Audio file formats in English

64-bit doubles (RAW) (.dbl)

This format uses 8-byte doubles in binary form—8bytes per sample mono, or 16 bytes per sample stereo interleaved.

The 64-bit doubles format has no header—it’s purely audio data, just like the raw PCM format.

8-bit signed (.sam)

This format is popular for building MOD files, since audio in MOD files is 8-bit signed. Many MOD editors allow

samples to be inserted fromor exported to files in this format. Files with the .samextension contain 8-bit signed raw

data, and by default, they have no headers. The sample rate starts off as 22050 Hz, but you can change the sample

rate after you open the file by choosing Edit > Adjust Sample Rate.

A/mu-Law Wave (.wav)

The A-Law and mu-Law WAV formats (CCITT standard G.711) are common in telephony applications. These

encoding formats compress the original 16-bit audio to 8-bit audio (for a 2:1 compression ratio) with a dynamic

range of about 13-bits (78 dB). While A-Law and mu-Law encoded waveforms have a higher signal-to-noise ratio

than 8-bit PCM, they also have a bit more distortion than the original 16-bit audio. Still, the quality is higher than

you would get with some 4-bit ADPCM formats.

Note: Files saved in this format expand automatically to 16-bits when opened, so you shouldn’t save 8-bit files in this


Choose from the following options:

A-Law 8-bit A slight variation of the standard mu-Law format and is found in European systems.

mu-Law 8-bit The international standard telecommunications encoding format and is the default option.


User Guide


ACM Waveform (.wav)

Microsoft ACM (Audio CompressionManager) is part of all 32-bit versions ofWindows. Adobe Audition supports

the ACMdriver, which enables you to open and save files in a variety of formats other than those directly supported

by Adobe Audition.

Some of these formats come as a standard part of Windows, while others are provided by third-parties. You may

acquire ACM formats when you install other software.

To save a file in an alternate format by using the ACMdriver, choose File > Save As, choose ACMWaveform as the

file format, and clickOptions. You can select fromamong various quality levels, and each level will give you different

options for formats and attributes.

Note: The ACMdriver you want to usemight require that the file be in a specific format before saving it. For example,

if you want to save a file in the DSP Group TrueSpeech format, you should first use the Edit > Convert Sample Type

command to convert the file to 8 KHz, mono, 16-bit, because that is the only format that the TrueSpeech ACM driver

supports. Formore information on any particularACMdriver, contact the creator of the format (such as DSPGroup for

TrueSpeech, or CCITT for the various CCITT formats) or the manufacturer of the hardware that uses the format in


Amiga IFF-8SVX (.iff, .svx)

The Amiga IFF-8SVX format is an 8-bit mono format from the Commodore Amiga computer.

Choose from the following options:

Data Formatted As Saves the audio file in uncompressed 8-bit Signed format (the default setting) or in the

compressed 4-bit Fibonacci Delta Encoded format.

Dithering From 16-bit Specifies a type of dithering for 16-bit files: TriangularDither, Shaped Gaussian Dither,Noise

Shaping A, orNoise Shaping B.NoDithering is the default. Formore information on types of dithering, see “Change

the bit depth of a file” on page 99.

Apple AIFF (.aif, .snd)

AIFF is the Apple® standard audio file format. AIFF supports mono or stereo files, 16-bit or 8-bit resolution, and a

wide range of sample rates. Adobe Audition supports only the PCM-encoded portion of the data, even though this

format (like Windows WAV) can contain any one of various data formats.

AIFF is a good choice for cross-platformcompatibility in bothWindows andMac OS. Before you open AIFF files in

Adobe Audition, add the .aif or .snd extension to the file and open it by using the Apple AIFF file filter. When you

transfer an AIFF file to aMacintosh, you can add the four character code “AIFF” in the file’s resource fork to have it

recognized. (The Macintosh identifies a file through its “resource,” which is removed when a file is opened on a

Windows computer. However, many Mac OS applications that support AIFF can recognize the PCM data without

this identifier.)

