Hydrogenaudio Knowledgebase - User contributions [en]

Lossless comparison

2020-06-24T22:40:16Z

12.218.147.66: /* Monkey's Audio (APE) */

The '''lossless comparison page''' aims to gather information about lossless codecs available so users can make an informed decision as to what lossless codec to choose for their needs.

== Introduction ==
Given the enormous number of [[lossless]] audio compressor choices available, it is a very difficult task to choose the one most suited for each person's needs. Some people take into consideration only compression performance when choosing a codec, but as the following table and article shows, there are several other features worth taking into consideration when making a choice.

For example, users wanting good multiplatform compatibility and robustness (e.g., people sharing live recordings) would favour [[WavPack]] or [[FLAC]]. Another user, looking for the very highest compression available, would go with [[OptimFROG]]. Someone wanting portable support would use [[FLAC]] or [[ALAC]], and so on. En fin, this is not a matter worth getting too worked up about. If you later find out the codec you chose isn't the best for your needs, you can just transcompress to another format, without risk of losing quality.

'''Note:''' for latest comparison of lossless compression, scroll down to the [[Lossless comparison#External links|Links section of this page]].

== Comparison Table ==


{| class="wikitable" cellspacing="2"
!width="120px"|'''Features'''
! width="90px" | [[#Free Lossless Audio Codec (FLAC)|FLAC]]
! width="90px" | [[#Apple Lossless Audio Codec (ALAC)|ALAC]]
! width="90px" | [[#WavPack (WV)|WavPack]]
! width="90px" | [[#Tom's_verlustfreier_Audiokompressor (TAK)|TAK]]
! width="90px" | [[#Monkey's_Audio (APE)|Monkey's]]
! width="90px" | [[#Windows Media Audio Lossless (WMAL)|WMAL]]
! width="90px" | [[#OptimFROG (OFR)|OptimFROG]]
! width="90px" | [[#True Audio (TTA)|TTA]]
|- 
| Encoding speed{{ref label|speed|A|A}}
| style="background: #00FF00" | very fast
| style="background: #CCFFCC" | fast
| style="background: #00FF00" | very fast
| style="background: #00FF00" | very fast
| style="background: #CCFFCC" | fast
| style="background: #CCFFCC" | fast
| style="background: #FFCC66" | slow
| style="background: #00FF00" | very fast
|- 
| Decoding speed{{ref label|speed|A|A}}
| style="background: #00FF00" | very fast
| style="background: #CCFFCC" | fast
| style="background: #CCFFCC" | fast
| style="background: #00FF00" | very fast
| style="background: #FFCC66" | slow
| style="background: #FFCC66" | average
| style="background: #FF9900" | very slow
| style="background: #CCFFCC" | fast
|- 
| Compression{{ref label|speed|A|A}}{{ref label|comp|B|B}}
| style="background: #CCFFCC" | 57.0%
| style="background: #CCFFCC" | 57.8%
| style="background: #CCFFCC" | 57.1%
| style="background: #00FF00" | 56.0%
| style="background: #00FF00" | 55.1%
| style="background: #FFCC66" | 58.4%
| style="background: #00FF00" | 54.6%
| style="background: #CCFFCC" | 56.6%
|-
| # presets
| style="background: #CCFFCC" | 9
| style="background: #CCFFCC" | 2
| style="background: #CCFFCC" | > 10
| style="background: #CCFFCC" | > 10
| style="background: #CCFFCC" | 5
| style="background: #CCFFCC" | 1
| style="background: #CCFFCC" | > 10
| style="background: #CCFFCC" | 1
|-
| Error handling{{ref label|error|C|C}}
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #CCFFCC" | yes{{ref label|error_ape|D|D}}
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
|-
| Tagging
| style="background: #00FF00" | Vorbis tags
| style="background: #00FF00" | iTunes
| style="background: #CCFFCC" | ID3/APEv2
| style="background: #00FF00" | APEv2
| style="background: #00FF00" | APEv2
| style="background: #00FF00" | ASF
| style="background: #CCFFCC" | ID3/APEv2
| style="background: #00FF00" | ID3v1/2 or APEv2
|-
| Hardware support
| style="background: #00FF00" | very good
| style="background: #CCFFCC" | good
| style="background: #FFCC66" | limited
| style="background: #FF9900" | no
| style="background: #FFCC66" | limited
| style="background: #FFCC66" | limited
| style="background: #FF9900" | no
| style="background: #FFCC66" | limited
|-
| Software support
| style="background: #00FF00" | very good
| style="background: #CCFFCC" | good
| style="background: #CCFFCC" | good
| style="background: #FFCC66" | average
| style="background: #CCFFCC" | good
| style="background: #CCFFCC" | good
| style="background: #FFCC66" | average
| style="background: #CCFFCC" | good
|-
| Hybrid/lossy
| style="background: #CCFFCC" | [[LossyWAV]]
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #CCFFCC" | [[LossyWAV]]
| style="background: #FF9900" | no
| style="background: #CCFFCC" | [[LossyWAV]]
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
|-
| RIFF chunks
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
|-
| Streaming
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
|-
| Open source
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #FFCC66" | no{{ref label|tak_os|F|F}}
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
|-
| Multichannel
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes{{ref label|multichannel_ape|E|E}}
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
|-
| OS support
| style="background: #00FF00" | All
| style="background: #00FF00" | All
| style="background: #00FF00" | All
| style="background: #CCFFCC" | Win/Wine
| style="background: #00FF00" | All
| style="background: #CCFFCC" | Win/Mac
| style="background: #00FF00" | Win/Mac/Linux
| style="background: #00FF00" | All
|}

{|
|-
|{{note label|speed|A|A}} Speed and Compression are based on '''each encoder's default settings''' and taken from [http://www.audiograaf.nl/downloads.html this comparison].
|-
|{{note label|comp|B|B}} The Compression ratio is compressed size/uncompressed size * 100. So, lower is better.
|-
|{{note label|error|C|C}} Error handling means that a codec can detect a corruption (flipped bit) in a file and warn the user about it, but it will still decode the rest of the file.
|-
|{{note label|error_ape|D|D}} The official Monkey's Audio decoder does not support decoding through errors, but this may be achieved with FFmpeg or Winamp, though likely not, when the "Insane" preset is used.
|-
|{{note label|multichannel_ape|E|E}} Since version 4.86
|-
|{{note label|tak_os|F|F}} Unofficial (but properly working) open source decoder is available as part of ffmpeg
== Codecs ==

These are the most popular lossless codecs, in alphabetical order:

=== Apple Lossless Audio Codec (ALAC) ===
https://alac.macosforge.org/trac

[[ALAC]] is a codec developed by Apple for usage in [[Apple iPod|iPod]] and AirPort Express.

'''ALAC pros'''
* [[Open source]] (encoding and decoding via FFmpeg and [[CueTools|CUETools]], decoding only via [http://craz.net/programs/itunes/alac.html a standalone decoder])
* Fast encoding
* Fast decoding
* Hardware support ([[Apple iPod|iPod]], AirPort Express)
* Software support (iTunes, Quicktime)
* Independent encoder implementation available: ffmpeg
* Streaming support
* Tagging support (QT tags)
* Supports [[multichannel]]. Limited to 8 channels. Only limited set of channels layouts is supported - https://github.com/nu774/qaac/wiki/Multichannel--handling
* Supports [[high resolution]]s
* Used by a few online stores

''' ALAC cons '''
* Limited software support
* No error detection/robustness<ref>[http://www.hydrogenaud.io/forums/index.php?s=&showtopic=33226&view=findpost&p=862031 HA forum post discussing ALAC robustness]</ref>
* No hybrid/lossy mode (and not [[LossyWAV]] compatible)
* Not very efficient

''' ALAC Other features '''
* Fits in the [[MP4]] container

=== Free Lossless Audio Codec (FLAC) ===
https://xiph.org/flac/

[[FLAC]] is a lossless codec developed by Josh Coalson. It's part of the Xiph multimedia portfolio, along with [[Opus]], [[Ogg]], [[Vorbis]], [[Speex]] and [[Theora]].

''' FLAC pros '''
* [[Open source]]
* Very fast decoding
* Very fast encoding
* Very good hardware support (Android, Marantz, Sonos, [http://xiph.org/flac/links.html many others])
* Very good software support
* Independent encoder implementations available: flake/ffmpeg, FLACCL
* Error robustness
* Streaming support
* Supports [[multichannel]]. Limited to 8 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is supported but support is not included in official specification. With reference encoder undocumented option --channel-map=none is needed to encode some non-standard layouts (e.g. 4.1; FL,FR,FC,BC), but no special options are needed with ffmpeg's encoder.
* Supports [[high resolution]]s
* Tagging support (FLAC tags)
* Supports [[RIFF]] chunks
* Pipe support
* Used by a few [http://xiph.org/flac/links.html#music online stores]

''' FLAC cons '''
* No hybrid/lossy mode (but is [[LossyWAV]] compatible)
* Does not handle 32-bit float and there is no encoder that can render to 32-bit integer

''' FLAC Other features '''
* Supports embedded CUE sheets (with [http://flac.sourceforge.net/faq.html#general__no_cuesheet_tags limitations])
* Includes MD5 hashes for quick integrity checking as standard
* Fits the [[Ogg]] and [[Matroska]] containers

=== Monkey's Audio (APE) ===
https://www.monkeysaudio.com/

[[Monkey's Audio]] is a very efficient lossless compressor developed by Matt Ashland.

''' APE pros '''
* High compression
* Fast encoding
* Good software support
* Supports [[multichannel]] (since version 4.86). Limited to 8 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is supported
* Supports [[high resolution]]s
* Simple and user friendly. Official GUI provided.
* Java version (multiplatform)
* Error robustness/decoding up to -c3000 (High compression)<ref>http://www.hydrogenaud.io/forums/index.php?showtopic=98984&st=0&p=821420&#entry821420</ref>
* Tagging support ([[ID3v1]], [[APE tags]])
* Supports [[RIFF]] chunks (only in the GUI encoder)
* Pipe support (only in a [http://www.etree.org/shnutils/shntool/ special] version)

''' APE cons '''
* Problematic license (Source available, but with no modification or redistribution rights. Encourages violating the GNU GPL license of other programs.)
* Slow decoding
* No hybrid/lossy mode (and not [[LossyWAV]] compatible)
* Limited hardware support (Rockbox, some Cowon players); poor battery life due to complicated decoding (see [http://www.rockbox.org/wiki/SoundCodecMonkeysAudio MP3 player benchmarks])
* Higher compression levels are extremely CPU intensive

''' APE Other features '''
* Includes MD5 hashes for quick integrity checking
* Supports APL image link files (similar to CUE sheets)

=== OptimFROG (OFR) ===
http://www.losslessaudio.org/

[[OptimFROG]] is a lossless format developed by Florin Ghido to become the champion in audio compression.

''' OFR pros '''
* Very high compression
* Good software support
* Error robustness
* Streaming support
* Supports [[high resolution]]s
* Hybrid/lossy mode
* Tagging support ([[ID3]], [[APE tags]])
* Supports [[RIFF]] chunks

''' OFR cons '''
* Closed source
* No [[multichannel]] audio support
* No hardware support
* Very slow decoding
* Slow encoding
* More than one tagging method allowed (ambiguity possible)

''' OFR Other features '''
* Supports 32bit float streams
* Includes MD5 hashes for quick integrity checking

=== Tom's verlustfreier Audiokompressor (TAK) ===
http://www.thbeck.de/Tak/Tak.html

[[TAK]] is a lossless codec developed by Thomas Becker.

''' TAK pros '''
* Very fast decoding
* Very fast encoding
* Very high efficiency
* Error robust
* Supports [[multichannel]]. Limited to 6 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is supported
* Supports [[high resolution]]s
* Tagging support
* Supports RIFF chunks
* Pipe support
* Streamable

''' TAK cons '''
* Closed source (but unofficial open source decoder is available as part of ffmpeg)
* No hybrid/lossy mode (but is [[LossyWAV]] compatible)
* No hardware support
* Average software support
* Doesn't support Unicode (yet)

''' TAK Other features '''
* Optional MD5 checksum

=== True Audio (TTA) ===
http://tta.tausoft.org/

[[TTA]] is a lossless codec developed by a international team of programmers.

''' TTA pros '''
* [[Open source]]
* Supports [[multichannel]]. Reference encoder/decoder is limited to 6 channels. ffmpeg's encoder/decoder is limited to 16 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is '''not''' supported
* Supports [[high resolution]]s
* Tagging support ([[ID3]]v1, ID3v2 or [[APEv2]])
* Embedded CUE sheets support
* Error robustness
* Pipe support
* Average compression
* Fast encoding/decoding
* Symmetric algorithm
* Ultra low latency

''' TTA cons '''
* No hybrid/lossy mode
* Doesn't support [[RIFF]] chunks
* Limited hardware support

''' TTA Other features '''
* Fits the [[Matroska]] container
* Password protection

=== WavPack (WV) ===
http://www.wavpack.com/

[[WavPack]] is a fast and featureful lossless codec developed by David Bryant.

