• MP3 (MPEG 1/ 2/ 2.5 Layer 3)
• MP2 (MPEG 1 Layer 2)
• WMA
• WAVE
• OGG
• FLAC
• Monkey's Audio (APE)
• MusePack (MPC)
• AIFF
• True Audio (TTA)
• OptimFROG (OFR)
• VOX

MPEG Layer-3 format. Very popular format for keeping of music.

It became the de facto standard for lossy audio encoding, due to the high compression rates (1/12 of the original size, still remaining considerable quality), the high availability of decoders and the low cpu requirements for playback. Sampling frequencies from 16khz to 24khz (mpeg2 layer 3) and 32khz to 48khz (mpeg1 layer 3). Formal and informal listening tests have shown that mp3 at the 192-256 kbps range provide encoded results undistinguishable from the original materials in most of the cases.

MPEG Version 2.5 was added lately to the MPEG 2 standard. It is an extension used for very low bitrate files, allowing the use of lower sampling frequencies. LAME codec is used for MP3 encoding.

The extension is *.mp3.

MP3 Website: www.mpeg.org
LAME Codec Website: www.mp3dev.org

[Back to the top]

MPEG Layer-2 format. Compression ratio is 1:6...1:8 corresponds to to 256..192 kbps for a stereo signal.

The file extension(s) are *.mp2 or *.mpa.

MP2 Website: www.mpeg.org

[Back to the top]

Windows Media Audio format. A special type of advanced streaming format file for use with audio content encoded with the Windows Media Audio codec.

Windows Media Format modules are required for the conversion. It comes with Windows Media Player, or you can download and install it from here.

The extension is *.wma.

WMA Website: www.microsoft.com

[Back to the top]

This format was created by Microsoft and IBM, and it has unfortunately become a popular standard. It specifies an arbitrary sampling rate, number of channels and sample size. It also specifies a number of application-specific blocks within the file. It has a plethora of different compression formats.

WAVE files can be converted by different codecs. It supports the following types of codecs:

• Pulse Code Modulation (PCM)
  Standard Windows WAV format for non-compressed audio files. Pulse Code Modulation (PCM) is the standard method of digitally encoding audio. It is the basic uncompressed data format.
• Adaptive Differential Pulse Code Modulation (ADPCM)
  Compressed WAV format. Adaptive Differential Pulse Code Modulation (ADPCM) is an audio compression scheme which compresses from 16-bit to 4-bit for a 4:1 compression ratio.
  
  ADPCM is a lossy compression mechanism. There are various flavors of ADPCM. This particular algorithm was suggested by Microsoft; its quality is similar to IMA (Interactive Multimedia Association) ADPCM. MS ADPCM compresses data recorded at various sampling rates. Sound is encoded as a succession of 4-bit nibbles. Each nibble represents the difference between the current sampled signal value and the previous value. The compression ratio obtained is relatively modest: 16-bit data samples encoded as 4-bit differences result in 4:1 compression format.
• A-law
  Compressed WAV format. A-Law (or CCITT standard G.711) is an audio compression scheme common in telephony applications. It is a slight variation of the u-Law compression format, and is found in European systems. This encoding format compresses original 16-bit audio down to 8 bits (for a 2:1 compression ratio) with a dynamic range of about 13-bits. Thus, a-law encoded waveforms have a higher s/n ratio than 8-bit PCM, but at the price of a bit more distortion than the original 16-bit audio. The quality is higher than you would get with 4-bit ADPCM formats. Encoding and decoding is rather fast and generally, widely supported.
• U-law
  Compressed WAV format. u-Law (or CCIUTT standard G.711) is an audio compression scheme and international standard in telephony applications. u-Law is very similar to A-Law, a variation of u-Law found in European systems. This encoding format compresses original 16-bit audio down to 8 bits (for a 2:1 compression ratio) with a dynamic range of about 13-bits. Thus, u-Law encoded waveforms have a higher s/n ratio than 8-bit PCM, but at the price of a bit more distortion than the original 16-bit audio. The quality is higher than you would get with 4-bit ADPCM formats. Encoding and decoding is rather fast and generally, widely supported.

The extension is *.wav.