ASCII Text Data (.txt)

Audio data can be read from or written to files in a standard text format, with each sample separated by a carriage

return, and channels separated by a tab character. An optional header can be placed before the data. If no header text

exists, then the data is assumed to be 16-bit signed decimal integers. The header is formatted as a KEYWORD: value


values for NORMALIZED are either TRUE or FALSE. For example,


User Guide






164 <tab> -1372

492 <tab> -876

Choose any of the following options:

Include Format Header Places a header before the data.

Normalized Data Normalizes the data between –1.0 and 1.0.


Creative Sound Blaster (.voc)

This format is for Sound Blaster and Sound Blaster Pro voice files. Adobe Audition supports both the older and

newer formats. The older format supports only 8-bit audio, mono to 44.1 kHz and stereo to 22 kHz. The newer

format supports both 8- and 16-bit audio.

Files in this format can contain information for looping and silence. If a file contains loops and silence blocks, they

expand when you open the file.

Choose one of the following options:

Old Style Saves audio as an 8-bit .voc file that can be played on any Sound Blaster card.

New Style Saves audio to the newer format that supports both 8- and 16-bit audio.

Dialogic ADPCM (.vox)

The Dialogic ADPCM (Adaptive Differential Pulse CodeModulation) format is used in telephony applications, and it’s optimized for voice at low sample rates. It supports only mono 16-bit audio, and like other ADPCM formats, it compresses the audio data to 4 bits/sample (4:1). This format has no header, so Adobe Audition assumes any .vox file is in Dialogic ADPCM format.

Note: Take note of the sample rate of the audio before saving it, as you need to enter it upon reopening the file.

DiamondWare Digitized (.dwd)

This format is used by DiamondWare Sound Toolkit, a programmer's library that lets you quickly and easily add high-quality interactive audio to games and multimedia applications. It supports both mono and stereo files at a variety of resolutions and sample rates.


The International Multimedia Association (IMA) flavor of ADPCM compresses 16-bit data to 4 bits/sample (4:1) by

using a different (faster) method than Microsoft ADPCM. It has different distortion characteristics, which can

produce either better or worse results depending on the sample being compressed. As withMicrosoft ADPCM, use

this format with 16-bit rather than 8-bit files. This compression scheme can be a good alternative to MPEG; it

provides reasonably fast decoding of 4:1 compression, and it degrades sample quality only slightly.

Choose from the following options:

2 bits/sample, 8:1 Produces files with the highest compression ratio (8:1) but with the lowest number of bits. Select this option if smaller file size is more important than audio quality. Keep in mind that this compression rate is less compatible than the standard 4-bit and is supported on fewer systems.

3 bits/sample, 5.3:1 Produces higher quality than the 2 bits option, but the quality isn’t quite as good as with the 4 bits and 5 bits options. Some systems might have problems playing back files with this compression rate, especially stereo files.

4 bits/sample, 4:1 Produces 4-bit files at a compression ratio of 4:1. This option is the default.

5 bits/sample, 3.2:1 Produces files with the highest quality, since more bits and a lower compression ratio are used.

However, this compression rate is less compatible than the standard 4-bit.

Microsoft ADPCM (.wav)

TheMicrosoft ADPCM format provides 4:1 compression. Files saved in this format expand automatically to 16-bits when opened, regardless of their original resolution. For this reason, use this format with 16-bit rather than 8-bit files.

Choose from the following options:

Single Pass (Lower Quality) Compresses files in a single pass. Use this option if you’re in a hurry. However, the quality is lower than if you use the Multiple Pass option. The time taken to read an ADPCM-compressed file is the same no matter which option you use.

Multiple Pass (Higher Quality) Compresses files in multiple passes, providing better quality. This setting is the default.

Block Size Offers three size options, each with a different compression ratio and quality level: Large (Default

Quality), with a compression ratio of 3.98:1;Medium (Good Quality), with a compression ratio of 3.81:1; and Small (High Quality), with a compression ratio of 3.25:1.

mp3PRO (.mp3)

The mp3PRO filter enables Adobe Audition to directly encode and decode .mp3 files.When you save a file to mp3

format, the audio is encoded and compressed according to the options you select. When you open an .mp3 file, the audio converts into the uncompressed internal format of Adobe Audition. As a result, you can save an .mp3 file in any format.

Avoid compressing the same audio tomp3more than once.Opening and resaving an .mp3 file causes it to be recompressed, so any artifacts from the compressing process become more pronounced.