''' WV pros '''
* [[Open source]]
* Fast decoding
* Very fast encoding
* Good efficiency
* Error robustness
* Streaming support
* Supports [[multichannel]]. Limited to 255 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is supported
* Supports [[high resolution]]s
* Hybrid/lossy mode
* Tagging support ([[ID3v1]], [[APE tags]])
* Supports [[RIFF]] chunks
* Ability to create self extracting files for Win32 platform
* Pipe support
* Good software support
* Works with Android (Through third party software, such as VLC.)
* Independent encoder implementation available. (FFmpeg WavPack)

''' WV cons '''
* Limited hardware player support ([http://www.rockbox.org/ RockBox])
* More than one tagging method allowed (Ambiguity possible, but unlikely as APEv2 tags have been the preferred method for quite some time.)

''' WV Other features '''
* Can compress the Direct-Stream Digital (DSD) audio recording format
* Supports 32bit float streams
* Supports embedded CUE sheets
* Accept audio files bigger than 4GB
* Includes MD5 hashes for quick integrity checking
* Can encode in both symmetrical and asymmetrical modes.
* Fits the [[Matroska]] container

=== Windows Media Audio Lossless (WMAL) ===
https://msdn.microsoft.com/en-us/library/ff819508(v=vs.85).aspx

WMA Lossless is the lossless codec developed by Microsoft to be featured in their Windows Media codec portfolio.

''' WMAL pros '''
* Streaming support
* Supports [[multichannel]] audio and [[high resolution]]s.
* Tagging support (proprietary)
* Pipe support

''' WMAL cons '''
* Limited hardware support (Microsoft Zune, Toshiba Gigabeat S and V. Both discontinued and obsolete. Rockbox, for 16-bit stereo files only.)
* Limited software support outside of the Microsoft Windows operating system.
* Extremely low efficiency
* Closed source
* No hybrid/lossy mode (but is [[LossyWAV]] compatible)
* Doesn't support [[RIFF]] chunks

''' WMAL Other features '''
* Fits the [[ASF]] container

=== Other Formats ===
Aside from the formats mentioned above, there are in fact quite a lot of other lossless formats. To keep the table and list brief and readable, a few formats have not been mentioned.

====DTS-HD Master Audio====
Similar to the MPEG-4 SLS format, this format has a core track in an older, more widely supported format, DTS. This core lossy track is made lossless by a secondary track with correction data. It is an optional codec in Blu-ray implementations. Its main use is surround sound encoding, and as is the case with MLP, the price of the encoder ensures it is only used in mastering of Blu-ray discs.

====LA====
http://www.lossless-audio.com/

LA features an extremely high compression (on par with OptimFrog highest modes, but a bit faster), but it hasn't been updated for more than 10 years. Furthermore, backward compatibility is not guaranteed, so using it for archiving might pose a few problems. It isn't able to cope with file corruption either, software support is very limited and isn't open source.

====MLP/Dolby TrueHD====
The [[MLP|MLP codec]] (of which the mathematical basis was used in Dolby TrueHD) is the codec used for DVD-Audio. It was mandatory in any HD-DVD implementation and optional for Blu-Ray in its Dolby TrueHD form. It is known to support the 'wasted bits' scheme used in LossyWAV. As encoders are very expensive, its use outside DVD/Blu-ray mastering environments is non-existent. Its main use is encoding surround sound data.

====MPEG-4 ALS====
MPEG-4 ALS is the successor to LPAC, which it was based on. It has been as a ISO standard and there is a reference encoder/decoder, but like TTA, it does not have features that make it stand out from other codecs, nor backing by a large organisation, so it hasn't much software and no hardware support.

====MPEG-4 SLS====
MPEG-4 SLS is a special codec, having a AAC core track and a 'correction track'. Also known as HD-AAC, SLS stands for Scalable to Lossless. However, there is to date still no affordable software to play, encode or decode (the lossless part of) SLS files.

====Shorten====
http://www.etree.org/shncom.html

Shorten was one of the first widely-used lossless formats, and it still occasionally found on the internet, especially in archives, for example etree.org. It is quite fast in both encoding and decoding, but doesn't compress very much. Furthermore, seeking has a troubled past as well as tagging. It is considered obsolete.

====Real Lossless====
Part of the Real codec suite, Real Lossless too hasn't any very special features that make it stand out. Just like WMA Lossless and Apple Lossless, it was created to fit in a codec suite, but unlike WMA Lossless and Apple Lossless, there is no hardware support and software support is limited. Compression is on par with most other codecs, but it is rather slow to encode.

====Oddball formats====
There are a few archaic formats of which encoders and decoders are hard to get by. Most of those would have disappeared by now, but some of them are being preserved for posterity at [[User:Rjamorim|rjamorim]]'s

* Advanced Digital Audio (ADA)
* [http://www.logarithmic.net/pfh/bonk Bonk]
* AudioZip
* Dakx WAV
* Entis Lab MIO
* LiteWave
* [http://www.nue.tu-berlin.de/menue/mitarbeiter/ehemalige_mitarbeiter/tilman_liebchen/lpac_-_lossless_audio_codec_for_windows_and_linux/ LPAC]
* Marian's a-Pac
* [http://mp3hd-toolkit.soft32.com/ mp3HD (MPEG-1 Audio Layer III HD)]
* Pegasus SPS
* [http://www.free-codecs.com/download/rk_audio_compressor.htm RK Audio (RKAU)]
* Ogg Squish/Tarkin
* Sonarc
* VocPack
* [http://www.firstpr.com.au/audiocomp/lossless/wavarc/ WavArc]
* [http://www.firstpr.com.au/audiocomp/lossless/WaveZip/ WaveZip]/MUSICompress

== See also ==
* [[Lossless]]

== External links ==
''' Other lossless compressions comparisons '''
''Sorted based on last '''update''' date.''

* [http://www.audiograaf.nl/downloads.html Martijn van Beurden's comparison] - tries to compare all codecs and settings with a balanced pool of music (last updated 2015-01-05)
* [http://www.squeezechart.com/audio.html Squeezechart audio] - tests as much codecs as possible, but not all their settings and with a limited test corpus (last updated 2013-10-31)
* <s>[http://synthetic-soul.co.uk/comparison/lossless/index.asp Synthetic Soul's comparison] (last update 2007-07-28)</s>
* <s>Johan De Bock's speed oriented comparison</s> - best choices speedwise are indicated in green, mostly electronic music (last updated 2006-07-22)
* <s>Hans Heijden's</s> -- used as reference to build the table (last updated 2006-07-07)
* <s>Josef Pohm's comparison, hosted by Synthetic Soul</s> (last update 2006-05-29)
* [http://www.bobulous.org.uk/misc/lossless_audio_2006.html Bobulous' lossless audio comparison] — a look at six lossless formats in terms of speed and file size (last updated 2006-05-22)
* <s>Jhan De Bock's size oriented comparison</s> - aimed only at the maximum compression setting for each codec (based on a somewhat limited set of samples, however) (last updated 2006-05-19)
* <s>Gruboolez'</s> -- comparing only classical music (last updated 2005-02-27)
* <s>Speek's</s> (last updated 2005-02-07)
*[http://www.firstpr.com.au/audiocomp/lossless/ Lossless Compression of Audio] Much information about oddball formats including comparison of them. (last updated 2005-10-21)

''' More on lossless compressions '''
* [http://web.archive.org/web/20080731103800/http://www.losslessaudioblog.com/ The Lossless Audio Blog], retrieved from archive.org - by windmiller, is a reliable and complete source of news about lossless compression.
* Go to the [http://www.hydrogenaudio.org/forums/index.php?showtopic=33226 Hydrogenaudio thread] to discuss this article.

== References ==

<references/>
[[Category:Guides]]

Lossless comparison

2020-06-24T22:38:55Z

12.218.147.66: /* WavPack (WV) */

The '''lossless comparison page''' aims to gather information about lossless codecs available so users can make an informed decision as to what lossless codec to choose for their needs.

== Introduction ==
Given the enormous number of [[lossless]] audio compressor choices available, it is a very difficult task to choose the one most suited for each person's needs. Some people take into consideration only compression performance when choosing a codec, but as the following table and article shows, there are several other features worth taking into consideration when making a choice.

For example, users wanting good multiplatform compatibility and robustness (e.g., people sharing live recordings) would favour [[WavPack]] or [[FLAC]]. Another user, looking for the very highest compression available, would go with [[OptimFROG]]. Someone wanting portable support would use [[FLAC]] or [[ALAC]], and so on. En fin, this is not a matter worth getting too worked up about. If you later find out the codec you chose isn't the best for your needs, you can just transcompress to another format, without risk of losing quality.

'''Note:''' for latest comparison of lossless compression, scroll down to the [[Lossless comparison#External links|Links section of this page]].

== Comparison Table ==


{| class="wikitable" cellspacing="2"
!width="120px"|'''Features'''
! width="90px" | [[#Free Lossless Audio Codec (FLAC)|FLAC]]
! width="90px" | [[#Apple Lossless Audio Codec (ALAC)|ALAC]]
! width="90px" | [[#WavPack (WV)|WavPack]]
! width="90px" | [[#Tom's_verlustfreier_Audiokompressor (TAK)|TAK]]
! width="90px" | [[#Monkey's_Audio (APE)|Monkey's]]
! width="90px" | [[#Windows Media Audio Lossless (WMAL)|WMAL]]
! width="90px" | [[#OptimFROG (OFR)|OptimFROG]]
! width="90px" | [[#True Audio (TTA)|TTA]]
|- 
| Encoding speed{{ref label|speed|A|A}}
| style="background: #00FF00" | very fast
| style="background: #CCFFCC" | fast
| style="background: #00FF00" | very fast
| style="background: #00FF00" | very fast
| style="background: #CCFFCC" | fast
| style="background: #CCFFCC" | fast
| style="background: #FFCC66" | slow
| style="background: #00FF00" | very fast
|- 
| Decoding speed{{ref label|speed|A|A}}
| style="background: #00FF00" | very fast
| style="background: #CCFFCC" | fast
| style="background: #CCFFCC" | fast
| style="background: #00FF00" | very fast
| style="background: #FFCC66" | slow
| style="background: #FFCC66" | average
| style="background: #FF9900" | very slow
| style="background: #CCFFCC" | fast
|- 
| Compression{{ref label|speed|A|A}}{{ref label|comp|B|B}}
| style="background: #CCFFCC" | 57.0%
| style="background: #CCFFCC" | 57.8%
| style="background: #CCFFCC" | 57.1%
| style="background: #00FF00" | 56.0%
| style="background: #00FF00" | 55.1%
| style="background: #FFCC66" | 58.4%
| style="background: #00FF00" | 54.6%
| style="background: #CCFFCC" | 56.6%
|-
| # presets
| style="background: #CCFFCC" | 9
| style="background: #CCFFCC" | 2
| style="background: #CCFFCC" | > 10
| style="background: #CCFFCC" | > 10
| style="background: #CCFFCC" | 5
| style="background: #CCFFCC" | 1
| style="background: #CCFFCC" | > 10
| style="background: #CCFFCC" | 1
|-
| Error handling{{ref label|error|C|C}}
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #CCFFCC" | yes{{ref label|error_ape|D|D}}
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
|-
| Tagging
| style="background: #00FF00" | Vorbis tags
| style="background: #00FF00" | iTunes
| style="background: #CCFFCC" | ID3/APEv2
| style="background: #00FF00" | APEv2
| style="background: #00FF00" | APEv2
| style="background: #00FF00" | ASF
| style="background: #CCFFCC" | ID3/APEv2
| style="background: #00FF00" | ID3v1/2 or APEv2
|-
| Hardware support
| style="background: #00FF00" | very good
| style="background: #CCFFCC" | good
| style="background: #FFCC66" | limited
| style="background: #FF9900" | no
| style="background: #FFCC66" | limited
| style="background: #FFCC66" | limited
| style="background: #FF9900" | no
| style="background: #FFCC66" | limited
|-
| Software support
| style="background: #00FF00" | very good
| style="background: #CCFFCC" | good
| style="background: #CCFFCC" | good
| style="background: #FFCC66" | average
| style="background: #CCFFCC" | good
| style="background: #CCFFCC" | good
| style="background: #FFCC66" | average
| style="background: #CCFFCC" | good
|-
| Hybrid/lossy
| style="background: #CCFFCC" | [[LossyWAV]]
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #CCFFCC" | [[LossyWAV]]
| style="background: #FF9900" | no
| style="background: #CCFFCC" | [[LossyWAV]]
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
|-
| RIFF chunks
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
|-
| Streaming
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
|-
| Open source
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #FFCC66" | no{{ref label|tak_os|F|F}}
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
|-
| Multichannel
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes{{ref label|multichannel_ape|E|E}}
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
|-
| OS support
| style="background: #00FF00" | All
| style="background: #00FF00" | All
| style="background: #00FF00" | All
| style="background: #CCFFCC" | Win/Wine
| style="background: #00FF00" | All
| style="background: #CCFFCC" | Win/Mac
| style="background: #00FF00" | Win/Mac/Linux
| style="background: #00FF00" | All
|}

{|
|-
|{{note label|speed|A|A}} Speed and Compression are based on '''each encoder's default settings''' and taken from [http://www.audiograaf.nl/downloads.html this comparison].
|-
|{{note label|comp|B|B}} The Compression ratio is compressed size/uncompressed size * 100. So, lower is better.
|-
|{{note label|error|C|C}} Error handling means that a codec can detect a corruption (flipped bit) in a file and warn the user about it, but it will still decode the rest of the file.
|-
|{{note label|error_ape|D|D}} The official Monkey's Audio decoder does not support decoding through errors, but this may be achieved with FFmpeg or Winamp, though likely not, when the "Insane" preset is used.
|-
|{{note label|multichannel_ape|E|E}} Since version 4.86
|-
|{{note label|tak_os|F|F}} Unofficial (but properly working) open source decoder is available as part of ffmpeg
== Codecs ==

These are the most popular lossless codecs, in alphabetical order:

=== Apple Lossless Audio Codec (ALAC) ===
https://alac.macosforge.org/trac

[[ALAC]] is a codec developed by Apple for usage in [[Apple iPod|iPod]] and AirPort Express.

