¾ When the boundary of a selection of an audio clip occurs at a place where the waveform transitions from
positive to negative or there is no rapid level change, it is called zero-crossing.
¾¾ Applying a visual fade or gain clips on audio files, in a sound track makes the transitions smoother and quick.
¾¾ The spectral display shows a waveform by its frequency components. The x-axis measures time and the y-axis
measures frequency.
¾¾ In a spectral display, the fundamental pitch is displayed as a bright blue line, the overtones appear in yellow to
red hues, and the corrected pitch appears as a bright green line.
¾¾ The format for saving a file depends on its usage. The options available for saving a file are Save, Save As,
Save Selection As, and Save All.
¾¾ On exporting a file, it is saved in the final format (WAV or Mp3) in which users can access the file.
¾¾ On converting audio files to mono files, individual channels of a stereo or surround file can be edited.
¾¾ Session templates comprise multitrack properties and clips that can be used across various projects. These
templates help in quickly starting projects that require similar settings and tasks.

¾¾ Open Media Format (OMF) is a common multitrack exchange format that supports many audio mixing
applications. Final Cut Pro Interchange format is based on human-readable XML files that can be edited offline
to revise text references, mix settings, and such other parameters.
¾¾ Batch processing is a method to save time and effort by automating repetitive tasks. It groups files to expedite
processing, resampling, or any other tasks.
¾¾ The sample rate of a file determines the frequency range of the waveform. While converting sample rates, only
certain sample rates are supported by sound cards.

2.6 Exercise
1. The __________ tool is used to move audio clips.

Time Selection

Move

Crop

Copy Unique Here

2. A crossfade is applied to an audio while _________________.

Snapping audio clips

Mixing audio clips

Pasting audio clips

Extracting audio clips

3. On using the Copy Here command, a unique copy of the audio file is created that requires no additional disc space.

True

False
4. T
he ____________ option brings the selection boundaries closer such that each boundary falls on the nearest
zero-crossing.

Adjust Selection Outward

Adjust Left Side to Right Side

Adjust Selection Inward

Adjust Right Side to Right

5. On choosing the Save Selection As command, ____________________.

Changes are saved under a different file name

Changes are saved to the currently selected audio as a new file

Changes are saved to the current file

Changes are saved to all open files in the current format

6. To process existing files, the Export check box should be selected.

True

False

7. The __________ of a file defines the dynamic range of any audio.

Bit depth

Waveform

Noise

Sample rate

8. While working with gain clips, ____________ indicate the comparison between the new and existing amplitudes.

Numbers

Start and end points

Noise

Waves
 Session 3
Working with Multitrack Editor and Recording Audio
Learning Outcomes
In this session, you will learn to:
¾¾ Explain the working of Multitrack Editor
¾¾ Work with Split, Trim, Ripple Delete, and Group Clip tools
¾¾ Automate mixes with envelopes and use Track Controls
¾¾ Operate Automatic Speech Alignment, Match Clip Volume, and Mixer Panel
¾¾ Explain how to record audio using Waveform editor and Multitrack Editor
¾¾ Explain how to insert and work with audio in video files

3.1 Working with Multitrack Editor

The Multitrack Editor is used to assemble clips, add effects, change levels and pan them, and create buses for routing
tracks to various effects.

3.1.1 Overview of Multitrack Editor

There are instances wherein multiple audio tracks need to be worked with in order to develop a composition. To enable
this, the Multitrack Editor is used.

Multitrack Editor mixes together multiple audio tracks, which creates layered soundtracks, and consequently an elaborate
musical composition gets developed. This tool enables recording and mixing unlimited tracks, and each of these tracks
can contain as many clips as required.

The Multitrack Editor is extremely flexible and edits the soundtracks in a real‑time editing environment. Thus, it is
possible to change settings during playback and immediately hear the results. Refer to Figure 3.1.

Figure 3.1: Multitrack Editor

The Editor panel in the Multitrack Editor displays various tools that enable mixing and editing of the sessions. The Track
Controls that appear on the left helps adjust specific settings, such as volume and pan. On the right of the Multitrack
Editor, clips and automation envelopes can be edited for each track.

3.1.2 Split, Trim, Ripple Delete, and Group Clips

Following is a detailed description of the tools available in the Multitrack Editor:
■■ Split clips

This tool is used to split audio clips and break them into separate clippings. Then, these clips can be independently
moved or edited. The tool has two more types:
¾¾ Split clips with the Razor tool
Select the toolbar, hold down the Razor tool, and choose one of the following from the pop-up menu:

●● Razor Selected Clips: It splits only those clips that are selected.
●● Razor All Clips: It splits all clips at a time when clicked.
After selecting any one of the options, click anywhere in the Editor panel where the split is required.
¾¾ Split all clips at the current-time indicator
First adjust the position of the current time indicator where the clips exist. Then select Clip and click the Split
option or press the Ctrl+K keys. Refer to Figure 3.2.

Figure 3.2: Split clips with the Razor tool and current-time indicator
■■ Trim clips:

This tool adjusts the length of the clip automatically with that of the selection by clicking the new Trim to Selection
command in the Multitrack Editor.

To do this, first select the audio in multitrack with Time Selection tool and click the Trim to Time Selection tool in
the Clip menu. Refer to Figure 3.3.
 Figure 3.3: Trim Clips
■■ Ripple Delete

This tool is useful for eliminating the range and collapsing the gap in the timeline. For this, select the Ripple Delete
command from the Edit menu, and select one of the following options:
¾¾ Selected Clips: This option swaps the remaining clips on the track and removes the selected clips.
¾¾ Time Selection in Selected Clips: This option removes the range from the selected clips.
¾¾ Time Selection in All Tracks: This option removes the range from all clips in the active session.
¾¾ Time Selection in Selected Track: This option removes the range only from the currently highlighted track in
the Editor panel.

In order to collapse gaps between clips, first right-click the empty space between the clips, then select Ripple
Delete, and click Gap.

Refer to Figure 3.4.

 Figure 3.4: Ripple Delete command
■■ Group Clips

Clips can be grouped as well as ungrouped by enabling the Group Clips option. For this, first select multiple
clips that have to be edited simultaneously and choose Clip. Then, select Groups and click Group Clips. This
option enables moving and editing of clips together as well as stretching them by enabling the Clip Stretching
option. The grouped selection can also be independently edited. For this, select Clip, choose Groups, and click
Suspend Groups. If the groups have to be re-applied throughout a session, the Suspend Groups command can
be deselected.

However, it is not mandatory to use the Group Clips option. If the clips have to be edited independently according
to one’s customization, right-click it, and select Remove Focus Clip from Groups option. Refer to Figure 3.5.
 Figure 3.5: Group Clips command

3.1.3 Automating Mixes with Envelopes

Envelopes are effects that help changing the attributes of sound over time. These are non destructive and only apply
effects to the audio and do not alter them. While using Audition, automation envelopes visually indicate settings at
specific points in time, and the same can be edited by dragging keyframes on envelope lines. These mixes can be
customized as per the requirement. Refer to Figure 3.6. Audition offers two types of automation in the Multitrack Editor—
namely, Clip Automation and Track Automation.

Figure 3.6: Automating mixes with envelopes

Automation is not supported by the Waveform Editor.

■■ Clip automation

Two default envelopes that appear in a clip include one that is used to set the volume, which are yellow lines initially
placed across the upper half of the clips and the other for pan, which are blue lines placed in the center. Both these
clips can be manipulated to create automation within a clip.
¾¾ Adjust automation with keyframes
Adobe Audition automatically calculates or interpolates all in-between values between keyframes, and it tracks
parameters using a transition method. These methods are available on right-clicking the envelope or the
envelope keyframe. They include Hold Keyframe, Spline Curves, Delete Selected Keyframes, and Select
All Keyframes.

●● Add a keyframe: For this, first locate the point on the playhead where the track parameter has to be
changed. Then, select the Add Keyframe icon from the Track Controls.

●● Navigate between track keyframes: To navigate between track keyframes, click the Editor panel, and
select a parameter from the Select menu, which appears in the bottom of the track controls. Then, choose
the Previous keyframe or Next keyframe icon.

●● Select multiple keyframes for a parameter: For selecting multiple keyframes for a single parameter,
right-click a keyframe, and click Select All Keyframes. Another alternative way is to hold down the Ctrl key
and select the specific keyframes.
¾¾ Show or hide clip envelopes
Envelopes in the clips are visible on the screen by default. However, they can be hidden for convenience and
smooth working. To make the hidden envelopes visible, from the View menu, select any one of the following:

●● Show Clip Volume Envelopes

●● Show Clip Pan Envelopes
●● Show Clip Effect Envelopes
■■ Track Automation Mode options

The following modes are made available in the Editor panel or Mixer for each track:
¾¾ Off: This option ignores track envelopes during playback and mixdown, but later they are visible so that manual
adding or adjustments can be done through keyframe.
¾¾ Read: This option enables the application of track envelopes during playback and mixdown, however any
changes made to the track are not recorded.
¾¾ Write: This option enables overwriting the current settings customized on the playback, once it is started.
¾¾ Latch: On enabling this option, the keyframes gets recorded with the first setting and continues till the playback
ends.
¾¾ Touch: This option can be used for overwriting certain sections of the automation while keeping the rest intact.
Moreover, once the adjustments in the track are stopped, the playback returns back to its previously recorded
values.
¾¾ Set audio clip properties: Multiple settings for selected clips can be edited. For this, select an audio clip and
choose Windows and click Properties. Refer to Figure 3.7. Following are the settings that are available:
 Figure 3.7: Clip properties

●● Clip Color: This option is used for changing the color. Click the swatch to select the same.
●● Clip Gain: This option helps smoothens the low and high volume clips that are difficult to be mixed.
●● Lock in Time: This option allows up and down movement to other tracks with a fixed timeline position.
●● Loop: This option enables clip looping.
●● Mute: This option is useful for inserting silence effect in the clip.
3.1.4 Using Track Controls
There exist two sections with multiple controls that primarily affect playback. One of them has a fixed set of controls,
whereas, in the other section controls can be customized to a particular function. Refer to Figure 3.8.