MP3/mp3PRO Encoder Options dialog box contains two sets of options: simple options for choosing an encoding method and more advanced options. To view the advanced options, click Advanced. To view only the basic options, click Simple.

Simple mp3 options

CBR (Constant Bitrate) Encodes the same bit rate throughout the entire file. This method is the most common and the most predictable for bandwidth and file size.

VBR (Variable Bitrate) Encodes higher bit rates for more complex material and lower bit rates for simpler material.

While it depends on the source material, VBR-encoded .mp3 files generally tend to be smaller than CBR-encoded

.mp3 files. Use the menu below the VBR option to choose a quality level from 10 (lowest quality but smaller file) to

100 (highest quality but larger file). Some mp3 players don’t support VBR-encoded files. For maximum compatibility,

select CBR.

MP3 Encodes the file to mp3, but without the PRO data.

mp3PRO Encodes the file with PRO data that helps re-create high frequencies in the compressed file, especially at

low bit rates. An mp3PRO file can still be played back by an mp3 player that doesn’t support the PRO data, but the

quality may be lower than for a standardmp3 file of that bit rate. For example, a 64 Kbps mp3PRO file sounds more

like a 112 Kbps or 128 Kbpsmp3 file if the player supportsmp3PRO, but it sounds like a 64 Kbpsmp3 file (or worse)

if the player doesn’t support mp3PRO.

Advanced mp3 options

Maximum Bandwidth (Available only if MP3 is selected) Specifies the highest frequency that will be encoded. Lower bandwidths help eliminate tinkly and phase-like effects but at the expense of reducing the higher frequencies.

CBR Bitrate (Available only if CBR and MP3 are selected) Specifies the bit rate for CBR encoding. The higher the number, the larger the file, but the better the quality. Valid values range from 20 Kbps to 320 Kbps.

Sample Rate (Available only if CBR and MP3 are selected) Specifies the sample rate of the destination file. (The

decoder will also use this rate.) Keep in mind that not all sample rates are valid for a particular bit rate.

VBR Quality (Available only if VBR is selected) Specifies the quality for VBR encoding. The higher the number, the larger the file, but the better the quality. Valid values range from 1 to 100.

Low Complexity Stereo (Available only if CBR and mp3PRO are selected) Encodes the audio as mono, with information on how to reconstruct the stereo signal at playback. A non-PRO decoder plays back only mono, but a PRO decoder plays back stereo. The stereo image is different fromthe original audio, but itmost often sounds better than

its mono counterpart.

Codec Provides three codec options. Depending on the type of audio, one codec might do a better job than the

others. Experiment to see which one is best for your project. Current-Best Quality is an extremely fast algorithm, and it generally does a very good job at lower bit rates, as well as giving more high-frequency detail without

unwanted artifacts. Unless you have a specific reason not to, use this setting. Legacy-Medium Quality (Fast) uses a different model for encoding and can be more complete at bit rates above 160 Kbps. Legacy-High Quality (Slow)

takes longer to encode, but the quality is higher than that of the Medium Quality option.

Allow Mid-Side Joint Stereo Combines the left and right channels by using a Mid-Side method when encoding middle quality bit rates and below. This option preserves surround-sound information by saving the common audio

in one channel while the difference between the channels is saved in the other.


User Guide


Allow Intensity Joint Stereo Combines the left andright channels for files encodedat lowbit rates. Some frequencies

are saved as mono and placed in the stereo field based on the intensity of the sound.

Note: Don’t use this option if the stereo audio contains surround-encoded material.

Allow Narrowing Of Stereo Image Usesmore data to represent a wider stereo image. This option allows the encoder

to narrow the image in some parts in order to make the overall audio quality better.

Set ‘Private’ Bit Sets the Private bit for each MPEG frame.

Set ‘Copyright’ Bit Sets the Copyrighted bit on the .mp3 file.

Set ‘Original’ Bit Sets the Original Copy bit, which designates that the .mp3 file is on its original media.

Padding Specifies the padding required by the decoder. ISO Padding is the default, but you can choose a different

setting if the decoder needs no padding or always needs padding.