'''ALAC pros'''
* [[Open source]] (encoding and decoding via FFmpeg and [[CueTools|CUETools]], decoding only via [http://craz.net/programs/itunes/alac.html a standalone decoder])
* Fast encoding
* Fast decoding
* Hardware support ([[Apple iPod|iPod]], AirPort Express)
* Software support (iTunes, Quicktime)
* Independent encoder implementation available: ffmpeg
* Streaming support
* Tagging support (QT tags)
* Supports [[multichannel]]. Limited to 8 channels. Only limited set of channels layouts is supported - https://github.com/nu774/qaac/wiki/Multichannel--handling
* Supports [[high resolution]]s
* Used by a few online stores

''' ALAC cons '''
* Limited software support
* No error detection/robustness<ref>[http://www.hydrogenaud.io/forums/index.php?s=&showtopic=33226&view=findpost&p=862031 HA forum post discussing ALAC robustness]</ref>
* No hybrid/lossy mode (and not [[LossyWAV]] compatible)
* Not very efficient

''' ALAC Other features '''
* Fits in the [[MP4]] container

=== Free Lossless Audio Codec (FLAC) ===
https://xiph.org/flac/

[[FLAC]] is a lossless codec developed by Josh Coalson. It's part of the Xiph multimedia portfolio, along with [[Opus]], [[Ogg]], [[Vorbis]], [[Speex]] and [[Theora]].

''' FLAC pros '''
* [[Open source]]
* Very fast decoding
* Very fast encoding
* Very good hardware support (Android, Marantz, Sonos, [http://xiph.org/flac/links.html many others])
* Very good software support
* Independent encoder implementations available: flake/ffmpeg, FLACCL
* Error robustness
* Streaming support
* Supports [[multichannel]]. Limited to 8 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is supported but support is not included in official specification. With reference encoder undocumented option --channel-map=none is needed to encode some non-standard layouts (e.g. 4.1; FL,FR,FC,BC), but no special options are needed with ffmpeg's encoder.
* Supports [[high resolution]]s
* Tagging support (FLAC tags)
* Supports [[RIFF]] chunks
* Pipe support
* Used by a few [http://xiph.org/flac/links.html#music online stores]

''' FLAC cons '''
* No hybrid/lossy mode (but is [[LossyWAV]] compatible)
* Does not handle 32-bit float and there is no encoder that can render to 32-bit integer

''' FLAC Other features '''
* Supports embedded CUE sheets (with [http://flac.sourceforge.net/faq.html#general__no_cuesheet_tags limitations])
* Includes MD5 hashes for quick integrity checking as standard
* Fits the [[Ogg]] and [[Matroska]] containers

=== Monkey's Audio (APE) ===
https://www.monkeysaudio.com/

[[Monkey's Audio]] is a very efficient lossless compressor developed by Matt Ashland.

''' APE pros '''
* High compression
* Fast encoding
* Good software support
* Supports [[multichannel]] (since version 4.86). Limited to 8 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is supported
* Supports [[high resolution]]s
* Simple and user friendly. Official GUI provided.
* Java version (multiplatform)
* Error robustness/decoding up to -c3000 (High compression)<ref>http://www.hydrogenaud.io/forums/index.php?showtopic=98984&st=0&p=821420&#entry821420</ref>
* Tagging support ([[ID3v1]], [[APE tags]])
* Supports [[RIFF]] chunks (only in the GUI encoder)
* Pipe support (only in a [http://www.etree.org/shnutils/shntool/ special] version)

''' APE cons '''
* Problematic license (source provided, no modification or redistribution rights)
* Slow decoding
* No hybrid/lossy mode (and not [[LossyWAV]] compatible)
* Limited hardware support (Rockbox, some Cowon players); poor battery life due to complicated decoding (see [http://www.rockbox.org/wiki/SoundCodecMonkeysAudio MP3 player benchmarks])
* Higher compression levels are extremely CPU intensive

''' APE Other features '''
* Includes MD5 hashes for quick integrity checking
* Supports APL image link files (similar to CUE sheets)

=== OptimFROG (OFR) ===
http://www.losslessaudio.org/

[[OptimFROG]] is a lossless format developed by Florin Ghido to become the champion in audio compression.

''' OFR pros '''
* Very high compression
* Good software support
* Error robustness
* Streaming support
* Supports [[high resolution]]s
* Hybrid/lossy mode
* Tagging support ([[ID3]], [[APE tags]])
* Supports [[RIFF]] chunks

''' OFR cons '''
* Closed source
* No [[multichannel]] audio support
* No hardware support
* Very slow decoding
* Slow encoding
* More than one tagging method allowed (ambiguity possible)

''' OFR Other features '''
* Supports 32bit float streams
* Includes MD5 hashes for quick integrity checking

=== Tom's verlustfreier Audiokompressor (TAK) ===
http://www.thbeck.de/Tak/Tak.html

[[TAK]] is a lossless codec developed by Thomas Becker.

''' TAK pros '''
* Very fast decoding
* Very fast encoding
* Very high efficiency
* Error robust
* Supports [[multichannel]]. Limited to 6 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is supported
* Supports [[high resolution]]s
* Tagging support
* Supports RIFF chunks
* Pipe support
* Streamable

''' TAK cons '''
* Closed source (but unofficial open source decoder is available as part of ffmpeg)
* No hybrid/lossy mode (but is [[LossyWAV]] compatible)
* No hardware support
* Average software support
* Doesn't support Unicode (yet)

''' TAK Other features '''
* Optional MD5 checksum

=== True Audio (TTA) ===
http://tta.tausoft.org/

[[TTA]] is a lossless codec developed by a international team of programmers.

''' TTA pros '''
* [[Open source]]
* Supports [[multichannel]]. Reference encoder/decoder is limited to 6 channels. ffmpeg's encoder/decoder is limited to 16 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is '''not''' supported
* Supports [[high resolution]]s
* Tagging support ([[ID3]]v1, ID3v2 or [[APEv2]])
* Embedded CUE sheets support
* Error robustness
* Pipe support
* Average compression
* Fast encoding/decoding
* Symmetric algorithm
* Ultra low latency

''' TTA cons '''
* No hybrid/lossy mode
* Doesn't support [[RIFF]] chunks
* Limited hardware support

''' TTA Other features '''
* Fits the [[Matroska]] container
* Password protection

=== WavPack (WV) ===
http://www.wavpack.com/

[[WavPack]] is a fast and featureful lossless codec developed by David Bryant.

''' WV pros '''
* [[Open source]]
* Fast decoding
* Very fast encoding
* Good efficiency
* Error robustness
* Streaming support
* Supports [[multichannel]]. Limited to 255 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is supported
* Supports [[high resolution]]s
* Hybrid/lossy mode
* Tagging support ([[ID3v1]], [[APE tags]])
* Supports [[RIFF]] chunks
* Ability to create self extracting files for Win32 platform
* Pipe support
* Good software support
* Works with Android (Through third party software, such as VLC.)
* Independent encoder implementation available. (FFmpeg WavPack)

''' WV cons '''
* Limited hardware player support ([http://www.rockbox.org/ RockBox])
* More than one tagging method allowed (Ambiguity possible, but unlikely as APEv2 tags have been the preferred method for quite some time.)

''' WV Other features '''
* Can compress the Direct-Stream Digital (DSD) audio recording format
* Supports 32bit float streams
* Supports embedded CUE sheets
* Accept audio files bigger than 4GB
* Includes MD5 hashes for quick integrity checking
* Can encode in both symmetrical and asymmetrical modes.
* Fits the [[Matroska]] container

=== Windows Media Audio Lossless (WMAL) ===
https://msdn.microsoft.com/en-us/library/ff819508(v=vs.85).aspx

WMA Lossless is the lossless codec developed by Microsoft to be featured in their Windows Media codec portfolio.

''' WMAL pros '''
* Streaming support
* Supports [[multichannel]] audio and [[high resolution]]s.
* Tagging support (proprietary)
* Pipe support

''' WMAL cons '''
* Limited hardware support (Microsoft Zune, Toshiba Gigabeat S and V. Both discontinued and obsolete. Rockbox, for 16-bit stereo files only.)
* Limited software support outside of the Microsoft Windows operating system.
* Extremely low efficiency
* Closed source
* No hybrid/lossy mode (but is [[LossyWAV]] compatible)
* Doesn't support [[RIFF]] chunks

''' WMAL Other features '''
* Fits the [[ASF]] container

=== Other Formats ===
Aside from the formats mentioned above, there are in fact quite a lot of other lossless formats. To keep the table and list brief and readable, a few formats have not been mentioned.

====DTS-HD Master Audio====
Similar to the MPEG-4 SLS format, this format has a core track in an older, more widely supported format, DTS. This core lossy track is made lossless by a secondary track with correction data. It is an optional codec in Blu-ray implementations. Its main use is surround sound encoding, and as is the case with MLP, the price of the encoder ensures it is only used in mastering of Blu-ray discs.

====LA====
http://www.lossless-audio.com/

LA features an extremely high compression (on par with OptimFrog highest modes, but a bit faster), but it hasn't been updated for more than 10 years. Furthermore, backward compatibility is not guaranteed, so using it for archiving might pose a few problems. It isn't able to cope with file corruption either, software support is very limited and isn't open source.

====MLP/Dolby TrueHD====
The [[MLP|MLP codec]] (of which the mathematical basis was used in Dolby TrueHD) is the codec used for DVD-Audio. It was mandatory in any HD-DVD implementation and optional for Blu-Ray in its Dolby TrueHD form. It is known to support the 'wasted bits' scheme used in LossyWAV. As encoders are very expensive, its use outside DVD/Blu-ray mastering environments is non-existent. Its main use is encoding surround sound data.

====MPEG-4 ALS====
MPEG-4 ALS is the successor to LPAC, which it was based on. It has been as a ISO standard and there is a reference encoder/decoder, but like TTA, it does not have features that make it stand out from other codecs, nor backing by a large organisation, so it hasn't much software and no hardware support.

====MPEG-4 SLS====
MPEG-4 SLS is a special codec, having a AAC core track and a 'correction track'. Also known as HD-AAC, SLS stands for Scalable to Lossless. However, there is to date still no affordable software to play, encode or decode (the lossless part of) SLS files.

====Shorten====
http://www.etree.org/shncom.html

Shorten was one of the first widely-used lossless formats, and it still occasionally found on the internet, especially in archives, for example etree.org. It is quite fast in both encoding and decoding, but doesn't compress very much. Furthermore, seeking has a troubled past as well as tagging. It is considered obsolete.

====Real Lossless====
Part of the Real codec suite, Real Lossless too hasn't any very special features that make it stand out. Just like WMA Lossless and Apple Lossless, it was created to fit in a codec suite, but unlike WMA Lossless and Apple Lossless, there is no hardware support and software support is limited. Compression is on par with most other codecs, but it is rather slow to encode.

====Oddball formats====
There are a few archaic formats of which encoders and decoders are hard to get by. Most of those would have disappeared by now, but some of them are being preserved for posterity at [[User:Rjamorim|rjamorim]]'s

* Advanced Digital Audio (ADA)
* [http://www.logarithmic.net/pfh/bonk Bonk]
* AudioZip
* Dakx WAV
* Entis Lab MIO
* LiteWave
* [http://www.nue.tu-berlin.de/menue/mitarbeiter/ehemalige_mitarbeiter/tilman_liebchen/lpac_-_lossless_audio_codec_for_windows_and_linux/ LPAC]
* Marian's a-Pac
* [http://mp3hd-toolkit.soft32.com/ mp3HD (MPEG-1 Audio Layer III HD)]
* Pegasus SPS
* [http://www.free-codecs.com/download/rk_audio_compressor.htm RK Audio (RKAU)]
* Ogg Squish/Tarkin
* Sonarc
* VocPack
* [http://www.firstpr.com.au/audiocomp/lossless/wavarc/ WavArc]
* [http://www.firstpr.com.au/audiocomp/lossless/WaveZip/ WaveZip]/MUSICompress

== See also ==
* [[Lossless]]

== External links ==
''' Other lossless compressions comparisons '''
''Sorted based on last '''update''' date.''