Figure 3.8: Controls

■■ Main track controls

The main track controls are the most commonly adjusted parameters for mixing.
¾¾ Mute and solo tracks: Tracks can be demarcated so that a particular track can be separately heard from the
rest of the mix. On the other hand, another option available is to mute the tracks to silence them. Following are
the ways to do the same:

●● To solo a track, click the Solo button from the Editor panel or Mixer. Tracks can also be automatically
removed from Solo mode, by pressing the Ctrl key and then clicking the track.

●● To mute a track, click the Mute button from the Editor panel or Mixer.
■■ Track area

The track area displays four controls that are placed as buttons on the left section of the Multitrack Editor’s toolbar.
Each of them is described as follows:
 ¾¾ Input and outputs
From the Inputs/Outputs area of the Editor panel or Mixer, following options are available:

●● From the Input menu, a hardware input can be chosen.

●● From the Output menu, a bus, a Master track, or a hardware output can be selected.
Click here to know more about Bus and Master track.
¾¾ Effects
The individual tracks have their own Effects Rack so that signal processing can be added to them. However,
as compared to the Waveform Editor Effects Rack, under Effects Rack the position of the effects can be
changed in the Multitrack Editor’s signal flow.
¾¾ Sends
Every track comprises a Sends area where buses can be created and controlled along with other parameters.
It also enables routing audio from a track to multiple buses, thus creating a tremendous signal-routing flexibility.
Each track provides up to 16 Sends, which the track output independently configures.
¾¾ EQ
One of the most important effects for multitrack production—EQ allows each track to make its own sonic space
in the audio frequency spectrum. This gives each track an option to insert a Parametric EQ effect.

Quick Test 3.1

1. ___________ tool is useful for eliminating the range and collapsing the gap in the timeline.
Trim Clips Split clips Group Clips Ripple Delete

2. Silence option is useful for inserting silence effect in the clip.

True False

3.1.5 Automatic Speech Alignment

Automatic Speech Alignment plays a very important role while processing dialogs in movies. Though, dialogs get
recorded during shootings but are often subject to noise and disturbances. At times, the actors are not close enough
to the microphone or the mic would come in the frame. This urges the need for the actors to come post shoot and dub
few parts.

Under Audition’s Automatic Speech Alignment feature, the original reference dialog, which contains noise and
disturbances, is loaded into an Audition track, and then the new dialog dubbed is recorded into a second track. Both of
them are compared using a combination of stretching and alignment, such that the new dialog matches to the reference
track.

To use Automatic Speech Alignment, in the Multitrack editor, select two clips containing the same dialog and of similar
length. Then select Clip, and choose Automatic Speech Alignment.

3.1.6 Match Clip Volume

If different clips having different volume need to be mixed, it can be matched using Match Clip Volume. Moreover, it is
safer to use Multitrack Editor as it is non destructive and the adjustments are completely reversible. Following are the
steps to do the matching:

1. Use the Move or Time Selection tool or press Ctrl+click to select multiple clips.

2. Choose Clip, and from there click Match Clip Volume.

3. From the pop-up menu, select any one of the following options:
¾¾ Loudness: It matches the sound to the average amplitude as specified by the user.
¾¾ Perceived Loudness: It matches the sound to the amplitude specified by the user keeping into account of
middle frequencies that the ears are most sensitive to.
¾¾ Peak Volume: It matches to the maximum amplitude as specified, along with normalizing the clip.
 ¾¾ Total RMS Amplitude: It matches the sound to an overall Root-Mean-Square amplitude as specified by the
user.

4. Enter a target volume.

3.1.7 Mixer Panel

The Mixer panel is an alternate way for mixing multitracks in the project. Unlike Multitrack Editor, which shows the clips
within various tracks, the Mixer does not show clips but has a corresponding Mixer channel for each Multitrack Editor
track. However, mixing can be done only after various tracks are recorded and edited. Refer to Figure 3.9.

Figure 3.9: Mixer panel

3.2 Recording Audio

3.2.1 Recording into the Waveform Editor
Audition allows recording from a microphone or any other portable device that can be plugged into the Line In port of
a sound card. However, adjustments need to be done with respect to the input signal so as to optimize signals to noise
levels. Following are the steps for recording into the Waveform Editor:

1. For this, first connect a microphone, guitar, portable music player output, mobile phone audio output, or other signal
source into a compatible audio interface input or internal audio input on the computer.

2. Then, launch Audition.

3. Choose File, select New, and click the audio file.

4. Click the Transport’s red Record button. This initiates the recording. Speak or playback whatever sound source that
 connects to the interface. It can be seen that a waveform gets generated in the Waveform Editor in real time, and
the meters reflect the current input signal level.

5. Select the Transport Pause button. Though, this pauses the recording, but the meters still show the incoming signal
level.

6. To resume, click the Pause button or the Record button again.

7. Click the Stop button to stop recording. This step finally selects the waveform, which was being recorded.

Refer to Figure 3.10.

Figure 3.10: Waveform editor

3.2.2 Recording into the Multitrack Editor

Audios can be recorded in the Multitrack Editor by overdubbing. This can be done by listening and playing recorded
tracks, thus creating sophisticated, layered compositions. Thus, each recording becomes a new audio clip on a track.
Following are the steps to record in the Multitrack Editor:

1. Choose File, select New, and click Multitrack Session.

2. An arrow to the left of the track points toward the input field. As soon as an input is selected, the track’s Arm
For Record button becomes available. After clicking the button, talk into the mic. The track’s meter indicates the
incoming level of sound.

3. Select the Transport’s Record button to start the recording.

Refer to Figure 3.11.

Figure 3.11: Recording into the Multitrack Editor

3.3 Working with Audio in Video Files

Using Multitrack editor, video files can be inserted in a session with a preview window. When a video file is inserted, the
video clip appears at the top, whereas the audio track at the bottom. The video clip can also be moved independently or
synchronized with the audio soundtrack. For synchronization, select both of them and press Ctrl+click. Refer to Figure
3.12.
 Figure 3.12: Audio usage in Video Files

Following steps can be used to import a video file:

1. In the Multitrack Editor, position the current-time indicator at the desired insertion point.

2. Choose Multitrack, click Insert File, and select a video file in a supported format.

3. When audio mixing for the video is completed, export a mixdown, and import it into any video application.

A session can contain only one video clip at a time.

3.4 Summary
In this session, Working with Multitrack Editor and Recording Audio, you learned that:
¾¾ The Multitrack Editor is used to assemble clips, add effects, change levels and panning, and create buses
for routing tracks to various effects. It is an extremely flexible and edits the soundtracks in a real‑time editing
environment.
¾¾ The Split tool is used to split audio clips and break them into separate clips. They include Split all clips at the
current-time indicator and Split clips with the Razor tool.
¾¾ The Trim tool adjusts the length of the clip automatically with that of the selection by clicking the new Trim to
Selection command in the Multitrack Editor.
¾¾ The Ripple Delete tool is useful for eliminating the range and collapsing the gap in the timeline.
¾¾ Clips can be grouped, ungrouped as well as regrouped by enabling the Group Clips option.
¾¾ Envelopes are effects that help changing the attributes of sound over time.
¾¾ While using Audition, automation envelopes visually indicate settings at specific points in time, and the same
can be edited by dragging keyframes on envelope lines.
¾¾ Audition offers two types of automation in the Multitrack Editor—namely, Clip Automation and Track
Automation.
¾¾ There exist two sections with multiple controls that primarily affect playback. One of them has a fixed set of
controls, whereas, in the other section controls can be customized to a particular function.
¾¾ Automatic Speech Alignment plays a very important role when processing dialogs in movies. If different clips
having different volume needs to be mixed, their volumes can be matched using Match Clip Volume. The Mixer
 panel is an alternate way for mixing Multitracks in the project. However, mixing can be done only after various
tracks are recorded and edited.
¾¾ Audition allows recording from a microphone or any other portable device that can be plugged into the Line
In port of a sound card and edited in a Waveform Editor. Tracks can be recorded in the Multitrack Editor by
overdubbing.
¾¾ Using Multitrack editor, video files can be inserted in a session with a preview. It can also be moved independently
or synchronized with the audio soundtrack.

3.5 Exercise
1. Clips can be grouped as well as ungrouped by enabling the Group Clips option.

True

False

2. ___________ option enables the application of track envelopes during playback and mixdown, however any
changes made to the track are not recorded.

Off

Write

Latch

Read

3. Under _____________, the two default envelopes that appear in a clip include one which set the volume and the
other is for panning.

Track Automation

Mixer panel

Clip Automation

Track Controls

4. __________ option can be used for overwriting certain sections of the automation while keeping the rest intact.