Set All Decoding To 32-Bit Determines how .mp3 files are opened in Adobe Audition. Selecting this option forces

Adobe Audition to upsample non-32-bit .mp3 files to 32-bit. Deselecting this option allows .mp3 files to be opened

with the original bit depth intact.

Encode Stereo As Dual Channel Encodes two audio channels with independent contents within one bitstream.

Write CRC Checksums Adds CRC checksums to the audio streamso that content can be verified for any errors when decoded.

NeXT/Sun (.au, .snd)

The NeXT/Sun format is standard on NeXT and Sun™ computers, and it has many data types. Adobe Audition supports the CCITT A-Law,mu-Law, G.721 ADPCM, and linear PCM data variants. LikeWindows PCM and AIFF, this format can support mono or stereo, 16- or 8-bit, and a wide range of sample rates when saved as linear PCM.

The NeXT/Sun format is most commonly used for compressing 16-bit data to 8-bit mu-law data. AU is used quite extensively on the Web and in Java applications and applets.

Choose from the following options:

mu-Law 8-bit Uses the mu-law 8-bit format to compress the file.

A-Law 8-bit Uses the A-law 8-bit format to compress the file.

G.721 ADPCM 4-bit Applies the standard CCITT G.721 compression to the file (ADPCM at 32Kbps).

Linear PCM Saves the file as uncompressed, linear PCM (Pulse Code Modulation).

Ogg Vorbis (.ogg)

The Ogg Vorbis format is comparable to other formats used to store and play digital music, such as MP3, VQF, and

AAC. Unlike those formats, however, Ogg Vorbis is license-free, so it’s often used for commercial video games.

When you save an OGG file, you can either select one of three basic encoding options, or use advanced settings for

detailed control:

VBR (Target Bitrate) Lets you specify the target bitrate in kilobits per second.Maintains audio quality by varying the

bitrate depending on the complexity of the audio being encoded. This method can maintain higher audio quality,

although file size is not as predictable as with Fixed Bitrate encoding.

VBR (Quality Index) Like VBR (Target Bitrate), but lets you specify quality on a scale of 0 to 10.


User Guide


Fixed Bitrate Varies the quality level as needed to ensure that the bit rate stays at the specified rate. This method

makes a consistently sized file, although the quality may not be as high as with Variable Bit Rate encoding.

Use Advanced Settings Enables the following options:

• Minimum, Target, and Maximum Bitrate Let you precisely specify compression settings.

• Bit Reservoir Size Specifies the amount of surplus bits to reserve during variable bitrate encoding.

• Bit Reservoir Bias Determines how surplus bits are distributed. Lower settings store surplus bits during consistent

audio, instead applying those bits to transient peaks and troughs. Higher settings store surplus bits during transients,

instead applying those bits to consistent audio. The default setting, 0.2, slightly favors transients.

• Impulse Noise Floor Sets the amplitude above which the encoder looks for artifacts in transient peaks. Lower

noise floors improve transient response but increase bitrate.

• Damping Time Determines how quickly bitrate returns to the targeted average. At lower settings, bitrate varies

less, but audio quality suffers; at higher settings, bitrate varies more, but audio quality improves.

• Lowpass Filter Specifies the highest frequency to retain in the encoded file.

SampleVision (.smp)

The Sample Vision format is native to the Turtle Beach SampleVision program. This format supports only mono,

16-bit audio. If a file is in a different format, Adobe Audition prompts you to convert it before saving it.

This format also supports loop points, which you can edit in theMarker List panel. The Label of the marker must be in the format Loop n, mwhere “n” is the loop number from 1 to 8, and “m” is themode (0 = no looping, 1 = forward

loop, 2 = forward/back loop).

Spectral Bitmap Image (.bmp)

Thoughmost applications store conventional images in BMP format, Adobe Audition can save and import audio in

this format.Whenyousave audio as a bitmapimage,AdobeAudition creates two image files:one reflects the spectral

frequency graph, the other stores data that correctly aligns phase when you reimport the graph. (The latter file

includes phase in the filename.) You can incorporate exported graphs into visual presentations, or modify them in

image-editing applications like Adobe Photoshop.

In an image-editing application, apply gradients to create audio fades, and adjust opacity to change audio amplitude.