* [http://www.audiograaf.nl/downloads.html Martijn van Beurden's comparison] - tries to compare all codecs and settings with a balanced pool of music (last updated 2015-01-05)
* [http://www.squeezechart.com/audio.html Squeezechart audio] - tests as much codecs as possible, but not all their settings and with a limited test corpus (last updated 2013-10-31)
* <s>[http://synthetic-soul.co.uk/comparison/lossless/index.asp Synthetic Soul's comparison] (last update 2007-07-28)</s>
* <s>Johan De Bock's speed oriented comparison</s> - best choices speedwise are indicated in green, mostly electronic music (last updated 2006-07-22)
* <s>Hans Heijden's</s> -- used as reference to build the table (last updated 2006-07-07)
* <s>Josef Pohm's comparison, hosted by Synthetic Soul</s> (last update 2006-05-29)
* [http://www.bobulous.org.uk/misc/lossless_audio_2006.html Bobulous' lossless audio comparison] — a look at six lossless formats in terms of speed and file size (last updated 2006-05-22)
* <s>Jhan De Bock's size oriented comparison</s> - aimed only at the maximum compression setting for each codec (based on a somewhat limited set of samples, however) (last updated 2006-05-19)
* <s>Gruboolez'</s> -- comparing only classical music (last updated 2005-02-27)
* <s>Speek's</s> (last updated 2005-02-07)
*[http://www.firstpr.com.au/audiocomp/lossless/ Lossless Compression of Audio] Much information about oddball formats including comparison of them. (last updated 2005-10-21)

''' More on lossless compressions '''
* [http://web.archive.org/web/20080731103800/http://www.losslessaudioblog.com/ The Lossless Audio Blog], retrieved from archive.org - by windmiller, is a reliable and complete source of news about lossless compression.
* Go to the [http://www.hydrogenaudio.org/forums/index.php?showtopic=33226 Hydrogenaudio thread] to discuss this article.

== References ==

<references/>
[[Category:Guides]]

Lossless comparison

2020-06-24T22:36:55Z

12.218.147.66: /* WavPack (WV) */

The '''lossless comparison page''' aims to gather information about lossless codecs available so users can make an informed decision as to what lossless codec to choose for their needs.

== Introduction ==
Given the enormous number of [[lossless]] audio compressor choices available, it is a very difficult task to choose the one most suited for each person's needs. Some people take into consideration only compression performance when choosing a codec, but as the following table and article shows, there are several other features worth taking into consideration when making a choice.

For example, users wanting good multiplatform compatibility and robustness (e.g., people sharing live recordings) would favour [[WavPack]] or [[FLAC]]. Another user, looking for the very highest compression available, would go with [[OptimFROG]]. Someone wanting portable support would use [[FLAC]] or [[ALAC]], and so on. En fin, this is not a matter worth getting too worked up about. If you later find out the codec you chose isn't the best for your needs, you can just transcompress to another format, without risk of losing quality.

'''Note:''' for latest comparison of lossless compression, scroll down to the [[Lossless comparison#External links|Links section of this page]].

== Comparison Table ==


{| class="wikitable" cellspacing="2"
!width="120px"|'''Features'''
! width="90px" | [[#Free Lossless Audio Codec (FLAC)|FLAC]]
! width="90px" | [[#Apple Lossless Audio Codec (ALAC)|ALAC]]
! width="90px" | [[#WavPack (WV)|WavPack]]
! width="90px" | [[#Tom's_verlustfreier_Audiokompressor (TAK)|TAK]]
! width="90px" | [[#Monkey's_Audio (APE)|Monkey's]]
! width="90px" | [[#Windows Media Audio Lossless (WMAL)|WMAL]]
! width="90px" | [[#OptimFROG (OFR)|OptimFROG]]
! width="90px" | [[#True Audio (TTA)|TTA]]
|- 
| Encoding speed{{ref label|speed|A|A}}
| style="background: #00FF00" | very fast
| style="background: #CCFFCC" | fast
| style="background: #00FF00" | very fast
| style="background: #00FF00" | very fast
| style="background: #CCFFCC" | fast
| style="background: #CCFFCC" | fast
| style="background: #FFCC66" | slow
| style="background: #00FF00" | very fast
|- 
| Decoding speed{{ref label|speed|A|A}}
| style="background: #00FF00" | very fast
| style="background: #CCFFCC" | fast
| style="background: #CCFFCC" | fast
| style="background: #00FF00" | very fast
| style="background: #FFCC66" | slow
| style="background: #FFCC66" | average
| style="background: #FF9900" | very slow
| style="background: #CCFFCC" | fast
|- 
| Compression{{ref label|speed|A|A}}{{ref label|comp|B|B}}
| style="background: #CCFFCC" | 57.0%
| style="background: #CCFFCC" | 57.8%
| style="background: #CCFFCC" | 57.1%
| style="background: #00FF00" | 56.0%
| style="background: #00FF00" | 55.1%
| style="background: #FFCC66" | 58.4%
| style="background: #00FF00" | 54.6%
| style="background: #CCFFCC" | 56.6%
|-
| # presets
| style="background: #CCFFCC" | 9
| style="background: #CCFFCC" | 2
| style="background: #CCFFCC" | > 10
| style="background: #CCFFCC" | > 10
| style="background: #CCFFCC" | 5
| style="background: #CCFFCC" | 1
| style="background: #CCFFCC" | > 10
| style="background: #CCFFCC" | 1
|-
| Error handling{{ref label|error|C|C}}
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #CCFFCC" | yes{{ref label|error_ape|D|D}}
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
|-
| Tagging
| style="background: #00FF00" | Vorbis tags
| style="background: #00FF00" | iTunes
| style="background: #CCFFCC" | ID3/APEv2
| style="background: #00FF00" | APEv2
| style="background: #00FF00" | APEv2
| style="background: #00FF00" | ASF
| style="background: #CCFFCC" | ID3/APEv2
| style="background: #00FF00" | ID3v1/2 or APEv2
|-
| Hardware support
| style="background: #00FF00" | very good
| style="background: #CCFFCC" | good
| style="background: #FFCC66" | limited
| style="background: #FF9900" | no
| style="background: #FFCC66" | limited
| style="background: #FFCC66" | limited
| style="background: #FF9900" | no
| style="background: #FFCC66" | limited
|-
| Software support
| style="background: #00FF00" | very good
| style="background: #CCFFCC" | good
| style="background: #CCFFCC" | good
| style="background: #FFCC66" | average
| style="background: #CCFFCC" | good
| style="background: #CCFFCC" | good
| style="background: #FFCC66" | average
| style="background: #CCFFCC" | good
|-
| Hybrid/lossy
| style="background: #CCFFCC" | [[LossyWAV]]
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #CCFFCC" | [[LossyWAV]]
| style="background: #FF9900" | no
| style="background: #CCFFCC" | [[LossyWAV]]
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
|-
| RIFF chunks
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
|-
| Streaming
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
|-
| Open source
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #FFCC66" | no{{ref label|tak_os|F|F}}
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
|-
| Multichannel
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes{{ref label|multichannel_ape|E|E}}
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
|-
| OS support
| style="background: #00FF00" | All
| style="background: #00FF00" | All
| style="background: #00FF00" | All
| style="background: #CCFFCC" | Win/Wine
| style="background: #00FF00" | All
| style="background: #CCFFCC" | Win/Mac
| style="background: #00FF00" | Win/Mac/Linux
| style="background: #00FF00" | All
|}

{|
|-
|{{note label|speed|A|A}} Speed and Compression are based on '''each encoder's default settings''' and taken from [http://www.audiograaf.nl/downloads.html this comparison].
|-
|{{note label|comp|B|B}} The Compression ratio is compressed size/uncompressed size * 100. So, lower is better.
|-
|{{note label|error|C|C}} Error handling means that a codec can detect a corruption (flipped bit) in a file and warn the user about it, but it will still decode the rest of the file.
|-
|{{note label|error_ape|D|D}} The official Monkey's Audio decoder does not support decoding through errors, but this may be achieved with FFmpeg or Winamp, though likely not, when the "Insane" preset is used.
|-
|{{note label|multichannel_ape|E|E}} Since version 4.86
|-
|{{note label|tak_os|F|F}} Unofficial (but properly working) open source decoder is available as part of ffmpeg
== Codecs ==

These are the most popular lossless codecs, in alphabetical order:

=== Apple Lossless Audio Codec (ALAC) ===
https://alac.macosforge.org/trac

[[ALAC]] is a codec developed by Apple for usage in [[Apple iPod|iPod]] and AirPort Express.

'''ALAC pros'''
* [[Open source]] (encoding and decoding via FFmpeg and [[CueTools|CUETools]], decoding only via [http://craz.net/programs/itunes/alac.html a standalone decoder])
* Fast encoding
* Fast decoding
* Hardware support ([[Apple iPod|iPod]], AirPort Express)
* Software support (iTunes, Quicktime)
* Independent encoder implementation available: ffmpeg
* Streaming support
* Tagging support (QT tags)
* Supports [[multichannel]]. Limited to 8 channels. Only limited set of channels layouts is supported - https://github.com/nu774/qaac/wiki/Multichannel--handling
* Supports [[high resolution]]s
* Used by a few online stores

''' ALAC cons '''
* Limited software support
* No error detection/robustness<ref>[http://www.hydrogenaud.io/forums/index.php?s=&showtopic=33226&view=findpost&p=862031 HA forum post discussing ALAC robustness]</ref>
* No hybrid/lossy mode (and not [[LossyWAV]] compatible)
* Not very efficient

''' ALAC Other features '''
* Fits in the [[MP4]] container

=== Free Lossless Audio Codec (FLAC) ===
https://xiph.org/flac/

[[FLAC]] is a lossless codec developed by Josh Coalson. It's part of the Xiph multimedia portfolio, along with [[Opus]], [[Ogg]], [[Vorbis]], [[Speex]] and [[Theora]].

''' FLAC pros '''
* [[Open source]]
* Very fast decoding
* Very fast encoding
* Very good hardware support (Android, Marantz, Sonos, [http://xiph.org/flac/links.html many others])
* Very good software support
* Independent encoder implementations available: flake/ffmpeg, FLACCL
* Error robustness
* Streaming support
* Supports [[multichannel]]. Limited to 8 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is supported but support is not included in official specification. With reference encoder undocumented option --channel-map=none is needed to encode some non-standard layouts (e.g. 4.1; FL,FR,FC,BC), but no special options are needed with ffmpeg's encoder.
* Supports [[high resolution]]s
* Tagging support (FLAC tags)
* Supports [[RIFF]] chunks
* Pipe support
* Used by a few [http://xiph.org/flac/links.html#music online stores]

''' FLAC cons '''
* No hybrid/lossy mode (but is [[LossyWAV]] compatible)
* Does not handle 32-bit float and there is no encoder that can render to 32-bit integer

''' FLAC Other features '''
* Supports embedded CUE sheets (with [http://flac.sourceforge.net/faq.html#general__no_cuesheet_tags limitations])
* Includes MD5 hashes for quick integrity checking as standard
* Fits the [[Ogg]] and [[Matroska]] containers

=== Monkey's Audio (APE) ===
https://www.monkeysaudio.com/

[[Monkey's Audio]] is a very efficient lossless compressor developed by Matt Ashland.

''' APE pros '''
* High compression
* Fast encoding
* Good software support
* Supports [[multichannel]] (since version 4.86). Limited to 8 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is supported
* Supports [[high resolution]]s
* Simple and user friendly. Official GUI provided.
* Java version (multiplatform)
* Error robustness/decoding up to -c3000 (High compression)<ref>http://www.hydrogenaud.io/forums/index.php?showtopic=98984&st=0&p=821420&#entry821420</ref>
* Tagging support ([[ID3v1]], [[APE tags]])
* Supports [[RIFF]] chunks (only in the GUI encoder)
* Pipe support (only in a [http://www.etree.org/shnutils/shntool/ special] version)

''' APE cons '''
* Problematic license (source provided, no modification or redistribution rights)
* Slow decoding
* No hybrid/lossy mode (and not [[LossyWAV]] compatible)
* Limited hardware support (Rockbox, some Cowon players); poor battery life due to complicated decoding (see [http://www.rockbox.org/wiki/SoundCodecMonkeysAudio MP3 player benchmarks])
* Higher compression levels are extremely CPU intensive

''' APE Other features '''
* Includes MD5 hashes for quick integrity checking
* Supports APL image link files (similar to CUE sheets)

=== OptimFROG (OFR) ===
http://www.losslessaudio.org/

[[OptimFROG]] is a lossless format developed by Florin Ghido to become the champion in audio compression.

''' OFR pros '''
* Very high compression
* Good software support
* Error robustness
* Streaming support
* Supports [[high resolution]]s
* Hybrid/lossy mode
* Tagging support ([[ID3]], [[APE tags]])
* Supports [[RIFF]] chunks

''' OFR cons '''
* Closed source
* No [[multichannel]] audio support
* No hardware support
* Very slow decoding
* Slow encoding
* More than one tagging method allowed (ambiguity possible)

''' OFR Other features '''
* Supports 32bit float streams
* Includes MD5 hashes for quick integrity checking

=== Tom's verlustfreier Audiokompressor (TAK) ===
http://www.thbeck.de/Tak/Tak.html

[[TAK]] is a lossless codec developed by Thomas Becker.

''' TAK pros '''
* Very fast decoding
* Very fast encoding
* Very high efficiency
* Error robust
* Supports [[multichannel]]. Limited to 6 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is supported
* Supports [[high resolution]]s
* Tagging support
* Supports RIFF chunks
* Pipe support
* Streamable

''' TAK cons '''
* Closed source (but unofficial open source decoder is available as part of ffmpeg)
* No hybrid/lossy mode (but is [[LossyWAV]] compatible)
* No hardware support
* Average software support
* Doesn't support Unicode (yet)

''' TAK Other features '''
* Optional MD5 checksum

=== True Audio (TTA) ===
http://tta.tausoft.org/

[[TTA]] is a lossless codec developed by a international team of programmers.