Touch

Latch

Off

Write

5. Every track comprises a __________ area where buses can be created and controlled along with other parameters.

Effects

Sends

EQ

Track controls
 Session 4
Working with Audio Effects
Learning Outcomes
In this session, you will learn to:
¾¾ Explain the different effects, such as:
●● Invert, Reverse, and Silence effects

●● Amplitude and Compression

●● Delay and Echo

●● Diagnostics

●● Filter And EQ

●● Modulation

●● Noise Reduction / Restoration

●● Reverb Effects

●● Special Effect

●● Stereo Imaginary Effects

●● Time and Pitch

Adobe Audition offers a wide range of effects which work with both Waveform and Multitrack Editors. However, a few of
them are compatible only in the Waveform Editor. Thus, in order to work with those effects that are compatible with both
of them, following three options are available to the users:
■■ Effects Rack

This is one of the most flexible ways of working with effects and allows creating a chain of up to 16 effects. These
effects can be independently worked with as well as effects can be added, deleted, replaced, or reordered.
¾¾ Apply effects in Effects Rack
Effects can be added in the Effects Rack by clicking the insert area. Then click the right arrow and choose
an effect from the drop-down menu. The Analog Delay effect can be turned off by clicking the Power button.
Playback can be started by pressing the Spacebar. Refer to Figure 4.1.

Figure 4.1: Apply effects in Effects Rack

 ¾¾ Removing, editing, replacing, and moving an effect
To remove a single effect from a file, click the effect’s insert area and press the Delete key, or alternatively click
the insert’s right arrow and select Remove Effect from the drop-down menu. However, if all effects have to be
removed from the Effect Rack, right-click or Ctrl+click anywhere on an effect’s insert, and then select Remove
All Effects.
To edit an effect, in case the effects window is hidden or closed, double-click the effect’s insert and then click
the insert’s right arrow. Then, select Edit Effect from the drop-down menu. Alternatively, right-click or Ctrl+click
anywhere in an effect’s insert and select Edit Selected Effect.
An effect can also be replaced with another one by clicking the insert’s right arrow and selecting a different effect
from the drop-down menu.
To move an effect to a different insert, click the effect’s insert and drag it to the desired destination insert.
■■ Effects menu

Although this option works faster when compared with Effects Rack, it however, applies only one specific effect to
an audio at a time.
■■ Favorites menu

Whenever a setting is required to be applied multiple times in a file, it can be saved as a preset. Such a preset
appears in the Favorites menu and selecting it will apply the present settings instantly to any audio that is selected.
However, this way is also one of the least flexible as any parameters cannot be changed before applying the effect.

4.1 Invert, Reverse, and Silence Effects

The Invert effect alters the signal’s polarity and produces no audible difference. Reverse flips the flow of the audio such
that the beginning is placed at the end and the end at the beginning. The Silence effect replaces the selected audio with
silence. However, by pressing the Undo command it is possible to undo the effects.

4.2 Amplitude and Compression

The Amplitude and Compression option increases the effect or soothes an audio signal. It can be combined with the
other effects in the Effect Rack as it operates in real time. This option can be accessed from the Effects menu.

4.2.1 Channel Mixer

The Channel Mixer changes the amount of left and right signal present in the left and right channels. Consider two
possible cases: converting stereo to mono and reversing the left and right channels. Refer to Figure 4.2 for Channel
Mixer and Figure 4.3 for Channel Mix.

Figure 4.2: Channel Mixer

 Figure 4.3: Channel Mix
¾¾ Input channel sliders
This option enables determining the percentage of the current channels for mixing with the output channel. For
instance, if the L value is 50 and R value is also 50, it will result in an output having equal audio of left and right
channels.
¾¾ Invert
This option reverses a channel’s phase. If all channels are inverted, it makes no actual difference. However, by
merely inverting a single channel can bring a drastic change in the sound.

4.2.2 DeEsser
This DeEsser option reduces the vocal sibilants, which means the ‘ess’ sounds. De-essing a file includes three steps
as follows:

1. Identify the frequencies in the track where such sounds are present.

2. Define the range by selecting it.

3. Set a Threshold. Thus, when the sound is exceeded by a sibilant, the gain is automatically controlled within the
specified range. This makes the ‘ess’ sound less prominent. Refer to Figure 4.4.
 Figure 4.4: DeEsser function

As shown in Figure 4.4, the graph reveals the processed frequencies. To see the amount of audio content in the
processes range, click the Preview button. Following are the sub-options available in the DeEsser function:
¾¾ Mode: This option compresses the frequencies. To compress evenly across the entire track, select the
Broadband option from the Mode. Alternatively, Multiband option will compress only the sibilance range by
slightly increasing the processing time.
¾¾ Threshold: This option lays the amplitude, above which the compression takes place.
¾¾ Center Frequency: This option identifies the frequency where the sibilance is most intense. To verify the range
identified, adjust this setting while audio playback.
¾¾ Bandwidth: This option ascertains the frequency range that triggers the compressor.
¾¾ Output Sibilance Only: This option plays the detected sibilance. Start playback, and adjust other settings in
the menu.
¾¾ Gain Reduction: This option indicates the compression level of the processed frequencies.
4.2.3 Dynamics Processing
The Amplitude and Compression option displays a list of effects of which Dynamics Processing effect can be used
as a compressor, limiter, or expander. This is because, as a compressor and limiter, Dynamics Processing produces
consistent volume levels and reduces the dynamic range. Whereas, by increasing the dynamic range having low-level
signals, it acts as an expander.

Apart from this, Dynamics Processing effect brings out minute changes that can be recognized only after constantly
hearing the track. However, if the track is being worked with Waveform Editor, it is advisable to save a copy of the original
file for future need. Refer to Figure 4.5.
 Figure 4.5: Dynamics Processing effect
■■ Dynamics tab

The following sub-options are available in the Dynamics tab as shown in Table 4.1:
Options Description
Graph Input level is depicted along the horizontal X axis of the graph, whereas the vertical Y axis depicts
the new output level. For instance, an audible sonic sound occurring around 20 decibels (dB) can
be boosted by altering the input signal at that level, but other things remain unchanged. To boost
the quiet sounds and suppress the loud ones, an inverse line from the upper left to the lower right
can be drawn.
Add point This option precisely adds control points in the graph by manually specifying the numerical input
and output levels rather than clicking the graph to add them.
Delete point This option deletes the selected point from the graph.
Invert This option flips the graph, thus converting compression into expansion or expansion into
compression.
Reset This option resets the graph to its default condition.
Spline Curves This option creates smooth, curved transitions between control points, instead of abrupt, linear
transitions.
Make-Up Gain This option enhances the processed signal.
Table 4.1: Dynamic tab options

The graph can be inverted only if the two default corners have points, such as 100, 100 and 0, 0. In addition to
this, it is necessary for the output level to increase from left to right, which means that each control point must
ascend higher towards the right.

■■ Settings tab

Refer to Table 4.2, which gives the options are available in the Settings tab as follows:
 Options Description
General This option manages the overall settings.
Look-Ahead Time This option manages the temporary sharp spikes that occur at the onset of extremely loud
signals, which extend beyond the compressor’s Attack Time settings. When the Look-Ahead
Time is extended, the compression occurs before the audio gets louder, limiting the amplitude
at a certain level. Whereas, by reducing Look-Ahead Time may prove desirable if the impact of
percussive music has to be enhanced.
Noise Gating This option entirely silences the signals that are expanded below the 50:1 ratio.
Level Detector This option figures out the original input amplitude.
Input Gain This option adds gain to the signal before it enters the Level Detector.
RMS mode This option determines sound levels based on the root-mean-square formula. RMS is an
averaging method that approximately matches the way people perceive volume. It precisely
reflects amplitudes in the Dynamics graph. For example, a limiter at 10 dB reflects an average
RMS amplitude of 10 dB.
Peak mode This option measures levels based on the peaks of the amplitude. This mode works only when
there are temporary spikes because of loud noise. Moreover, it is a difficult method as compared
with the RMS mode as the peaks are not precisely reflected in the Dynamics graph.
Gain Processor This option will enhance or soothes the signal depending on the amplitude detected.
Output Gain This option adds gain to the output signal after all dynamics processing is done.
Attack Time This option defines time taken by the output’s signal to reach the specified level. For example,
if audio suddenly drops 30 dB, then the specified attack time passes before the output level
changes.
Release Time This option measures the level of the milliseconds at which the current output level is maintained.
Link Channels This option facilitates processing of all channels evenly, by preserving the stereo or surround
balance. Therefore, if a drum beat is compressed on the left channel, its right channel will also
reduce by an equal amount.
Band Limiting It restricts dynamics processing to a specific frequency range.
Low Cutoff It is the lowest frequency that dynamics processing affects.
High Cutoff It is the highest frequency that dynamics processing affects.
Table 4.2: Settings tab options

4.2.4 Hard Limiter Effect

The Hard Limiter effect is mostly used to suppress audio levels so that they do not rise above a specified Threshold.
Mostly this effect is applied with an input boost, such that the overall volume increases avoiding any kind of distortion.
Following are the sub-options that are available in this menu:
¾¾ Maximum Amplitude: This option sets the maximum sample amplitude allowed.
¾¾ Input Boost: This option makes a selection louder without clipping it. However, when increased, it also increases
the compression. To create loud, high-impact pop audio, extreme settings can be applied in this option.
¾¾ Look-Ahead Time: It defines the time limit, in milliseconds, required to soothe the audio before hitting the
loudest point.

The minimum allowable value is at least 5 milliseconds as lower than this will result in audible distortion in
audio.

¾¾ Release Time: This option defines the time required for an audio to resume its normal sound levels in case
it experiences an extremely loud sound. Measured in milliseconds, a setting of around 100 works well and
preserves very low bass frequencies.