[Back to the top]

Ogg Vorbis format. Ogg Vorbis is an audio compression format. It is roughly comparable to other formats used to store and play digital music, such as MP3, VQF, AAC, and other digital audio formats.
Ogg Vorbis is a fully open, non-proprietary, patent-and-royalty-free, general-purpose compressed audio format for mid to high quality (8kHz-48.0kHz, 16+ bit, polyphonic) audio and music at fixed and variable bitrates from 16 to 128 kbps/channel.

The file extension(s) is *.ogg.

OGG Website: www.vorbis.com

[Back to the top]

Grossly oversimplified, FLAC is similar to MP3, but lossless, meaning that audio is compressed in FLAC without any loss in quality. This is similar to how Zip works, except with FLAC you will get much better compression because it is designed specifically for audio, and you can play back compressed FLAC files in your favorite player just like you would an MP3 file.

The file extension(s) are *.flac.

FLAC Website: flac.sourceforge.net

[Back to the top]

Monkey's Audio is a fast and easy way to compress digital music.

Unlike traditional methods such as mp3, ogg, or wma that permanently discard quality to save space, Monkey's Audio only makes perfect, bit-for-bit copies of your music and always sounds exactly the same as the original but still saving a lot of space. You can always decompress Monkey's Audio files back to the exactly the same original files. 

The file extension(s) are *.ape or *.mac.

APE Website: www.monkeysaudio.com

[Back to the top]

MusePack is an audio compression format with a strong emphasis on high quality.

It's not lossless, but it is designed for transparency, so that you won't be able to hear differences between the original lossless file and the much smaller MPC.

It is based on the MPEG 1 Layer II algorithms, but has rapidly developed and vastly improved and is now at an advanced stage in which it contains heavily optimized code.

The file extension(s) are *.mpc, *.mp+ or *.mpp.

MPC Website: www.musepack.net

[Back to the top]

Audio Interchange File Format, a format for storing digital audio samples in a file. This standard format for sound files was defined by Apple.

The file extension(s) are *.aif or *.aiff.

[Back to the top]

The True Audio (TTA) codec is a free, simple, realtime lossless audio compressor. Based on adaptive prognostic filters, TTA has compared favorably to a majority of its popular open-source peers. The codec was built to offer adequate compression levels while maintaining high operation speeds.

The TTA lossless audio codec performs lossless compression on multichannel 8, 16 and 24-bit data of WAV audio files. The term "lossless" refers to the fact that such compression results in literally no data or quality loss; when decompressed, the audio file data are bit-identical to those of their originals. Compression ratios achieved by the TTA codec vary, depending on music type, but range from 30% - 70% of the original. The TTA lossless compressed audio format supports both ID3v1 and ID3v2 information tags.

The file extension(s) are *.tta.

TTA Website: www.true-audio.com

[Back to the top]

OptimFROG is a lossless audio compression program. Its main goal is to reduce at maximum the size of audio files, while permitting bit identical restoration for all input. It is similar with the ZIP compression, but it is highly specialized to compress audio data.

OptimFROG obtains asymptotically the best lossless audio compression ratios. It has Windows, Linux, and Mac versions, fully featured input plug-ins for the Windows Media Player, foobar2000, Winamp2/3/5, dBpowerAMP, XMPlay, QCD, and XMMS audio players (with bitstream error resilience, ID3v1.1 and APEv2 read tagging support, ID3v2 compatible), optimal support for all integer PCM wave formats up to 32 bits and an extensible streamable (error tolerant) compressed format. It is also fast, the default mode encodes CD quality audio data at 12.4x real-time and decodes at 17.4x real-time on AMD Athlon XP 1800+ (the fastest mode encodes at 28.1x real-time and decodes at 24.7x real-time). Self-extracting (sfx) archives can also be created with a small overhead of just 54 KB.

The compression ratios which can obtained with OptimFROG are generally ranging from 25% (silent classical music) to 70% (loud rock music) of the original audio file size. This is less compared with around 13% obtained with high quality MP3 files (~176 kb), but you have the great advantage of archiving and listening at perfect copies of your original music.

OFR Website: www.losslessaudio.org

The file extension(s) are *.ofr or *.ofs.

[Back to the top]

Dialogic ADPCM format. The Dialogic ADPCM format is commonly found in telephony applications, and has been optimized for low sample rate voice. It will only save mono 16-bit audio, and like other ADPCM formats, it compresses to 4-bits/sample (for a 4:1 ratio). This format has no header, so any file format with the extension .VOX will be assumed to be in this format.

The file extension(s) are *.vox.

[Back to the top]