''' TTA pros '''
* [[Open source]]
* Supports [[multichannel]]. Reference encoder/decoder is limited to 6 channels. ffmpeg's encoder/decoder is limited to 16 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is '''not''' supported
* Supports [[high resolution]]s
* Tagging support ([[ID3]]v1, ID3v2 or [[APEv2]])
* Embedded CUE sheets support
* Error robustness
* Pipe support
* Average compression
* Fast encoding/decoding
* Symmetric algorithm
* Ultra low latency

''' TTA cons '''
* No hybrid/lossy mode
* Doesn't support [[RIFF]] chunks
* Limited hardware support

''' TTA Other features '''
* Fits the [[Matroska]] container
* Password protection

=== WavPack (WV) ===
http://www.wavpack.com/

[[WavPack]] is a fast and featureful lossless codec developed by David Bryant.

''' WV pros '''
* [[Open source]]
* Fast decoding
* Very fast encoding
* Good efficiency
* Error robustness
* Streaming support
* Supports [[multichannel]]. Limited to 255 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is supported
* Supports [[high resolution]]s
* Hybrid/lossy mode
* Tagging support ([[ID3v1]], [[APE tags]])
* Supports [[RIFF]] chunks
* Ability to create self extracting files for Win32 platform
* Pipe support
* Good software support
* Works with Android (Through third party software, sucha as VLC.)

''' WV cons '''
* Limited hardware player support ([http://www.rockbox.org/ RockBox])
* More than one tagging method allowed (Ambiguity possible, but unlikely as APEv2 tags have been the preferred method for quite some time.)

''' WV Other features '''
* Can compress the Direct-Stream Digital (DSD) audio recording format
* Supports 32bit float streams
* Supports embedded CUE sheets
* Accept audio files bigger than 4GB
* Includes MD5 hashes for quick integrity checking
* Can encode in both symmetrical and asymmetrical modes.
* Fits the [[Matroska]] container

=== Windows Media Audio Lossless (WMAL) ===
https://msdn.microsoft.com/en-us/library/ff819508(v=vs.85).aspx

WMA Lossless is the lossless codec developed by Microsoft to be featured in their Windows Media codec portfolio.

''' WMAL pros '''
* Streaming support
* Supports [[multichannel]] audio and [[high resolution]]s.
* Tagging support (proprietary)
* Pipe support

''' WMAL cons '''
* Limited hardware support (Microsoft Zune, Toshiba Gigabeat S and V. Both discontinued and obsolete. Rockbox, for 16-bit stereo files only.)
* Limited software support outside of the Microsoft Windows operating system.
* Extremely low efficiency
* Closed source
* No hybrid/lossy mode (but is [[LossyWAV]] compatible)
* Doesn't support [[RIFF]] chunks

''' WMAL Other features '''
* Fits the [[ASF]] container

=== Other Formats ===
Aside from the formats mentioned above, there are in fact quite a lot of other lossless formats. To keep the table and list brief and readable, a few formats have not been mentioned.

====DTS-HD Master Audio====
Similar to the MPEG-4 SLS format, this format has a core track in an older, more widely supported format, DTS. This core lossy track is made lossless by a secondary track with correction data. It is an optional codec in Blu-ray implementations. Its main use is surround sound encoding, and as is the case with MLP, the price of the encoder ensures it is only used in mastering of Blu-ray discs.

====LA====
http://www.lossless-audio.com/

LA features an extremely high compression (on par with OptimFrog highest modes, but a bit faster), but it hasn't been updated for more than 10 years. Furthermore, backward compatibility is not guaranteed, so using it for archiving might pose a few problems. It isn't able to cope with file corruption either, software support is very limited and isn't open source.

====MLP/Dolby TrueHD====
The [[MLP|MLP codec]] (of which the mathematical basis was used in Dolby TrueHD) is the codec used for DVD-Audio. It was mandatory in any HD-DVD implementation and optional for Blu-Ray in its Dolby TrueHD form. It is known to support the 'wasted bits' scheme used in LossyWAV. As encoders are very expensive, its use outside DVD/Blu-ray mastering environments is non-existent. Its main use is encoding surround sound data.

====MPEG-4 ALS====
MPEG-4 ALS is the successor to LPAC, which it was based on. It has been as a ISO standard and there is a reference encoder/decoder, but like TTA, it does not have features that make it stand out from other codecs, nor backing by a large organisation, so it hasn't much software and no hardware support.

====MPEG-4 SLS====
MPEG-4 SLS is a special codec, having a AAC core track and a 'correction track'. Also known as HD-AAC, SLS stands for Scalable to Lossless. However, there is to date still no affordable software to play, encode or decode (the lossless part of) SLS files.

====Shorten====
http://www.etree.org/shncom.html

Shorten was one of the first widely-used lossless formats, and it still occasionally found on the internet, especially in archives, for example etree.org. It is quite fast in both encoding and decoding, but doesn't compress very much. Furthermore, seeking has a troubled past as well as tagging. It is considered obsolete.

====Real Lossless====
Part of the Real codec suite, Real Lossless too hasn't any very special features that make it stand out. Just like WMA Lossless and Apple Lossless, it was created to fit in a codec suite, but unlike WMA Lossless and Apple Lossless, there is no hardware support and software support is limited. Compression is on par with most other codecs, but it is rather slow to encode.

====Oddball formats====
There are a few archaic formats of which encoders and decoders are hard to get by. Most of those would have disappeared by now, but some of them are being preserved for posterity at [[User:Rjamorim|rjamorim]]'s

* Advanced Digital Audio (ADA)
* [http://www.logarithmic.net/pfh/bonk Bonk]
* AudioZip
* Dakx WAV
* Entis Lab MIO
* LiteWave
* [http://www.nue.tu-berlin.de/menue/mitarbeiter/ehemalige_mitarbeiter/tilman_liebchen/lpac_-_lossless_audio_codec_for_windows_and_linux/ LPAC]
* Marian's a-Pac
* [http://mp3hd-toolkit.soft32.com/ mp3HD (MPEG-1 Audio Layer III HD)]
* Pegasus SPS
* [http://www.free-codecs.com/download/rk_audio_compressor.htm RK Audio (RKAU)]
* Ogg Squish/Tarkin
* Sonarc
* VocPack
* [http://www.firstpr.com.au/audiocomp/lossless/wavarc/ WavArc]
* [http://www.firstpr.com.au/audiocomp/lossless/WaveZip/ WaveZip]/MUSICompress

== See also ==
* [[Lossless]]

== External links ==
''' Other lossless compressions comparisons '''
''Sorted based on last '''update''' date.''

* [http://www.audiograaf.nl/downloads.html Martijn van Beurden's comparison] - tries to compare all codecs and settings with a balanced pool of music (last updated 2015-01-05)
* [http://www.squeezechart.com/audio.html Squeezechart audio] - tests as much codecs as possible, but not all their settings and with a limited test corpus (last updated 2013-10-31)
* <s>[http://synthetic-soul.co.uk/comparison/lossless/index.asp Synthetic Soul's comparison] (last update 2007-07-28)</s>
* <s>Johan De Bock's speed oriented comparison</s> - best choices speedwise are indicated in green, mostly electronic music (last updated 2006-07-22)
* <s>Hans Heijden's</s> -- used as reference to build the table (last updated 2006-07-07)
* <s>Josef Pohm's comparison, hosted by Synthetic Soul</s> (last update 2006-05-29)
* [http://www.bobulous.org.uk/misc/lossless_audio_2006.html Bobulous' lossless audio comparison] — a look at six lossless formats in terms of speed and file size (last updated 2006-05-22)
* <s>Jhan De Bock's size oriented comparison</s> - aimed only at the maximum compression setting for each codec (based on a somewhat limited set of samples, however) (last updated 2006-05-19)
* <s>Gruboolez'</s> -- comparing only classical music (last updated 2005-02-27)
* <s>Speek's</s> (last updated 2005-02-07)
*[http://www.firstpr.com.au/audiocomp/lossless/ Lossless Compression of Audio] Much information about oddball formats including comparison of them. (last updated 2005-10-21)

''' More on lossless compressions '''
* [http://web.archive.org/web/20080731103800/http://www.losslessaudioblog.com/ The Lossless Audio Blog], retrieved from archive.org - by windmiller, is a reliable and complete source of news about lossless compression.
* Go to the [http://www.hydrogenaudio.org/forums/index.php?showtopic=33226 Hydrogenaudio thread] to discuss this article.

== References ==

<references/>
[[Category:Guides]]

Lossless comparison

2020-06-24T22:32:04Z

12.218.147.66: /* Windows Media Audio Lossless (WMAL) */

The '''lossless comparison page''' aims to gather information about lossless codecs available so users can make an informed decision as to what lossless codec to choose for their needs.

== Introduction ==
Given the enormous number of [[lossless]] audio compressor choices available, it is a very difficult task to choose the one most suited for each person's needs. Some people take into consideration only compression performance when choosing a codec, but as the following table and article shows, there are several other features worth taking into consideration when making a choice.

For example, users wanting good multiplatform compatibility and robustness (e.g., people sharing live recordings) would favour [[WavPack]] or [[FLAC]]. Another user, looking for the very highest compression available, would go with [[OptimFROG]]. Someone wanting portable support would use [[FLAC]] or [[ALAC]], and so on. En fin, this is not a matter worth getting too worked up about. If you later find out the codec you chose isn't the best for your needs, you can just transcompress to another format, without risk of losing quality.

'''Note:''' for latest comparison of lossless compression, scroll down to the [[Lossless comparison#External links|Links section of this page]].

== Comparison Table ==


{| class="wikitable" cellspacing="2"
!width="120px"|'''Features'''
! width="90px" | [[#Free Lossless Audio Codec (FLAC)|FLAC]]
! width="90px" | [[#Apple Lossless Audio Codec (ALAC)|ALAC]]
! width="90px" | [[#WavPack (WV)|WavPack]]
! width="90px" | [[#Tom's_verlustfreier_Audiokompressor (TAK)|TAK]]
! width="90px" | [[#Monkey's_Audio (APE)|Monkey's]]
! width="90px" | [[#Windows Media Audio Lossless (WMAL)|WMAL]]
! width="90px" | [[#OptimFROG (OFR)|OptimFROG]]
! width="90px" | [[#True Audio (TTA)|TTA]]
|- 
| Encoding speed{{ref label|speed|A|A}}
| style="background: #00FF00" | very fast
| style="background: #CCFFCC" | fast
| style="background: #00FF00" | very fast
| style="background: #00FF00" | very fast
| style="background: #CCFFCC" | fast
| style="background: #CCFFCC" | fast
| style="background: #FFCC66" | slow
| style="background: #00FF00" | very fast
|- 
| Decoding speed{{ref label|speed|A|A}}
| style="background: #00FF00" | very fast
| style="background: #CCFFCC" | fast
| style="background: #CCFFCC" | fast
| style="background: #00FF00" | very fast
| style="background: #FFCC66" | slow
| style="background: #FFCC66" | average
| style="background: #FF9900" | very slow
| style="background: #CCFFCC" | fast
|- 
| Compression{{ref label|speed|A|A}}{{ref label|comp|B|B}}
| style="background: #CCFFCC" | 57.0%
| style="background: #CCFFCC" | 57.8%
| style="background: #CCFFCC" | 57.1%
| style="background: #00FF00" | 56.0%
| style="background: #00FF00" | 55.1%
| style="background: #FFCC66" | 58.4%
| style="background: #00FF00" | 54.6%
| style="background: #CCFFCC" | 56.6%
|-
| # presets
| style="background: #CCFFCC" | 9
| style="background: #CCFFCC" | 2
| style="background: #CCFFCC" | > 10
| style="background: #CCFFCC" | > 10
| style="background: #CCFFCC" | 5
| style="background: #CCFFCC" | 1
| style="background: #CCFFCC" | > 10
| style="background: #CCFFCC" | 1
|-
| Error handling{{ref label|error|C|C}}
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #CCFFCC" | yes{{ref label|error_ape|D|D}}
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
|-
| Tagging
| style="background: #00FF00" | Vorbis tags
| style="background: #00FF00" | iTunes
| style="background: #CCFFCC" | ID3/APEv2
| style="background: #00FF00" | APEv2
| style="background: #00FF00" | APEv2
| style="background: #00FF00" | ASF
| style="background: #CCFFCC" | ID3/APEv2
| style="background: #00FF00" | ID3v1/2 or APEv2
|-
| Hardware support
| style="background: #00FF00" | very good
| style="background: #CCFFCC" | good
| style="background: #FFCC66" | limited
| style="background: #FF9900" | no
| style="background: #FFCC66" | limited
| style="background: #FFCC66" | limited
| style="background: #FF9900" | no
| style="background: #FFCC66" | limited
|-
| Software support
| style="background: #00FF00" | very good
| style="background: #CCFFCC" | good
| style="background: #CCFFCC" | good
| style="background: #FFCC66" | average
| style="background: #CCFFCC" | good
| style="background: #CCFFCC" | good
| style="background: #FFCC66" | average
| style="background: #CCFFCC" | good
|-
| Hybrid/lossy
| style="background: #CCFFCC" | [[LossyWAV]]
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #CCFFCC" | [[LossyWAV]]
| style="background: #FF9900" | no
| style="background: #CCFFCC" | [[LossyWAV]]
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
|-
| RIFF chunks
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
|-
| Streaming
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
|-
| Open source
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #FFCC66" | no{{ref label|tak_os|F|F}}
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
|-
| Multichannel
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes{{ref label|multichannel_ape|E|E}}
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
|-
| OS support
| style="background: #00FF00" | All
| style="background: #00FF00" | All
| style="background: #00FF00" | All
| style="background: #CCFFCC" | Win/Wine
| style="background: #00FF00" | All
| style="background: #CCFFCC" | Win/Mac
| style="background: #00FF00" | Win/Mac/Linux
| style="background: #00FF00" | All
|}

{|
|-
|{{note label|speed|A|A}} Speed and Compression are based on '''each encoder's default settings''' and taken from [http://www.audiograaf.nl/downloads.html this comparison].
|-
|{{note label|comp|B|B}} The Compression ratio is compressed size/uncompressed size * 100. So, lower is better.
|-
|{{note label|error|C|C}} Error handling means that a codec can detect a corruption (flipped bit) in a file and warn the user about it, but it will still decode the rest of the file.
|-
|{{note label|error_ape|D|D}} The official Monkey's Audio decoder does not support decoding through errors, but this may be achieved with FFmpeg or Winamp, though likely not, when the "Insane" preset is used.
|-
|{{note label|multichannel_ape|E|E}} Since version 4.86
|-
|{{note label|tak_os|F|F}} Unofficial (but properly working) open source decoder is available as part of ffmpeg
== Codecs ==

These are the most popular lossless codecs, in alphabetical order:

=== Apple Lossless Audio Codec (ALAC) ===
https://alac.macosforge.org/trac

[[ALAC]] is a codec developed by Apple for usage in [[Apple iPod|iPod]] and AirPort Express.