If this value is too large, audio may sound quiet and not resume normal levels for some time.
 ¾¾ Link Channels: This option connects the loudness levels of all channels together, thereby preserving the stereo
or surround balance.

4.2.5 Multi Band Compressor Effect

As each frequency contains dynamic content, Multiband Compressor effect enables compressing these bands
independently, thus making it a powerful tool for audio mastering. It contains controls that precisely determine crossover
frequencies and applies band specific compression settings. Refer to Table 4.3.

Options Description
Crossover This option defines crossover frequencies that determine the width of each band. The Low,
Midrange, and High frequencies can be entered manually, or the crossover markers can
be dragged above the graph.
Solo buttons This option enables hearing of specific frequency bands. By clicking one Solo button, only
one band can be heard in isolation, or multiple buttons can be selected so as to hear two
or more bands together.
Bypass buttons This option makes individual bands pass through without processing.
Threshold sliders This option manages the input level, where the compression begins. Generally, the values
range from 60 to 0 dB, depending upon the audio content and musical style. If extreme
peaks and dynamic range has to be retained, Thresholds should be 5 dB below the peak
input level. Whereas, if the audio has to be compressed greatly and the dynamic range has
to be reduced, settings around 15 dB below the peak input level can be done.
Input Level meters This option manages input amplitude. To reset peak and clip indicators, double-click the
meters.
Gain Reduction meters This option calculates the amplitude reduction level with red meters that extend from the
top till the bottom.
Gain This option enhances or attenuates the amplitude after compression. The range available
is from 18 to +18 dB, where 0 is unity gain.
Ratio This option sets a compression ratio between 1:1 and 30:1. Generally, settings range from
2.0 to 5.0, and higher settings produce an extremely compressed sound.
Attack This option determines the time by which compression is to be applied when the audio
exceeds the Threshold limit. Generally, the values range from 0 to 500 milliseconds. The
default value of 10 milliseconds works with most of the audio. However, faster settings may
work better for audio with fast changes, but they sound unnatural for less percussive audio.
Release This option defines by when compression should stop after audio drops below the
Threshold. Generally, the values range from 0 to 5000 milliseconds. The default value of
10 milliseconds works with most of the audio. However, faster settings for faster audio and
slower settings for less percussive audio is advisable.
Output Gain This option enhances or attenuates the overall output level after compression. Generally,
values range from 18 to +18 dB, where 0 is unity gain. To reset peak and clip indicators,
double-click the meters.
Limiter This option applies limiting post Output Gain at the end of the signal path. The Threshold,
Attack, and Release settings can be specified which should be less aggressive than
similar band specific settings. Then a Margin setting can be specified to determine the
absolute ceiling relative to 0 Decibels relative to Full Scale (dBFS).
Spectrum On Input This option displays the frequency spectrum of the input signal, instead of the output signal,
in the multiband graph. To quickly review the amount of compression applied, toggle this
option on and off.
Brickwall Limiter This option applies immediate, hard limits at the current Margin setting or deselect this
option to apply slower soft limiting. Setting a limit gives a peak sound only at the level set,
whereas the rest of the audio remains unaffected.
Link Band Controls This option adjusts the compression settings for all bands, along with preserving the relative
differences between the bands.
Table 4.3: Multi Band compressor effect options
4.2.6 Normalize Effect
The Normalize effect determines a peak level for a file or a selection. For instance, if an audio is normalized to 100%,
the maximum amplitude of 0 dBFS is achieved. However, normalizing it between –3 and –6 dBFS provides a cushion
for further processing. Following are the sub-options available:
¾¾ Normalize: This option sets the highest peak of the audio relative to the maximum amplitude.
¾¾ Normalize All Channels Equally: The amplification amount is calculated on the basis of all channels of a
stereo or surround waveform. Deselecting this option will result the amount to be calculated separately for each
channel, by amplifying one channel’s value considerably more than others.
¾¾ DC Bias Adjust: This option manages the position of the waveform in the wave display. To bring the waveform
in the center, set the percentage to zero. To bring the entire selected waveform above or below the center line,
specify a positive or negative percentage.

The Normalize effect enhances the entire file or selected section uniformly. Therefore, if the original audio
reaches a loud peak of 80% and a quiet low of 20%, normalizing the audio to 100% enhances the loud peak to
100% and the quiet low to 40%.

4.2.7 Single Compressor Effect

The Single-band Compressor effect decreases the dynamic range, thereby providing consistent volume levels and
increased loudness. This type of compression is particularly effective for voice-overs, as it requires the speaker’s volume
to stand out against the background audio. Following are the sub-options available:
¾¾ Threshold: This option manages the input level, where the compression begins. Though, the best setting
depends on audio content and style, if extreme peaks and dynamic range has to be retained, Thresholds should
be 5 dB below the peak input level. Whereas, if the audio has to be compressed greatly and the dynamic range
has to be reduced, settings around 15 dB below the peak input level can be done.
¾¾ Ratio: This option sets a compression ratio between 1:1 and 30:1. Generally, settings range from 2.0 to 5.0, and
higher settings produce an extremely compressed sound.
¾¾ Attack: This option determines the time by which compression is to be applied when the audio exceeds the
Threshold limit. The default value of 10 milliseconds works with most of the audio materials. However, faster
settings may work better for audio with fast changes, such as percussion recordings.
¾¾ Release: This option defines by when compression should stop after the audio drops below the Threshold. The
default value of 100 milliseconds works with most of the audio. However, faster settings for faster audio and
slower settings for less percussive audio is advisable.
¾¾ Output Gain: This option enhances or attenuates the overall output level after compression. Generally, values
range from 30 dB to +30 dB, where 0 is unity gain.

4.2.8 Tube Compressor Effect

The Tube-modeled Compressor recreates the effect as created by the vintage hardware compressors. It adds subtle
distortion that makes the audio pleasant. Following are the sub-options available:
¾¾ Threshold slider: This option manages the input level, where the compression begins. Generally, the values
range from 60 to 0 dB, depending upon the audio content and musical style. If extreme peaks and dynamic
range has to be retained, Thresholds should be 5 dB below the peak input level. Whereas, if the audio has to
be compressed greatly and the dynamic range has to be reduced, settings around 15 dB below the peak input
level can be done.
¾¾ Input Level meters: Appearing to the left of the slider, these meters calculates the input amplitude. To reset
peak and clip indicators, double-click the meters.
¾¾ Gain Reduction meters: Appearing to the right of the slider, these meters measure the level of amplitude
reduction with red bars that extend from the top till the bottom.
¾¾ Gain: This option enhances or attenuates the amplitude after compression. The range available is from 18 to
+18 dB, where 0 is unity gain.
¾¾ Ratio: This option sets a compression ratio between 1:1 and 30:1. Generally, settings range from 2.0 to 5.0, and
higher settings produce an extremely compressed sound.
¾¾ Attack: This option determines the time by which compression is to be applied when the audio exceeds the
 Threshold limit. Generally, the values range from 0 to 500 milliseconds. The default value of 10 milliseconds
works with most of the audio. However, faster settings may work better for audio with fast changes, but they
sound unnatural for less percussive audio.

4.2.9 Volume Envelope Effect

The Volume Envelope effect enables changing the audio by adding and deleting boosts and fades over time. For this,
click the Editor menu and drag the yellow line, wherein the top portion represents 100% amplification and the bottom
represents 100% attenuation. Refer to Figure 4.6.

Figure 4.6: Volume Envelope effect

As shown in Figure 4.6, the yellow line is used to drag and adjust amplitude percentages, and click to add keyframes
for adding boosts and fades. Whereas, Spline curves can be used to make smooth and curved transitions between
keyframes, rather than linear ones.

Quick Test 4.1

1. Whenever a setting is required to be applied multiple times in a file, it can be saved as a preset in the ____________.
Effects Rack Effects menu File menu Favorites menu

2. This DeEsser option reduces the vocal sibilants, which means the ‘ess’ sounds.
True False

3. The ____________ recreates the effect as created by the vintage hardware compressors and adds subtle
distortion to the audio.
Hard Limiter effect Multi Band compressor effect
Normalize effect Tube-modeled Compressor

4.3 Delay and Echo

When an original signal is copied as separate files and reoccur every millisecond, such an effect is called a Delay.
Whereas, in case of Echoes, sounds are delayed such that an exact copy of the original sound is heard. Thus, both
delays and echoes are a great way to add ambience to the sound. Following are the various effects that are available in
the Delay and Echo explained in detail.