To protect your work, apply visual watermarks above the audible frequency range.

For information about importing spectral graphs, as well as photographs, logos, and other visually-oriented files, see

“Import a bitmap image as audio” on page 44.

Windows Media Audio (.wma)

The WMA format utilizes a perceptual compression scheme and lets you select from three different encoding


Constant Bit Rate Encoding Varies the quality level as needed to ensure that the bit rate stays the same. Thismethod

makes a consistently sized file, although the quality may not be as high as with Variable Bit Rate encoding.

Variable Bit Rate Encoding Maintains the audio quality by varying the bit rate depending on the complexity of the

audio passage being encoded. Thismethod can maintain higher quality audio in the file, although the file size is not

as predictable as with Constant Bit Rate encoding.

Mathematically Lossless Encoding Compresses to a smaller file size than WAV, but results in no fidelity loss.


User Guide


After you select an encoding option, you can set the desired quality. Just as with stereoWMAfiles, the higher quality

setting you select, the larger the file size, and vice versa.

Windows PCM (.wav, .bwf)

The Microsoft Windows PCM format supports both mono and stereo files at a variety of resolutions and sample rates. It follows the RIFF (Resource Information File Format) specification and allows for extra user-information to

be embedded and saved with the file. The WAV format reproduces digital audio by using PCM (Pulse Code

Modulation)—PCM doesn’t require compression and is considered a lossless format.

You can include BroadcastWave metadata inWindows PCM files. (See “Add audio file information” on page 253.)

The following options are available for 32-bit files; no options are available for 8- or 16-bit files:

32-bit Normalized Float (type 3) – Default The internal format for Adobe Audition and the standard floating point format for type 3 .wav files. Values are normalized to the range of +/–1.0, and although values beyond this range are

saved, clipping may occur in some programs that read them back in. (Adobe Audition won’t clip audio but will instead read the same value back if it’s beyond this range.)

4-byte PCM (type 1, 32-bit) Saves 32-bit audio as 32-bit integers, retaining 192 dB of dynamic range. However,

signal-to-noise ratio in saved files is 144.5 dB, reflecting the 32-bit float format that Adobe Audition uses internally.

24-bit Packed Int (type 1, 24-bit) Saves straight 24-bit integers so any data beyond the bounds is clipped. The .wav

BitsPerSample is set to 24 and BlockAlign is set to 3 bytes per channel.

24-bit Packed Int (type 1, 20-bit) Saves straight 24-bit integers so any data beyond the bounds is clipped. The .wav

BitsPerSample is set to 20 and BlockAlign is set to 3 bytes per channel. The extra 4 bits are actually the remaining

valid bits when saving, and they are used when reading (thus still giving 24-bit accuracy if those bits were actually present when writing). Applications either fill those last 4 bits with zeros or with actual data; analog/digital

converters that generate 20 bits of valid data automatically set the remaining 4 bits to zero. Any type 1 format with BlockAlign set to 3 bytes per channel is assumed to be packed integers, and a BitsPerSample value between 17 and

24 will read in all 24 bits and assume the remaining bits are either accurate or set to zero.

32-bit 24.0 Float (type 1, 24-bit) – Non-Standard Saves full 32-bit floats (ranging from +/–8million), but the .wav

BitsPerSample is set to 24 while BlockAlign is still set to 4 bytes per channel.

16.8 float – Obsolete/Compatibility The internal format used by Adobe Audition 1.0. Floating point values range from+/–32768.0, but larger and smaller values are valid and aren’t clipped since the floating point exponent is saved

as well. The .wav BitsPerSample is set to 32 and BlockAlign is set to 4 bytes per channel.

Enable Dithering Dithers 32-bit files when they are saved to a PCM format (20-bit, 24-bit, or 32-bit). This option is available only for a 32-bit file that you select to save to a nonfloating-point type format. It applies a Triangular dither

with a depth of 1.0 and no noise shaping. If you wish to apply a noise-shaped dither, use the Edit > Convert Sample Type command to dither the audio first, and then save the file without dithering enabled in the file format options.

PCM Raw Data (.pcm, .raw)

This format is simply the PCM dump of all data for the wave. No header information is contained in the file. For this

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