'''ALAC pros'''
* [[Open source]] (encoding and decoding via FFmpeg and [[CueTools|CUETools]], decoding only via [http://craz.net/programs/itunes/alac.html a standalone decoder])
* Fast encoding
* Fast decoding
* Hardware support ([[Apple iPod|iPod]], AirPort Express)
* Software support (iTunes, Quicktime)
* Independent encoder implementation available: ffmpeg
* Streaming support
* Tagging support (QT tags)
* Supports [[multichannel]]. Limited to 8 channels. Only limited set of channels layouts is supported - https://github.com/nu774/qaac/wiki/Multichannel--handling
* Supports [[high resolution]]s
* Used by a few online stores

''' ALAC cons '''
* Limited software support
* No error detection/robustness<ref>[http://www.hydrogenaud.io/forums/index.php?s=&showtopic=33226&view=findpost&p=862031 HA forum post discussing ALAC robustness]</ref>
* No hybrid/lossy mode (and not [[LossyWAV]] compatible)
* Not very efficient

''' ALAC Other features '''
* Fits in the [[MP4]] container

=== Free Lossless Audio Codec (FLAC) ===
https://xiph.org/flac/

[[FLAC]] is a lossless codec developed by Josh Coalson. It's part of the Xiph multimedia portfolio, along with [[Opus]], [[Ogg]], [[Vorbis]], [[Speex]] and [[Theora]].

''' FLAC pros '''
* [[Open source]]
* Very fast decoding
* Very fast encoding
* Very good hardware support (Android, Marantz, Sonos, [http://xiph.org/flac/links.html many others])
* Very good software support
* Independent encoder implementations available: flake/ffmpeg, FLACCL
* Error robustness
* Streaming support
* Supports [[multichannel]]. Limited to 8 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is supported but support is not included in official specification. With reference encoder undocumented option --channel-map=none is needed to encode some non-standard layouts (e.g. 4.1; FL,FR,FC,BC), but no special options are needed with ffmpeg's encoder.
* Supports [[high resolution]]s
* Tagging support (FLAC tags)
* Supports [[RIFF]] chunks
* Pipe support
* Used by a few [http://xiph.org/flac/links.html#music online stores]

''' FLAC cons '''
* No hybrid/lossy mode (but is [[LossyWAV]] compatible)
* Does not handle 32-bit float and there is no encoder that can render to 32-bit integer

''' FLAC Other features '''
* Supports embedded CUE sheets (with [http://flac.sourceforge.net/faq.html#general__no_cuesheet_tags limitations])
* Includes MD5 hashes for quick integrity checking as standard
* Fits the [[Ogg]] and [[Matroska]] containers

=== Monkey's Audio (APE) ===
https://www.monkeysaudio.com/

[[Monkey's Audio]] is a very efficient lossless compressor developed by Matt Ashland.

''' APE pros '''
* High compression
* Fast encoding
* Good software support
* Supports [[multichannel]] (since version 4.86). Limited to 8 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is supported
* Supports [[high resolution]]s
* Simple and user friendly. Official GUI provided.
* Java version (multiplatform)
* Error robustness/decoding up to -c3000 (High compression)<ref>http://www.hydrogenaud.io/forums/index.php?showtopic=98984&st=0&p=821420&#entry821420</ref>
* Tagging support ([[ID3v1]], [[APE tags]])
* Supports [[RIFF]] chunks (only in the GUI encoder)
* Pipe support (only in a [http://www.etree.org/shnutils/shntool/ special] version)

''' APE cons '''
* Problematic license (source provided, no modification or redistribution rights)
* Slow decoding
* No hybrid/lossy mode (and not [[LossyWAV]] compatible)
* Limited hardware support (Rockbox, some Cowon players); poor battery life due to complicated decoding (see [http://www.rockbox.org/wiki/SoundCodecMonkeysAudio MP3 player benchmarks])
* Higher compression levels are extremely CPU intensive

''' APE Other features '''
* Includes MD5 hashes for quick integrity checking
* Supports APL image link files (similar to CUE sheets)

=== OptimFROG (OFR) ===
http://www.losslessaudio.org/

[[OptimFROG]] is a lossless format developed by Florin Ghido to become the champion in audio compression.

''' OFR pros '''
* Very high compression
* Good software support
* Error robustness
* Streaming support
* Supports [[high resolution]]s
* Hybrid/lossy mode
* Tagging support ([[ID3]], [[APE tags]])
* Supports [[RIFF]] chunks

''' OFR cons '''
* Closed source
* No [[multichannel]] audio support
* No hardware support
* Very slow decoding
* Slow encoding
* More than one tagging method allowed (ambiguity possible)

''' OFR Other features '''
* Supports 32bit float streams
* Includes MD5 hashes for quick integrity checking

=== Tom's verlustfreier Audiokompressor (TAK) ===
http://www.thbeck.de/Tak/Tak.html

[[TAK]] is a lossless codec developed by Thomas Becker.

''' TAK pros '''
* Very fast decoding
* Very fast encoding
* Very high efficiency
* Error robust
* Supports [[multichannel]]. Limited to 6 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is supported
* Supports [[high resolution]]s
* Tagging support
* Supports RIFF chunks
* Pipe support
* Streamable

''' TAK cons '''
* Closed source (but unofficial open source decoder is available as part of ffmpeg)
* No hybrid/lossy mode (but is [[LossyWAV]] compatible)
* No hardware support
* Average software support
* Doesn't support Unicode (yet)

''' TAK Other features '''
* Optional MD5 checksum

=== True Audio (TTA) ===
http://tta.tausoft.org/

[[TTA]] is a lossless codec developed by a international team of programmers.

''' TTA pros '''
* [[Open source]]
* Supports [[multichannel]]. Reference encoder/decoder is limited to 6 channels. ffmpeg's encoder/decoder is limited to 16 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is '''not''' supported
* Supports [[high resolution]]s
* Tagging support ([[ID3]]v1, ID3v2 or [[APEv2]])
* Embedded CUE sheets support
* Error robustness
* Pipe support
* Average compression
* Fast encoding/decoding
* Symmetric algorithm
* Ultra low latency

''' TTA cons '''
* No hybrid/lossy mode
* Doesn't support [[RIFF]] chunks
* Limited hardware support

''' TTA Other features '''
* Fits the [[Matroska]] container
* Password protection

=== WavPack (WV) ===
http://www.wavpack.com/

[[WavPack]] is a fast and featureful lossless codec developed by David Bryant.

''' WV pros '''
* [[Open source]]
* Fast decoding
* Very fast encoding
* Good efficiency
* Error robustness
* Streaming support
* Supports [[multichannel]]. Limited to 255 channels. Channel mask in [https://docs.microsoft.com/ru-ru/windows/win32/api/mmreg/ns-mmreg-waveformatextensible WAVEFORMATEXTENSIBLE] is supported
* Supports [[high resolution]]s
* Hybrid/lossy mode
* Tagging support ([[ID3v1]], [[APE tags]])
* Supports [[RIFF]] chunks
* Ability to create self extracting files for Win32 platform
* Pipe support
* Good software support

''' WV cons '''
* Limited hardware player support ([http://www.rockbox.org/ RockBox])
* More than one tagging method allowed (ambiguity possible)

''' WV Other features '''
* Can compress the Direct-Stream Digital (DSD) audio recording format
* Supports 32bit float streams
* Supports embedded CUE sheets
* Accept audio files bigger than 4GB
* Includes MD5 hashes for quick integrity checking
* Can encode in both symmetrical and asymmetrical modes.
* Fits the [[Matroska]] container

=== Windows Media Audio Lossless (WMAL) ===
https://msdn.microsoft.com/en-us/library/ff819508(v=vs.85).aspx

WMA Lossless is the lossless codec developed by Microsoft to be featured in their Windows Media codec portfolio.

''' WMAL pros '''
* Streaming support
* Supports [[multichannel]] audio and [[high resolution]]s.
* Tagging support (proprietary)
* Pipe support

''' WMAL cons '''
* Limited hardware support (Microsoft Zune, Toshiba Gigabeat S and V. Both discontinued and obsolete. Rockbox, for 16-bit stereo files only.)
* Limited software support outside of the Microsoft Windows operating system.
* Extremely low efficiency
* Closed source
* No hybrid/lossy mode (but is [[LossyWAV]] compatible)
* Doesn't support [[RIFF]] chunks

''' WMAL Other features '''
* Fits the [[ASF]] container

=== Other Formats ===
Aside from the formats mentioned above, there are in fact quite a lot of other lossless formats. To keep the table and list brief and readable, a few formats have not been mentioned.

====DTS-HD Master Audio====
Similar to the MPEG-4 SLS format, this format has a core track in an older, more widely supported format, DTS. This core lossy track is made lossless by a secondary track with correction data. It is an optional codec in Blu-ray implementations. Its main use is surround sound encoding, and as is the case with MLP, the price of the encoder ensures it is only used in mastering of Blu-ray discs.

====LA====
http://www.lossless-audio.com/

LA features an extremely high compression (on par with OptimFrog highest modes, but a bit faster), but it hasn't been updated for more than 10 years. Furthermore, backward compatibility is not guaranteed, so using it for archiving might pose a few problems. It isn't able to cope with file corruption either, software support is very limited and isn't open source.

====MLP/Dolby TrueHD====
The [[MLP|MLP codec]] (of which the mathematical basis was used in Dolby TrueHD) is the codec used for DVD-Audio. It was mandatory in any HD-DVD implementation and optional for Blu-Ray in its Dolby TrueHD form. It is known to support the 'wasted bits' scheme used in LossyWAV. As encoders are very expensive, its use outside DVD/Blu-ray mastering environments is non-existent. Its main use is encoding surround sound data.

====MPEG-4 ALS====
MPEG-4 ALS is the successor to LPAC, which it was based on. It has been as a ISO standard and there is a reference encoder/decoder, but like TTA, it does not have features that make it stand out from other codecs, nor backing by a large organisation, so it hasn't much software and no hardware support.

====MPEG-4 SLS====
MPEG-4 SLS is a special codec, having a AAC core track and a 'correction track'. Also known as HD-AAC, SLS stands for Scalable to Lossless. However, there is to date still no affordable software to play, encode or decode (the lossless part of) SLS files.

====Shorten====
http://www.etree.org/shncom.html

Shorten was one of the first widely-used lossless formats, and it still occasionally found on the internet, especially in archives, for example etree.org. It is quite fast in both encoding and decoding, but doesn't compress very much. Furthermore, seeking has a troubled past as well as tagging. It is considered obsolete.

====Real Lossless====
Part of the Real codec suite, Real Lossless too hasn't any very special features that make it stand out. Just like WMA Lossless and Apple Lossless, it was created to fit in a codec suite, but unlike WMA Lossless and Apple Lossless, there is no hardware support and software support is limited. Compression is on par with most other codecs, but it is rather slow to encode.

====Oddball formats====
There are a few archaic formats of which encoders and decoders are hard to get by. Most of those would have disappeared by now, but some of them are being preserved for posterity at [[User:Rjamorim|rjamorim]]'s

* Advanced Digital Audio (ADA)
* [http://www.logarithmic.net/pfh/bonk Bonk]
* AudioZip
* Dakx WAV
* Entis Lab MIO
* LiteWave
* [http://www.nue.tu-berlin.de/menue/mitarbeiter/ehemalige_mitarbeiter/tilman_liebchen/lpac_-_lossless_audio_codec_for_windows_and_linux/ LPAC]
* Marian's a-Pac
* [http://mp3hd-toolkit.soft32.com/ mp3HD (MPEG-1 Audio Layer III HD)]
* Pegasus SPS
* [http://www.free-codecs.com/download/rk_audio_compressor.htm RK Audio (RKAU)]
* Ogg Squish/Tarkin
* Sonarc
* VocPack
* [http://www.firstpr.com.au/audiocomp/lossless/wavarc/ WavArc]
* [http://www.firstpr.com.au/audiocomp/lossless/WaveZip/ WaveZip]/MUSICompress

== See also ==
* [[Lossless]]

== External links ==
''' Other lossless compressions comparisons '''
''Sorted based on last '''update''' date.''