4.3.1 Analog Delay

The Analog Delay effect generates the warm sound waves of vintage hardware delay units. Its options apply characteristic
distortion and adjust the stereo coverage. To simulate distinct echoes, specify delay time of 35 milliseconds or more,
whereas shorter time can be specified to create subtle effects. Following are the sub-options available:
 ¾¾ Mode: This option defines the type of hardware emulation, equalization levels, and sound distortion
characteristics. The Tape and Tube effects specify the character of vintage delay, whereas Analog reflects
electronic delay lines.
¾¾ Dry Out: This option defines the original, unprocessed audio level.
¾¾ Wet Out: This option defines the delayed, processed audio level.
¾¾ Delay: This option determines the delay length measured in milliseconds.
¾¾ Feedback: This option repeats echoes by resending the delayed audio through the delay line. For instance, if
the delayed audio is set at 20%, then the audio is delayed at one-fifth of its original volume, which results in the
fading away of audio gently. Whereas, if the same is set to 200%, delayed audio is double its original volume,
Which makes the echo quickly come up with intensity.
¾¾ Trash: This option surges distortion levels and enhances low frequencies, thus warmifying the audio.
¾¾ Spread: This option defines the stereo width of the delayed signal.
4.3.2 Delay Effect
The Delay effect can create single echoes as well as other effects. As specified earlier, a delay of 35 milliseconds or
more create discrete echoes, while delays between 15 to 34 milliseconds can create a simple chorus or flanging effect.
However, if the delay is between 1 to 14 milliseconds, it resembles a mono sound coming either from the left or the right
side, while the actual volume levels for left and right are the same. Following are the sub-options available:
¾¾ Delay Time: This option is used for creating delays for both the left and right channels, ranging from 500
milliseconds to +500 milliseconds. Thus, if a negative value is entered, the delayed portion is heard before the
original part.
¾¾ Mix: This option manages the ratio of processed-Wet signal to the original-Dry signal in order to be mixed into
the final output. A value of 50 mixes the two proportionately.
¾¾ Invert: This option flips the phase of the delayed signal, creating phase-cancellation effects similar to comb
filters.

4.3.3 Echo Effect

The Echo effect can be used to create varying echoes, such as Grand Canyon type “Hello ello llo lo o” to metallic
ones, or create drain pipe sounds by changing the delay amount. This effect adds a sequence of repetitive, decaying
echoes to a sound which brings a change in its ambience. Echoes can be used to alter a room’s sound from reflective
to absorptive. Following are the various sub-options that are available:
¾¾ Delay Time: This option defines the gap between each echo which can be mentioned in terms of milliseconds,
beats, or samples. Thus, if the delay time is 100 milliseconds, then it results a delay of 1/10th second between
successive echoes.
¾¾ Feedback: This option sets the fall-off ratio of an echo, wherein the successive echo tails off at a higher edge
than the previous one. Thus, a decay setting of 0% results in no echoes, whereas at 100% an echo is generated
which never gets quieter.
¾¾ Echo Level: This option defines the percentage of echoed signal to mix with the original signal in the final
output.
¾¾ Lock Left & Right: This option links the sliders for Decay, Delay, and Initial Echo Volume options, thereby
establishing similar settings for each channel.
¾¾ Echo Bounce: This option enables the bouncing back of echoes back and forth between the left and right
channels. For this, set the initial echo volume to 100% for one channel and 0% for the other. If settings are set
for both, then two sets of echo will be created that will bounce from one channel to the other.
¾¾ Successive Echo Equalization: This option makes each subsequent echo to pass through an eight-band
equalizer, to create a natural sound absorption. Thus, a setting of 0 does not change the band, while a setting
of 15 decreases that frequency by 15 dB. As there is a gap of -15 dB between each echo, some frequencies
may finish faster than others.
¾¾ Delay Time Units: This option refers to the units used, such as milliseconds, beats, or samples for the Delay
Time.
4.4 Diagnostics
This tool is useful for smoothening the audio by removing clicks, distortion, or silence as well as adds markers where
silence exists. The sub-options available in this effect are as follows:
■■ DeClicker effect

The DeClicker effect identifies and removes clicks and pops from wireless microphones, vinyl records, and other
audio sources. The option available in the DeClicker resemble with the Automatic Click Remover, which can be
combined with the other effects in the Effects Rack and apply in the Multitrack Editor. To apply multiple scan and
repairs automatically, the effect has to be manually applied. However, before applying the DeClicker effect, it can
be evaluated and then selected accordingly. The following options can be accessed by clicking the Diagnostics
panel and selecting Settings:
¾¾ Threshold: This option is used for determining noise sensitivity. The settings range from 1 to 100, the default
being 30. If the settings are kept at lower values, it may detect more clicks and pops along with including audio
which can be retained.
¾¾ Complexity: This option determines noise complexity. The settings range from 1 to 100, the default being 16.
However, if settings are kept at higher values, it may process the file more but can also degrade audio quality.
■■ DeClipper options

The DeClipper effect is used for repairing clipped waveforms in audio by filling it with new audio data. Clipping
occurs when the amplitude of an audio is more than the maximum current bit depth. It also happens when the audio
is recorded at high levels. Clipping can be detected when the boxes at the far extreme right turns red and can be
monitored by watching the Level Meters. Clipped areas visually look like broad flat areas at the waveform’s top and
resembles a static distortion sonically. The following sub-options can be accessed by clicking the Diagnostics panel
and selecting Settings:
¾¾ Gain: This option defines the amount of attenuation that should occur before processing. Click the Auto option
to specify the gain setting on average input amplitude.
¾¾ Tolerance: This option defines the amplitude variation in clipped regions. For instance, a value of 0% will detect
clipping only in horizontal lines at maximum amplitude, whereas 1% detects clipping beginning at 1% below
maximum amplitude.
¾¾ Minimum Clip Size: This option defines the length of the shortest run of clipped samples which has to be
repaired. Lower the values, higher would be the percentage of clipped samples, and higher values will repair
clipped samples only if they are lipped samples appear before and after them.
¾¾ Interpolation: This option uses spline curves to reproduce the frequency content of audio clipped. Though it
is faster, it can introduce false new frequencies. The Fast Fourier Transforms (FFT) are used in FFT option to
reproduce clipped audio.
■■ Delete Silence and Mark Audio options

The Delete Silence and Mark Audio options define silence in the audio, and they either remove or mark them. If
silence is automatically marked, it tightens the tracks without changing the foreground audio, while marking silence
automatically helps navigate to audio cues to enable editing. The following sub-options can be accessed by clicking
the Diagnostics panel and selecting Settings:
¾¾ Define Silence As: This option defines the amplitude and duration identified as silence.
¾¾ Define Audio As: This option defines the amplitude and duration identified as audio content.
¾¾ Find Levels: This option automatically calculates the silence and audio signal levels based on file content.
¾¾ Fix By: The Choose Shortening Silence option will reduce silent passages to amount as specified. The
Choose Deleting Silence option mutes the silent passages but retains the original file length.

4.5 Filter and EQ

4.5.1 FFT Filter
FFT is an algorithm that quickly analyzes frequency and amplitude. This effect draws curves or notches easily, rejecting
or enhancing specific frequencies. Refer to Figure 4.7.
 Figure 4.7: FFT Filter

Following are various sub-options available:

¾¾ Scale: This option manages the arrangement of frequencies along the horizontal x axis.
●● To control low frequencies, select Logarithmic option. A logarithmic scale resembles closely as to how
people hear sounds.

●● Select Linear option, for a detailed, high frequency work with evenly spaced intervals in frequency.
¾¾ Spline Curves: This option produces smoother, curved transitions between control points, rather than more
abrupt, linear transitions.
¾¾ Advanced options: This option can be accessed by clicking the triangle:
●● FFT Size: This option defines the size of the FFT, which later specifies the tradeoff between frequency and
accuracy. Higher values can be prescribed for sharp, precise frequency filters, whereas lower values for
reduced transient artifacts in percussive audio. Ideal values range from 1024 and 8192.

●● Window: This option defines the FFT shape, and each option results giving a separate frequency response
curve.

4.5.2 Graphic Equalizer Effect

This effect enhances or attenuates specific frequency bands and provides a graphic representation of the output EQ
curve. Unlike the Parametric Equalizer, the Graphic Equalizer uses preset frequency bands to enable fast equalization.
Refer to Figure 4.8.
 Figure 4.8: Graphic Equalizer

Following are the intervals at which the user can space frequency bands:
¾¾ One octave (10 bands)
¾¾ One half octave (20 bands)
¾¾ One third octave (30 bands)
Graphic equalizers having fewer bands provide quicker adjustment, whereas more bands provide greater precision.
¾¾ Gain sliders: This option specifies the exact boost or attenuation value for the chosen band.
¾¾ Range: This option specifies the range of the slider controls with the value ranging between 1.5 and 120 dB.
¾¾ Accuracy: This option manages the accuracy level for equalization. Although, higher accuracy levels require
more processing time, they give a better frequency feedback in the lower ranges. However, to equalize only
higher frequencies, lower accuracy levels can be used.
¾¾ Master Gain: This option compensates for an overall volume level that is too soft or too loud post the EQ
settings are done. No master gain adjustment is measured at 0, which is the default value.

4.5.3 Notch Filter Effect

The Notch Filter effect removes to a maximum of six user defined frequency bands, which includes even narrow
frequency bands, such as a 60 Hz hum, leaving the surrounding frequencies unaffected. Refer to Figure 4.9.
 Figure 4.9: Notch Filter effect
¾¾ Frequency: This option defines the center frequency for each notch.
¾¾ Gain: This option determines the amplitude for each notch.
¾¾ Notch width: This option specifies a frequency range for all notches. The three options range from Narrow to
Very Narrow to Super Narrow. It is advisable to use the Narrow setting for a second order filter that removed
adjacent frequencies and Super Narrow for a sixth order filter.

Greater attenuation removes a wide range of neighboring frequencies. It is advisable to limit 30 dB of attenuation
for a Narrow setting, 60 dB for Very Narrow, and 90 dB for Super Narrow.

¾¾ Ultra-Quiet: This option works only with high-end headphones and monitoring systems for eliminating noise
and artifacts, although requiring too much processing.

4.5.4 Parametric Equalizer Effect

This effect provides maximum control over tonal equalization. Unlike the Graphic Equalizer, which gives a fixed number
of frequencies and Q bandwidths, the Parametric Equalizer gives complete control over frequency, Q bandwidths,
and gain settings. Thus, reducing small range of frequencies and simultaneously boosting a broad low frequency with
insertion of a notch filter can be done.

A second order Infinite Impulse Response (IIR) filters are used by Parametric Equalizer, which are very quick and
provide accurate frequency resolution, whereas Graphic Equalizer provide slightly improved phase accuracy.

Click here to know more about the settings.

4.6 Modulation
4.6.1 Chorus Effect
Using a direct simulation method, Chorus effect can be used to enhance a vocal track or add stereo effect to mono audio,
finally resulting in a lush, rich sound. This effect generates an audio containing voices or instruments simultaneously
being played by adding short delays with a small amount of feedback.

Chorus effect makes each voice sound distinct from the original by slightly changing the timing, intonation, and vibrato.
The Feedback options enable adding of details to the result. Refer to Figure 4.10.
 Figure 4.10: Chorus Flanger
Click here to know more about the characteristics of each voice in the chorus.

4.6.2 Flanger Effect

This effect is created by adding approximate short delays to the original signal. Refer to Table 4.4 for its options:

Options Description
Initial Delay Time This option sets the time from which the flanging should start behind the original signal.
This effect occurs by moving over time from a setting of initial delay to a second delay.
Final Delay Time This option sets the time at which flanging ends behind the original signal.
Stereo Phasing This option defines the left and right delay time separately in terms of degrees. For instance,
at 180 degrees, the initial delay of the right channel is set to occur at the same time as the
left channel’s final delay. This option can be used to reverse the initial/final delay settings
for the left and right channels, thereby creating a circular, psychedelic effect.
Feedback This option defines the flanged signal’s percentage that is fed back into the flanger. In
the absence of feedback, only the original signal is used. If present, the effect uses a
percentage of the affected signal before the current point of playback.
Modulation Rate This option defines the speed used by the delay to move from the initial to final delay times,
in terms of cycles per second (Hz) or beats per minute (beats). Smaller the setting, wider
would be the varying effects.
Mode This option enables three ways of flanging:

Inverted: This option flips the delayed signal, thus periodically cancelling the audio instead
of reinforcing the signal. If the Original Expanded mix settings are kept at 50/50, the
waves are cancelled to silence whenever the delay is at zero.

Special Effects: This option combines the normal and inverted flanging effects, wherein
the delayed signal is added and the leading signal is subtracted.

Sinusoidal: This option makes the transition from initial delay to final delay and is followed
by a sine curve. In the other case, the transition becomes linear, and the delays are at a
constant rate. However, if Sinusoidal is selected, the signal is at the extremes than between
the delays.
Mix This option manages the mix of original and flanged signal. Both signals are required to
achieve the characteristic of cancellation and reinforcement occurring during flanging. At
100%, no flanging occurs at all, whereas when delayed at 100%, the output comes as a
wavering sound, like a bad tape player.
Table 4.4: Flanger effect options
4.6.3 Phaser Effect
Phasing shifts an audio signal’s phase and recombines it with the original, thereby creating psychedelic effects dated
back in 1960s. However, unlike the Flanger effect, which uses variable delays, the Phaser effect clears a sequence of
phase-shifting filters towards and from an upper frequency. It greatly alters the stereo image, thereby creating unearthly
sounds. Following are the sub-options available:
¾¾ Stages: This option specifies the number of phase-shifting filters. Higher the setting, denser would be the
phasing effects.
¾¾ Intensity: This option quantifies phase shifting, which is applied to the signal.
¾¾ Depth: This option defines the level at which the filters travel below the upper frequency.
¾¾ Mod Rate: This option controls the speed at which the filters travel to and from the upper frequency, which has
to be specified in Hz (cycles per second).
¾¾ Phase Diff: This option defines the difference in phases between stereo channels. Positive values start the
shifts in the left channel, whereas negative values in the right. The maximum values of +180 and -180 degrees
creates a complete different and sonically identical audio.
¾¾ Upper Freq: This option manages the upper-most frequency from which the filters sweep. To produce a dramatic
effect, select a frequency near the middle of the range.
¾¾ Feedback: This option enables inserting a percentage of the phaser output back to the input, thereby
strengthening the effect. Negative values overturn phase before feeding audio back.
¾¾ Mix: This option manages the ratio of original to processed audio.
¾¾ Output Gain: This option adjusts the output level after processing as follows:
●● By implementing effects in the Waveform Editor

●● By implementing effects in the Multitrack Editor

●● Use effect presets

Quick Test 4.2

1. The Delay effect adds a sequence of repetitive, decaying echoes to a sound which brings a change in its ambience.
True False

2. The __________ effect identifies and removes clicks and pops from wireless microphones, vinyl records, and
other audio sources.
DeClicker DeClipper Delay Echo

3. The Hard Limiter effect is mostly used to suppress audio levels so that they do not rise above a specified threshold.
True False

4.7 Noise Reduction/Restoration

4.7.1 Techniques for Restoring Audio
Audio problems can be solved by Audition using two important features. They include Spectral Display and Diagnostic
or Noise Reduction effects. Refer to Figure 4.11. Spectral Display is used for identifying and selecting noise or
individual artifacts ranges. Whereas, Diagnostic or Noise Reduction effects are used for fixing the following:
¾¾ The splitting sound arising from wireless microphones or old vinyl records.
¾¾ Extra background noises like wind rumble, tape hiss, or power-line hum.
¾¾ Phase removal resulting because of poorly placed stereo microphones or misaligned tape machines.
 Figure 4.11: Noise Reduction

4.7.2 Apply the Noise Reduction Effect

1. From the Waveform Editor, select the range containing only noise and should be at least half a second long.

2. From the Effects menu, select Noise Reduction/Restoration, and click Capture Noise Print.

3. In the Editor panel, select the range from which the noise has to be removed.

4. From the Effects menu, select Noise Reduction/Restoration, and click Noise Reduction.

5. Adjust the desired options.

■■ Noise Reduction options
¾¾ Capture Noise Print: This option is used for extracting the noise from a selected range, which includes only the
background noise. However, additional information about the background noise is also procured as that it can
be removed from the remaining waveform.
¾¾ Save the Current Noise Print: This option is used for saving the noise print as an .fft file, which contains details
about the sample type, FFT size and its three FFT coefficients, such as the lowest, highest, and the power
average.
¾¾ Load a Noise Print from Disk: This option opens any noise print previously saved in the FFT format and can
be applied only to identical sample types.
¾¾ Graph: This option represents the frequency along the horizontal x axis and the amount of noise reduction
along the vertical y axis. Whereas, the blue control curve manages the noise reduction in varying ranges.
¾¾ Noise Floor: Higher values indicate high amplitude of detected noise at each frequency, whereas low values
indicate the lowest amplitude. Threshold shows the amplitude below which noise reduction occurs.
¾¾ Scale: This option 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 sounds.

●● For detailed, high frequency work with evenly spaced intervals in frequency, select Linear.
¾¾ Channel: This option displays the selected channel in the graph, where the noise reduction is the same for all
channels.
 ¾¾ Select Entire File: This option applies a captured noise print to an entire file.
¾¾ Noise Reduction: This option controls the amount of noise reduction in the output signal. While previewing the
audio, these settings can be adjusted so as to achieve maximum noise reduction with minimum artifacts.
¾¾ Reduce By: This option determines the amplitude reduction of detected noise. Values between 6 and 30 dB is
desirable and lower values are useful for reducing bubbly artifacts.
¾¾ Output Noise Only: This option previews only noise for checking whether the effect usefully removes the
desirable audio.

4.7.3 Adaptive Noise Reduction Effect

The Adaptive Noise Reduction effect quickly deletes noise, such as background sounds, rumble, and wind. It can
also be combined with other effects in the Effects Rack and apply it in the Multitrack Editor as it operates in real time.
However, the standard Noise Reduction effect can also be obtained as an offline process in the Waveform Editor, which
is sometimes more effective in removing continuous noise, such as hiss or hum.

4.7.4 Automatic Click Remover Effect

To quickly remove crackle and static from vinyl recordings, the Automatic Click Remover effect can be applied. It
can be applied on a large or a small pop. This effect provides similar options as that of DeClicker effect, and can be
combined 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, it has to be manually applied it multiple times.
¾¾ Threshold: This option is used for determining noise sensitivity. The settings range from 1 to 100, the default
being 30. If the settings are kept at lower values, it may detect more clicks and pops along with including audio
which can be retained.
¾¾ Complexity: This option determines noise complexity. The settings range from 1 to 100, the default being 16.
However, if settings are kept at higher values, it may process the file more but can also degrade audio quality.

4.7.5 Automatic Phase Correction Effect

The Automatic Phase Correction effect addresses azimuth errors which arise out of misaligned tape heads, stereo
spreading from incorrect microphone placement, and many other phase-related problems. Following are the sub-options
available:
¾¾ Global Time Shift: This option is used for activating the left and right channel shift sliders, so as to apply an
even phase shift to the selected audios.
¾¾ Auto Align Channels and Auto Center Panning: This option is useful for aligning the phase and panning for
a series of distinct time gaps, which can be specified using the following options:
¾¾ Time Resolution: This option specifies the number of milliseconds in each processed interval. Smaller values
increase accuracy; larger ones increase performance.
¾¾ Responsiveness: This option defines overall processing speed. Slower the settings, higher the accuracy, and
faster the settings, higher the performance.
¾¾ Channel: This option defines the channel’s phase where the correction has to be applied.
¾¾ Analysis Size: This option defines the number of samples appearing in each analyzed audio unit.
4.7.6 DeHummer Effect
One of the most common application, DeHummer effect deletes narrow frequency bands and their harmonics as well
as resolves the power line hum from lighting and electronics. However, DeHummer can also apply a notch filter that
deletes an overly echoing frequency from the audio.