* [http://www.audiograaf.nl/downloads.html Martijn van Beurden's comparison] - tries to compare all codecs and settings with a balanced pool of music (last updated 2015-01-05)
* [http://www.squeezechart.com/audio.html Squeezechart audio] - tests as much codecs as possible, but not all their settings and with a limited test corpus (last updated 2013-10-31)
* <s>[http://synthetic-soul.co.uk/comparison/lossless/index.asp Synthetic Soul's comparison] (last update 2007-07-28)</s>
* <s>Johan De Bock's speed oriented comparison</s> - best choices speedwise are indicated in green, mostly electronic music (last updated 2006-07-22)
* <s>Hans Heijden's</s> -- used as reference to build the table (last updated 2006-07-07)
* <s>Josef Pohm's comparison, hosted by Synthetic Soul</s> (last update 2006-05-29)
* [http://www.bobulous.org.uk/misc/lossless_audio_2006.html Bobulous' lossless audio comparison] — a look at six lossless formats in terms of speed and file size (last updated 2006-05-22)
* <s>Jhan De Bock's size oriented comparison</s> - aimed only at the maximum compression setting for each codec (based on a somewhat limited set of samples, however) (last updated 2006-05-19)
* <s>Gruboolez'</s> -- comparing only classical music (last updated 2005-02-27)
* <s>Speek's</s> (last updated 2005-02-07)
*[http://www.firstpr.com.au/audiocomp/lossless/ Lossless Compression of Audio] Much information about oddball formats including comparison of them. (last updated 2005-10-21)

''' More on lossless compressions '''
* [http://web.archive.org/web/20080731103800/http://www.losslessaudioblog.com/ The Lossless Audio Blog], retrieved from archive.org - by windmiller, is a reliable and complete source of news about lossless compression.
* Go to the [http://www.hydrogenaudio.org/forums/index.php?showtopic=33226 Hydrogenaudio thread] to discuss this article.

== References ==

<references/>
[[Category:Guides]]

Vorbis

2020-06-08T04:43:26Z

12.218.147.66: /* Cons */ Fix typo

{{featured}}
{{Codec Infobox
| name = Ogg Vorbis
| logo = [[Image:Fish logo.png]]
| type = lossy
| purpose = General audio compression at bitrates ~64–400 kbps
| maintainer = Christopher Montgomery, Xiph Community
| website = http://www.vorbis.com/
}}
{{infobox ha recommended encoder
| encoder = [[aoTuV]]
| encoder version = beta 5
| encoder release date = {{start date and age|2006|10|24}}
}}
'''Vorbis''' (commonly used inside the [[Ogg]] container) is a fully open, non-proprietary, patent-free (subject to [http://www.hydrogenaudio.org/forums/index.php?showtopic=13531 speculation]), and royalty-free, general-purpose compressed audio format for mid to high quality (8 khz–48.0 kHz, 16+ bit, [[multichannel]]) audio and music at fixed and variable bitrates from 16 to >256 kbps/channel. This places vorbis in the same competitive class as audio representations such as MPEG-4 ([[AAC]]), and similar to, but higher performance than [[MP3]], TwinVQ ([[VQF]]), [[WMA]] and [[PAC]]. Vorbis is the first of a planned family of Ogg multimedia coding formats being developed as part of Xiph.org's ogg multimedia project.

==Introduction==
Informal listening test suggests Vorbis to be comparable to MPEG-4 [[AAC]] at most bitrates and [[Musepack]] at 128 kbps. Transparency is generally reached at about 150–170 kbps (-q 5) (with some exceptions). The encoder is reasonably young and unoptimized, so further improvements can always be expected.

Unfortunately, Xiph.org has failed to improve Vorbis at a steady rate since its initial 1.0 release in July 2002 (due to other developement projects and time constraints). Since then development has been led by other coders such as [http://sjeng.org/vorbisgt3.html Garf] and [http://www.geocities.jp/aoyoume/aotuv/ Aoyumi]. Aoyumi's '''[[aoTuV]]''' series of encoders was incorporated into the September 2004 release of 1.1, which brought about the first quality improvements across the board for 2 years. Aoyumi's Beta 4.51 was found to be very good, so it was re-branded into aoTuV Release 1 and it was the recommended encoder until June 2007. The latest tuning is aoTuV beta 5, which improves further on the low-bitrate quality without sacrificing compression, and it is currently the recommended Vorbis encoder at Hydrogenaudio.

At the time being, Aoyumi's tuning (since aoTuV Release 1 up to aoTuV Beta 5) has not been incorporated yet into the 'official' Vorbis line.

Vorbis has had success with many recent video game titles employing Vorbis as opposed to MP3 (with Epic Games' Unreal Tournament 2003 and Unreal Tournament 2004, the PC port of Microsoft's Halo and Uru being notable examples). (Ogg) Vorbis is also an official part of the [http://www.openal.org/extensions.html OpenAL] API extension library, used in many popular [http://www.openal.org/titles.html computer games]. On April 10, 2006, [http://www.radgametools.com/ RAD Game Tools] integrated (Ogg) Vorbis support to their Miles Sound System (MSS), which has been used in over 3,200 games worldwide. This ensures that future games utilizing MSS will have the capability to play (Ogg) Vorbis files. Check out [http://wiki.xiph.org/index.php/Games_that_use_Vorbis xiph wiki] for a full list of games confirmed to use (Ogg) Vorbis.

Vorbis was recently adopted in May of 2010 as the open source codec for Google's new [http://www.webmproject.org/ WebM] project. WebM is combination of the BSD licensed VP8 video codec, Vorbis, and the [[webm]] container a subset of the [[Matroska]] container. It is expected to obtain widespread adoption with a major backing by many hardware based chip manufactures and with the release of Google's new mobile Android platform and Google TV by the year 2011.

'''Before encoding files using (Ogg) Vorbis, check out the [[Recommended Ogg Vorbis|Recommended (Ogg) Vorbis]] article to determine what encoder to use and what settings are recommended by Hydrogenaudio.'''

===Pros===
* (Ogg) Vorbis specification is in the public domain; it is free for commercial or noncommercial use, under both (LGPL and BSD licenes)
* Easy to use high-level API (Application Programming Interface)
* Good all-round performance (>48 kbps – a leading codec at [http://web.archive.org/web/20070403025834/http://www.rjamorim.com/test/multiformat128/results.html 128 kbps])
* Well written [http://www.xiph.org/ogg/vorbis/docs.html specs]
* Supported by most portable (Ogg) [[Vorbis#Supporting Digital Audio Players|DAPs]]
* Suitable for internet-streaming (via [http://www.icecast.org/ Icecast] and other methods)
* Fully [[gapless]] playback
* High potential for further tuning
* Structured to allow the design for a hybrid filterbank

===Cons===
* Limited official development (third-party development is always encouraged)
* Some implementations are more computationally intensive to decode than MP3 (depending upon the architecture and [[Tremor]] optimizations).

== Technical information ==
* Multiple block sizes for window switching including overlap (powers of two only) ''(128/1024, 256/2048, 512/4096)''
* Customly designed [[window function]] is applied similiar to the sine window. it has (good sidelobe rejection)
:<math>w_k = \sin{(\frac{\pi}{2} \cdot sin^2[(\pi\div2n \cdot (k+0.5))]}</math>
* Psychoacoustics masking is exploited via an ([[ATH]] model)
* Masking curves are computed from an ''emperically'' adjusted set of [http://www.zainea.com/masking2.htm Ehmer Curves]
* Modified Discrete Cosine Transform ([[MDCT]]) is used for noise analysis
* Fast Fourier Transform ([[FFT]]) is used for tonal analysis
* Global masking curve is a mixture between calculated FFT+MDCT curves and ATH curves overlayed
* Floor 1 or the noise-floor (envelope) is calculated using the global masking curve & piecewise linear approximation divided by spectrum to generate the residue (fine detail). The Levinson-Durbin [http://www.data-compression.com/speech.html#ana LPC model] in Floor 0 is no longer used, however the code still exists
* [[Noise normalization]] is applied to compensate for energy lost in certain frequency bands due to quantization (rounding).
* The channels are [[channel coupling|coupled]] ''strictly'' by residue using ([http://us.xiph.org/ogg/vorbis/doc/stereo.html point/phase stereo] and lossless)
* Multistage [[Vector quantization]] is used for coding the noise-floor and residue backend using ''trained'' codebooks.
* [[Huffman coding]] is used to minimize vector codeword redundancy

== Software ==
=== Encoders ===
* [[Oggenc]] official command-line encoder (Win32/Posix)
* [[OggDropXPd|OggdropXPd]] advanced drag-and-drop encoder by John33 (Win32)
* [[Lancer]] SSE-optimized vorbis encoder utility and libraries by BlackSword (Win32/Posix)
* [http://www.saunalahti.fi/cse/foobar2000/ foo_vorbisenc] vorbis encoder library for foobar2000 (Win32)

=== Decoders ===
* [http://www.rarewares.org/ogg.html OggDec] for Windows, by John33, a very featureful command line decoder (Win32)
* [[Ogg123]] for Unix systems (GPL), a very simple to use command-line player. (Win32/Posix)
* [http://www.illiminable.com/ogg/ illiminable Ogg Directshow Filters] also plays Speex, Theora and FLAC (Win32)
* [http://www.xiph.org/quicktime/ XiphQT] (Xiph's QuickTime Components) allows playback in [[QuickTime]]/[[iTunes]] (Win32/MacPPC/MacIntel)
* [http://corevorbis.corecodec.org/ CoreVorbis] DirectShow filter (Win32)

=== ReplayGain ===
* [http://www.rarewares.org/ogg.html VorbisGain] to apply [[ReplayGain]] on Vorbis files (Win32)
** Instructions to integrate VorbisGain into foobar2000, Winamp, and Windows Explorer can be found in [http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=41880&view=findpost&p=396612 this HA thread] ''A precompiled script of the procedure (in RAR format) can be found in [http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=45196&view=findpost&p=397803 this HA posting]''
** RPM packages for VorbisGain are available [http://rpm.pbone.net/index.php3?stat=3&search=vorbisgain&srodzaj=3 here], and source code for VorbisGain is available [http://sjeng.org/vorbisgain.html here]
* [http://www.rarewares.org/quantumknot/vorbisgain.gz VorbisGain Static GCC 4 compile] (Posix)
* [[foobar2000]]'s ''ReplayGain scanner'' supports (Ogg) Vorbis files
* [[Winamp]]'s ''ReplayGain Analyzer'' supports (Ogg) Vorbis files
* [[MediaMonkey]]'s ''Volume Levelling'' supports (Ogg) Vorbis files

=== Splitters ===
The following utilities are used to splice Vorbis streams without decoding/re-encoding.

* [http://www.free-codecs.com/download/Ogg_Cutter.htm Ogg Cutter] (Win32)
* [http://mp3splt.sourceforge.net/ mp3splt] (Win32)
* [http://www.xiph.org/downloads/ vcut] (CLI tool part of the official vorbis-tools package) (Win32/Posix)
* [http://sourceforge.net/projects/ogg-cut/ (Ogg) Vorbis Stream Cutter (ogg-cut)] (Posix)

=== Taggers ===
Most tagger supporting (Ogg) Vorbis are listed in [[download page#Tagging Utilities|the download page]].
* [http://www.rarewares.org/files/ogg/vorbiscomment-1.1.1.zip Vorbis Comment]

== Supported digital audio players ==
The following list contains some players that support Vorbis playback.
* [[Apple iPod]] with [[Rockbox]] firmware – check out this [http://www.hydrogenaudio.org/forums/index.php?s=32eeac65958144db631c8a739b41983c&showtopic=40992 HA thread]
* [http://www.ifreemax.com/ FreeMax] FW-960
* [http://www.iaudiophile.net/ iAudio] [[IAudio M3|M3]], M5, U2, G3, X5, I5, 7, D2, F2, T2, A3, Q5W, A2; VorbisGain support only on X5/M5 with [[Rockbox]] firmware
* [[iRiver H-Series]] with [[Rockbox]] firmware
* [[MPIO H-Series]]
* [[Neuros]] with [[Rockbox]] firmware
* [[Rio Karma]]
* [http://www.samsung.com/Products Samsung]
* [http://www.slimdevices.com/ Slim Devices: Squeezebox] External player
* [http://www.sony.com/electronics/walkman-digital-music-players/t/walkman Sony Walkman]
* [http://www.yepp.co.kr/ Yepp] YP-T6, YP-T7, YP-C1, YP-F1, YP-53 (Firmware 1.200), other..

A longer list can be found at [http://wiki.xiph.org/index.php/PortablePlayers xiph's wiki].

'''Important note:''' There may be players out there that support (Ogg) Vorbis, although they are not marketed as such.

== External links ==
The following links contain information surrounding the (Ogg) Vorbis codec that can be found on Hydrogenaudio and elsewhere throughout the web.