4.7.7 Hiss Reduction

This effect is used for reducing the hiss sound from the audios, which is consistent throughout the audio and is sourced
from cassettes, vinyl records, or microphone Preamps. It also lowers the amplitude of a frequency range if it goes below
the Threshold, called the noise floor, whereas those ones above the Threshold remain unaffected.
4.8 Reverb Effects
Assume a room wherein an audio is being played. Here, the sound hits and bounces off the walls, ceiling, and floor
and then reaches the ears. These sounds reach with a good speed such that they are heard as a single sound and not
as separate echoes. Thus, a sonic ambience produces an impression of space. This reflected sound is referred to as
reverberation or reverb. Refer to Figure 4.12.

Figure 4.12: Reverb

4.8.1 Convolution Effect

This kind of effect can be observed from closets to concert halls. Convolution-based reverbs use impulse files to generate
audio spaces. Following are the various sub-options available:
¾¾ Impulse: This option identifies the file that generates an audio space. Select Load to add a custom impulse file
in WAV or AIFF format.
¾¾ Mix: This option manages the ratio of original to reverberant sound.
¾¾ Room Size: This option defines a percentage of the full room specified by the impulse file. Larger percentage
indicates a longer reverb.
¾¾ Damping LF: This option is used for adding clarity and generating a precise sound by decreasing low-frequency,
bass components, and muddiness.
¾¾ Damping HF: This option is used for generating a warmer, lusher sound by decreasing high-frequency, transient
parts, and harshness.
¾¾ Pre-Delay: This option specifies the time that the reverb will take to produce to maximum amplitude. A setting of
short pre-delay 0-10 milliseconds creates a natural sound. Whereas, longer pre-delays of 50 milliseconds and
more produces interesting special effects.
¾¾ Width: This option manages the stereo spread. A 0 value produces a mono reverb signal.
¾¾ Gains: This option is used either for boosting or mellowing down the amplitude after processing.
4.8.2 Full Reverb Effect
Unique options, such as Perception, which simulates room irregularities, left/right location, which places the source off
center, and Room Size and Dimension, which realistically simulate rooms can be customized using this option.

To simulate the surfaces of the wall and tone, the reverb frequency absorption can be altered using a three-band,
parametric EQ in the Coloration section.

Click here to know more about the settings.

4.8.3 Studio Reverb Effect

Studio Reverb effect simulates audio spaces. It is quicker and less processor intensive than the other reverb effects,
however, because it is not based on convolution. Thus, real time changes can be made quickly and effectively in the
Multitrack Editor, without pre-rendering effects on a track. Following are the sub-options available:
¾¾ Room Size: This option sets the room size.
¾¾ Decay: This option adjusts the amount of echo decay in terms of milliseconds.
¾¾ Early Reflections: This option controls the percentage of the first perceiving echoes, creating a feel of the
overall room size. A high value feels artificial, whereas a low value loses the audio cues for the room’s size. Half
the volume of the original one is an ideal option.
¾¾ Stereo Width: This option manages the spread across the stereo channels. 0% produces a mono reverb signal,
whereas the maximum stereo separation is at 100%.
¾¾ High Frequency Cut: This option defines the highest frequency at which reverb can occur.
¾¾ Low Frequency Cut: This option defines the lowest frequency at which reverb can occur.
¾¾ Damping: This option adjusts the attenuation amount, which is applied to the high frequencies of the reverb
signal.
¾¾ Diffusion: This option generates the reverberated signal’s absorptions as it is reflected off the surfaces. Lower
the settings, more are the echoes, whereas higher the settings, smoother reverberation with fewer echoes is
created.
¾¾ Dry: This option defines the percentage of source audio to output with the effect.
¾¾ Wet: This option defines the percentage of reverb to output.
4.8.4 Surround Reverb Effect
This effect is basically applicable for 5.1 sources, but it can also generate surround ambience for mono or stereo
sources as well. In the Waveform Editor, select Edit, and then click Convert Sample Type for converting a file to 5.1
from mono or stereo, and then apply Surround Reverb. In case of Multitrack Editor, the mono or stereo tracks can be
directly sent to a 5.1 bus or master with Surround Reverb.
¾¾ Damping LF: This option is used for adding clarity and generating a precise sound by decreasing low-frequency,
bass components, and muddiness.
¾¾ Damping HF: This option is used for generating a warmer, lusher sound by decreasing high-frequency, transient
parts, and harshness.

Click here to know more about the settings.

4.9 Special Effect

4.9.1 Distortion
This option produces identical curves in both positive and negative graphs.
■■ Positive and Negative graphs

Both Positive and negative graphs indicate separate distortion curves for positive and negative values respectively.
The horizontal x axis indicates input level in terms of decibels, whereas the vertical y axis denotes the output level.
The default diagonal line indicates an undistorted signal, showing a direct relationship between input and output
values. Click and drag the points to create and adjust on graphs, whereas drag the points off the graph to remove
them.

4.9.2 Doppler Shifter Effect

This effect is useful for creating an increase and decrease in pitch, especially when a moving object nears and then
moves far away. Refer to Figure 4.13. It has two options— namely, Straight Line and Circular.
 Figure 4.13: Doppler shifter
■■ Straight Line options:
¾¾ The Starting Distance Away option sets the virtual starting point in terms of meters.
¾¾ The Velocity option specifies the speed, in terms of meters per second, at which the effect moves.
¾¾ The Coming From option defines the direction from which the effect appears to come, in terms of degrees.
¾¾ The Passes In Front option specifies how far the effect looks like passing in the listener’s front, measured in
terms of meters.
¾¾ The Passes On Right option specifies how far the effect looks like passing to the listener’s right.
■■ Circular options:
¾¾ The Radius option sets the circular dimensions of the effect in terms of meters.
¾¾ The Velocity option specifies the speed, in terms of meters per second, at which the effect moves.
¾¾ The Starting Angle option specifies the starting of the effect’s virtual angle, in terms of meters.
¾¾ The Center In Front By option defines the distance at which the sound source is from the listener’s front.
¾¾ The Center On Right By option defines how far the sound source is from the listener’s right.
■■ Adjust Volume Based on Distance or Direction

This option adjusts the effect’s volume based on the values mentioned.
■■ Quality Level

This option makes available six different levels of processing quality. Lower the quality levels, lower processing time,
higher the levels, higher the time but with better sound results.

4.9.3 Guitar Suite Effect

This effect implements a series of processors that use and alter the sound of guitar tracks. The Compressor stage
produces a tighter sound by reducing the dynamic range. Filter, Distortion, and Box Modeler stages generate effects
similar to which guitarists use to create expressive, artistic performances. Following are the sub-options available:
¾¾ Compressor: This option decreases the dynamic range so as to maintain consistent amplitude, such that the
sound of the guitar stand out in the audio.
¾¾ Filter: This option generates guitar filters ranging from resonators to talk boxes. Choose an option from this
menu, and then set its sub-options.
¾¾ Type: This option specifies the frequencies to be filtered. Specify Lowpass for filtering high frequencies,
Highpass for filtering low frequencies, or Bandpass for filtering frequencies above and below a center frequency.
¾¾ Freq: This option determines the cutoff frequency for Lowpass and Highpass filtering, or the center frequency
for Bandpass filtering.
¾¾ Resonance: This option is used for feeding back frequencies near the cutoff, and adds sharpness with low
 settings and whistling harmonics with high settings.
¾¾ Distortion: This option is used for adding a sonic edge used in solo guitar audios. To change the distortion
character, select an option from the Type menu.
¾¾ Amplifier: This option generates various amplifier and speaker combinations for creating unique tones.
¾¾ Mix: This option manages the ratio of original to processed audio.
4.9.4 Mastering Effect
Mastering refers to a complete process of optimizing audio files for mediums, such as radio, video, CD, or the Web.
Mastering an audio is possible by clicking Special option and selecting Mastering effect.

However, before mastering an audio, it is essential to take into consideration the nature of the destination audio and
according adjust the settings so as to deliver a clear and articulate sound. Refer to Figure 4.14.

Figure 4.14: Mastering

Click here to know more about the properties.

4.9.5 Vocal Enhancer Effect

The Vocal Enhancer effect improves the quality of voice-over recordings immediately. The Male and Female modes
automatically reduce sibilance and plosives, as well as low rumbles occurred due to microphone placing. Those modes
also apply microphone modeling and compression to give vocals a characteristic radio sound. The Music mode used
soundtracks as they better suit a voice-over.
¾¾ Male: This option optimizes audio for a man’s voice.
¾¾ Female: This option optimizes audio for a woman’s voice.
¾¾ Music: This option applies compression and equalization to music or background audio.

4.10 Stereo Imaginary Effects

4.10.1 Center Channel Extractor
The Center Channel Extractor effect applies mostly on voice, bass, and lead instruments where the sounds are
panned to the center by keeping or deleting frequencies that are common to both the left and right channels. As a result,
volume of vocals, bass, or kick drum, can be enhanced as well as can be removed to create a karaoke mix. Following
are the two tabs available:
¾¾ Extraction tab: This limits the extraction to audio after achieving certain properties.
●● Extract: Select audio either in the Center, Left, Right, or Surround channel, or choose the Custom option
and specify the exact phase degree, pan percentage, and delay time for audio that has to be extracted or
removed.

●● Frequency Range: This option is used to set the range that has to be extracted or removed. Some of the
 predefined ranges include Male Voice, Female Voice, Bass, and Full Spectrum. Click Custom for defining
a frequency range.
¾¾ Discrimination tab: This option includes settings that identify the center channel.
●● Crossover Bleed: To make the audio bleed and sound more realistic, move the slider to the left. Whereas,
for separating the center channel material from the mix, move the slider to the right to further separate.

●● Phase Discrimination: In general, a range from 2 to 7 works with most of the audio, wherein higher
numbers are better for extracting the center channel, while lower values are good for removing the center
channel. However, it has to be noted that lower values allow bleed through more but not effectively separate
vocals from a mix. However, such settings prove good for capturing all the center material.
■■ Center and Side Channel Levels

This effect specifies how much amount of selected signal has to be extracted or removed. The sliders can be moved
up for including additional material.
¾¾ Advanced options: To access its sub-options, click the triangle.
¾¾ FFT Size: This option specifies the FFT size, wherein low settings give good speed and high settings improves
quality. In general, settings are between 4096 and 8192.
¾¾ Overlays: This option the number of overlapped FFT windows. Higher values produce smoother results or a
chorus like effect and take longer time to process. Whereas, lower values produce bubbly sounding background
noises. In general, values are between 3 to 9.
¾¾ Window Width: This option specifies each FFT window’s percentage . Values between 30% to 100% work best.
■■ Amplitude Discrimination and Amplitude Bandwidth

Adobe Audition adds the left and right channels for creating a third channel that removes similar frequencies. If the
amplitude at each frequency is similar, the in-phase audio which is common to both channels is also considered.
Lower values for Amplitude Discrimination and Amplitude Bandwidth cut more audio from the mix along with
vocals. Higher values make the extraction rely more on the phase of the material and less on the channel’s amplitude.
Amplitude Discrimination settings between 0.5 and 10 and Amplitude Bandwidth settings between 1 and 20 are
best suitable.
■■ Spectral Decay Rate

This option can be kept at 0% for faster processing or between 80% and 98% to even out background distortions.

4.10.2 Graphic Phase Shifter

This effects manages the waveform’s wave by adding control points to a graph. The x-axis measures frequency, while
the y-axis displays the degree of phase to shift, where zero represents no phase shift. A simulated stereo can be
produced by creating a zigzag pattern , which is at the extreme high end of one channel. Following are the sub-options
available:
¾¾ Frequency Scale: This option places the x-axis on a linear or a logarithmic scale, which works at lower
frequencies. However, Linear option can be selected if details of higher frequencies have to be worked with.
¾¾ Range: This option sets the values of the vertical ruler on a 360° or 180° scale.
¾¾ Channel: This option specifies on which channels the phase shift has to be applied.
¾¾ FFT Size: This option specifies the FFT size. Higher sizes create precise results but also take longer time to
process.

4.11 Time and Pitch

4.11.1 Automatic Pitch Correction
¾¾ Scale: This option defines which of the scale types — namely, Major, Minor, or Chromatic best suits the
audio. Major or Minor correct notes to the specific music key, whereas chromatic corrects to the nearest note
irrespective of the key.
¾¾ Key: This option manages the intended key for the corrected audio. In fact, the combination of scale and key
determines the key signature. However, this option will be available only if Major or Minor scale types are set.
¾¾ Attack: This option manages the duration taken for correcting the pitch towards the scale tone. However, fast
 settings are advised for short duration notes.
¾¾ Sensitivity: This option defines the Threshold beyond which notes are not corrected. Sensitivity is measured in
terms of cents, wherein 100 cents equals to a semitone.
¾¾ Reference Channel: This option is used for choosing a source channel where the pitch changes are most clear.
This effect analyzes only that channel which is chosen, but the pitch correction is equally applied to all channels.
¾¾ FFT Size: This option is used to set the FFT size of each piece of data that the effect is processing. It is
advisable to use smaller values for correcting higher frequencies.
¾¾ Calibration: This option specifies the standard tuning for the audio. Though, the source audio is recorded using
a different standard, A4 values can be specified from 410 to 470 Hz.
¾¾ Correction meter: This option helps preview audio, along with displaying the amount for flat and sharp tones
correction.

4.11.2 Manual Pitch Correction

The Manual Pitch Correction effect in the Time and Pitch option visually adjusts pitch using the Spectral Pitch
Display. The Spectral Pitch Display displays the fundamental pitch as a bright blue line, and overtones in yellow to
red hues. The Corrected Pitch appears as a bright green line. In the Manual Pitch Correction window, the following
options are available:
¾¾ Reference Channel: This option is used for choosing a source channel where the pitch changes are most clear.
This effect analyzes only that channel which is chosen, but the pitch correction is equally applied to all channels.
¾¾ Spline Curves: This option produces smoother transitions for applying different pitch correction overtime using
envelope keyframes.
¾¾ Pitch Curve Resolution: This option is used to set the FFT size of each piece of data that the effect is processing.
It is advisable to use smaller values for correcting higher frequencies. For voices, a setting of 2048 or 4096
sounds more natural, and a setting of 1024 creates robotic effects.

4.12 Summary
In this session, Working with Audio Effects, you learned that:
¾¾ Adobe Audition offers a wide range of effects which work with both Waveform and Multitrack Editors. In order
to work with the effects that are compatible with both of them, the Effect Rack, the Effects menu, and the
Favorites menu can be used.
¾¾ The Invert effect alters the signal’s polarity, Reverse flips the flow of the audio and the Silence effect replaces
the selected audio with silence.
¾¾ The Amplitude and Compression option increases the effect or soothes an audio signal. It can be combined
with the other effects in the Effect rack as it operates in real time. It has other effects such as Channel Mixer,
DeEsser, Dynamics Processing, Hard Limiter effect, Multi Band Compressor effect, Normalize effect,
Single Compressor effect, Tube Compressor effect, and Volume Envelope effect.
¾¾ When an original signal is copied as separate files and reoccur every millisecond, such an effect is called a
Delay. Whereas, in case of Echoes, sounds are delayed such that an exact copy of the original sound is heard.
Thus, both delays and echoes are a great way to add ambience to the sound.
¾¾ Diagnostics tool is useful for smoothening the audio by removing clicks, distortion, or silence as well as adds
markers where silence exists. It has options such as DeClicker effect, DeClipper options, and Delete Silence
and Mark Audio options.
¾¾ FFT refers to Fast Fourier Transform, which is an algorithm that quickly analyzes frequency and amplitude.
The Graphic Equalizer effect enhances or attenuates specific frequency bands and provides a graphic
representation of the output EQ curve. The Notch Filter effect removes to a maximum of six user defined
frequency bands, which includes even narrow frequency bands, such as a 60 Hz hum, leaving the surrounding
frequencies unaffected. The Parametric Equalizer effect provides maximum control over tonal equalization.
¾¾ Modulation includes effects, such as Chorus effect, Flanger effect, and Phaser effect.
¾¾ Noise Reduction / Restoration options include Adaptive Noise Reduction effect, Automatic Click Remover
Effect, Automatic Phase Correction effect, DeHummer effect, and Hiss Reduction.
¾¾ Reflected sound is referred to as reverberation or reverb. It includes effects, such as Convolution effect, Full
Reverb effect, Studio Reverb effect, and Surround Reverb effect.
 ¾¾ Special effects include Distortion, Doppler Shifter effect, Guitar Suite effect, Mastering effect, and Vocal
Enhancer effect.
¾¾ Stereo Imaginary Effects include Center Channel Extractor and Graphic Phase Shifter.
¾¾ Time and Pitch includes Automatic and Manual pitch correction.

4.13 Exercise
1. The ____________ effect enhances or attenuates specific frequency bands and provides a graphic representation
of the output EQ curve.

Graphic Equalizer effect

Hard Limiter Effect

DeClicker effect

Doppler Shifter Effect

2. The __________ effect is used for repairing clipped waveforms in audio by filling it with new audio data.

DeClicker

DeClipper

Flanger

DeHummer

3. ___________ effect can be used to enhance a vocal track or add stereo effect to mono audio, finally resulting in a
lush, rich sound.

Echo

Delay

Distortion

Chorus

4. Reverb Fast Fourier Transform is an algorithm that quickly analyzes frequency and amplitude.

True

False

5. The Graphic Phase Shifter effect applies mostly on voice, bass, and lead instruments where the sounds are panned
to the center.

True

False
 Answer Key

Quick Answers Exercise Answers

Quick Test 1.1 Exercise 1.6
1. False 2. False 1. True 2. False
3. Frequency 4. False
Quick Test 2.1 5. Effect Rack 6. True
1. False 2. False 7. Encoding

Quick Test 3.1 Exercise 2.6

1. Ripple Delete 2. False 1. Move 2. Mixing audio clips
3. False 4. Adjust Selection
Quick Test 4.1 Inward
1. Favorites menu 2. True 5. Changes are saved 6. True
3. T u b e - m o d e l e d to the currently
Compressor selected audio as a
new file
7. Bit depth 8. Numbers
Quick Test 4.2
1. False 2. DeClicker
Exercise 3.5
3. True
1. True 2. Read
3. Clip Automation 4. Latch
5. Sends

Exercise 4.13
1. Graphic Equalizer 2. DeClipper
effect
3. Chorus 4. True
5. False
 Bibliography
Adobe Audition CS6 Classroom in a Book
- Adobe Creative Team