=== Hydrogenaudio Wiki ===
* [[Ogg]] (Container)
* [[Listening Tests#Multiformat Tests|Listening tests comparing Vorbis against MP3, AAC, WMA, etc.]]
* [[Recommended Ogg Vorbis|Recommended settings for encoding with Vorbis]] and its related [http://www.hydrogenaudio.org/forums/index.php?showtopic=15049 HA thread]
* [[EAC and Ogg Vorbis|Configuring EAC and Vorbis as an external command-line encoder]]

=== Websites ===
* [http://www.vorbis.com Vorbis official website] (updated continually)
* [http://en.wikipedia.org/wiki/Vorbis Vorbis at Wikipedia]
* [http://www.playogg.org/ PlayOgg initiative] by the [http://www.fsf.org/ Free Software Foundation]
* [http://www.audiocoding.com/modules/wiki/?page=Ogg+Vorbis (Ogg) Vorbis at AudioCoding]
* [http://www.rarewares.org/ogg.html (Ogg) Vorbis binaries at Rarewares]
* [http://www.geocities.jp/aoyoume/aotuv/ Aoyumi's homepage of tuned versions of Vorbis encoder and current beta binaries]
* [http://homepage3.nifty.com/blacksword/index_e.htm The (Ogg) Vorbis Acceleration Project] – Archer/[[Lancer]] homepage for optimized versions of aoTuV Vorbis encoder and other SSE optmizations
* [http://www.xiph.org/ Xiph.org Foundation]

=== Scientific/R&D ===
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=20132&st=0 Noise Normalization and HF Boost problem solution that ultimatly lead to the aoTuV tunings (HA Thread)]
* [http://www.free-comp-shop.com/vorbis.pdf Keith Wright rendition of understanding the MDCT in Vorbis by defining it's basic trig properties (PDF)]
* [http://www.mp3-tech.org/programmer/docs/embedded_vorbis_thesis.pdf (Ogg) Vorbis decoder for an embedded system (Master Thesis in PDF)]
* [http://wiki.xiph.org/index.php/Bounties Xiph.org Vorbis bounties]

{{navbox audio codecs}}

[[Category:Lossy]]

MP3

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12.218.147.66: /* Pros */ Patents have expired. There is no more licensing program.

'''MPEG-1 Audio Layer 3''', more commonly referred to as MP3, is a popular digital audio encoding and lossy compression format, designed to greatly reduce the amount of data required to represent audio, yet still sound like a faithful reproduction of the original uncompressed audio to most listeners. It was invented by a team of European engineers who worked in the framework of the EUREKA 147 DAB digital radio research program, and it became an ISO/IEC standard in 1991.

== History ==
The MP3 algorithm development started in 1987, with a joint cooperation of [http://www.iis.fraunhofer.de/ Fraunhofer IIS-A] and the University of Erlangen. It is standardized as ISO-MPEG Audio Layer-3 (IS 11172-3 and IS 13818-3).

It soon became the de facto standard for lossy audio encoding, due to the high [[compression rates]] (1/11 of the original size, still retaining considerable quality), the high availability of decoders and the low CPU requirements for playback. (486 DX2-100 is enough for real-time decoding)

It supports [[multichannel]] files (see [http://www.mp3surround-format.com/ page]), [[sampling rate]]s from 16 kHz to 24 kHz (MPEG2 Layer 3) and 32 kHz to 48 kHz (MPEG1 Layer 3)

Formal and informal listening tests have shown that MP3 at the 160-224 kbps range provide encoded results indistinguishable from the original materials in most of the cases.

== Encoding and decoding ==
=== Encoding of MP3 audio ===
The MPEG-1 standard does not include a precise specification for an MP3 encoder. The decoding algorithm and file format, as a contrast, are well defined. Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information in the raw audio (or rather its MDCT representation in the frequency domain). During encoding 576 time domain samples are taken and are transformed to 576 frequency domain samples. If there is a transient 192 samples are taken instead of 576. This is done to limit the temporal spread of quantization noise accompanying the transient.

This is the domain of psychoacoustics: the study of subjective human perception of sounds.

As a result, there are many different MP3 encoders available, each producing files of differing quality. Comparisons are widely available, so it is easy for a prospective user of an encoder to research the best choice. It must be kept in mind that an encoder that is proficient at encoding at higher bitrates (such as LAME, which is in widespread use for encoding at higher bitrates) is not necessarily as good at other, lower bitrates.

=== Decoding of MP3 audio ===
Decoding, on the other hand, is carefully defined in the standard. Most decoders are "bitstream compliant", meaning that the decompressed output they produce from a given MP3 file will be the same (within a specified degree of rounding tolerance) as the output specified mathematically in the ISO/IEC standard document. The MP3 file has a standard format which is a frame consisting of 384, 576, or 1152 samples (depends on MPEG version and layer) and all the frames have associated header information (32 bits) and side information (9, 17, or 32 bytes, depending on MPEG version and stereo/mono). The header and side information help the decoder to decode the associated Huffman encoded data correctly.

Therefore, for the most part, comparison of decoders is almost exclusively based on how computationally efficient they are (i.e., how much memory or CPU time they use in the decoding process).

== MP3 file structure ==
[[Image:MP3 file structure.png|thumb|right|500px|Breakdown of an MP3 File's Structure]]
An MP3 file is made up of multiple MP3 frames which consist of the MP3 header and the MP3 data. This sequence of frames is called an Elementary stream. Frames are independent items: one can cut the frames from a file and an MP3 player would be able to play it. The MP3 data is the actual audio payload. The diagram shows that the MP3 header consists of a sync word which is used to identify the beginning of a valid frame. This is followed by a bit indicating that this is the MPEG standard and two bits that indicate that layer 3 is being used, hence MPEG-1 Audio Layer 3 or MP3. After this, the values will differ depending on the MP3 file. The range of values for each section of the header along with the specification of the header is defined by ISO/IEC 11172-3.

Most MP3 files today contain ID3 metadata which precedes or follows the MP3 frames; this is also shown in the diagram.

===VBRI, XING, and LAME headers===
MP3 files often begin with a single frame of silence which contains an extra header that, when supported by decoders, results in the entire frame being treated as informational instead of being played (although some are known to do both). The extra header is in the frame's data section, before the actual silent audio data, and was originally intended to help with the playback of VBR files.

Xing and Fraunhofer each developed their own formats for this header. The Xing-format header is just called the ''Xing header'' or ''XING header''. The Fraunhofer-format header is called the ''VBRI header'' or ''VBR Info header''.

====Seek table====
Both formats specify a table of seek points, which help players correlate playback position (e.g., in seconds, or as a percentage) with byte offsets in the file.

====Gapless playback info====
In addition to the seek-point table, the Fraunhofer format contains a combined encoder delay & padding value (measured in samples), which can assist [[gapless playback]]. The encoder delay value is the number of samples added to the beginning of the audio data, and the encoder padding value is the number of samples added to the end. There's also a decoder delay, usually 529 samples of junk samples added to the beginning by the decoder. To determine the starting and ending samples of the non-delay, non-padding portion of the decoder output, MP3 players can perform the following calculation:

<nowiki>
gapless_range_start = encoder_delay + decoder_delay
if encoder_padding < decoder_delay:
gapless_range_end = total_samples
else:
gapless_range_end = total_samples - encoder_padding + decoder_delay</nowiki>

Alternatively, when <code>encoder_padding</code> < <code>decoder_delay</code>, a player could feed an extra MP3 frame to the decoder (e.g. a silent frame, or the first frame of the next MP3 in a sequence), and then use the second <code>gapless_range_end</code> calculation. At least one player (Rockbox) does the latter to handle an uncommon type of MP3 encoded specially for gapless playback, where one long stream is split up and written into separate files.

====LAME tag====
The [[LAME]] encoder extends the Xing header format. This modified header is sometimes called a ''LAME header'' or ''LAME tag'', although the actual LAME tag is only the LAME-specific data embedded in unused space in the header.

When the header was first added in LAME 3.12, the LAME tag contained only a 20-byte LAME version string. In LAME 3.90, this region was expanded to hold additional data, such as:
* audio and info tag CRCs
* separate delay & padding values for gapless playback
* various encoder settings (expanded in LAME 3.94 to include presets)
The modified header is also included in CBR files (effective LAME 3.94), with "Info" instead of "Xing" near the beginning.

====Specs====
The Fraunhofer VBRI header and the LAME tag have explicit specifications. The Xing format can only be inferred from the C code the company provided to read the headers. Here are links to the code and specs:
* [http://gabriel.mp3-tech.org/mp3infotag.html LAME MP3 Info Tag spec]
* [http://www.all4mp3.com/tools/tech-and-tools.php All4mp3 mp3 Tech & Tools downloads] - official distribution site for Fraunhofer's ''Source code to add VBRi header to mp3 file'' (contains header spec) and ''MP3 VBR-Header SDK'' (header-reading C code sample)
* [http://www.mp3-tech.org/programmer/sources/vbrheadersdk.zip Xing Variable Bitrate MP3 Playback SDK]
* [http://mp3decoders.mp3-tech.org/decoders_lame.html#delays Information about common encoder and decoder delays]

== Technical information ==
=== Codec block diagram ===
A basic functional block diagram of the MPEG1 layer 3 audio codec is as shown below.
[[Image:Layer3_block.png|frame|center|Block diagram of the MPEG1 layer 3 audio]]

=== The hybrid polyphase filterbank ===

The polyphase [[filterbank]] is the key component common to all layers of MPEG1 audio compression. The purpose of the polyphase filterbank is to divide the audio signal into 32 equal-width [[frequency]] [[subband]]s, by using a set of [[bandpass filters]] covering the entire audio frequency range (a set of 512 tap FIR Filters).

====Polyphase Filterbank Formula====
[[Image:Poly_samples.png|frame|center|Polyphase filterbank]]

Audio is processed by frames of 1152 samples per audio channel. The polyphase filter groups 3 groups of 12 samples (3x12=36) samples per subband as seen from the picture above (3x12x32 subbands=1152 samples).

The polyphase filter bank and its inverse are not [[lossless]] transformations. Even without [[quantize|quantization]], the inverse transformation cannot perfectly recover the original signal. However by design the error introduced by the filter bank is small and inaudible. [[Image:Mdct.png|frame|center|MDCT]] MDCT formula: <math> X(m)= \sum_{k=0}^{n-1}f(k)x(k)\cos [{ {\pi \over {2n}} ({2k+1+{n \over 2}})({2m+1})}],~m=0 ... {n \over 2}-1</math> 

Layer 3 compensates for some of the filter bank deficiencies by processing the filter bank output with a Modified Discrete Cosine Transform ([[MDCT]]). The polyphase [[filterbank]] and the MDCT are together called as the hybrid filterbank. The hybrid filterbank adapts to the signal characteristics (block switching depending on the signal etc.).

The 32 [[subband]] signals are subdivided further in frequency content by applying a 18-spectral point or 6-spectral point MDCT. Layer 3 specifies two different MDCT block lengths: a long block (18 spectral points) or a short block (6 spectral points).

Long blocks have a higher frequency resolution. Each subband is transformed into 18 spectral coefficients by MDCT, yielding a maximum of 576 spectral coefficients (32x18=576 spectral lines) each representing a bandwidth of 41.67Hz at 48kHz sampling rate. At 48kHz sampling rate a long block has a time resolution of about x ms. There is a 50% overlap between successive transform windows, so the window size is 36 for long blocks.

Short blocks have a higher time resolution. Short block length is one third of a long block and used for transients to provide better time (temporal) resolution. Each subband is transformed into 6 spectral coefficients by MDCT, yielding a maximum of 192 spectral coefficients (32x6=192 spectral lines) each representing a bandwidth of 125Hz at 48kHz [[sampling rate]]. At 48kHz sampling rate a short block has impulse response of 18.6ms. There is a 50% overlap between successive transform windows, so the window size is 12 for short blocks.

Time resolution of long blocks and time resolution of short blocks are not constants, but jitter depending on the position of the sample in the transformed block. See [http://hydrogenaudio.org/musepack/klemm/www.personal.uni-jena.de/~pfk/mpp/timeres.html here] for diagrams showing the average time resolutions of different codecs.

[[Image:Freqlines.png|center|frame|Psychoacoustic-MDCT]]

Block switching ([[MDCT]] window switching) is triggered by [[Psychoacoustic|psycho acoustics]].

For a given frame of 1152 samples, the MDCT's can all have the same block length (long or short) or have a mixed-block mode (mixed-block mode for Lame is in development).

Unlike only the polyphase [[filterbank]], without quantization the MDCT transformation is [[lossless]].

Once the MDCT converts the audio signal into the [[frequency domain]], the [[aliasing]] introduced by the subsampling in the filterbank can be partially cancelled. The decoder has to undo this so that the inverse MDCT can reconstruct the [[subband]] samples in their original aliased form for reconstruction by the synthesis filterbank.

=== The psychoacoustic model ===

This section is a work in progress. It is incomplete and data is still being gathered.

==== Concepts ====
;[[Critical band]]s
: Much of what is done in simultaneous [[masking]] is based on the existence of critical bands. The hearing works much like a non-uniform filterbank, and the critical bands can be said to approximate the characteristics of those filters. Critical bands does not really have specific "on" and "off" frequencies, but rather width as a function of [[frequency]] - critical [[bandwidth]]s.

;Tonality estimation

;Spreading function
: Masking does not only occur within the [[critical band]], but also spreads to neighboring bands. A spreading function SF(z,a) can be defined, where z is the frequency and a the amplitude of a masker. This function would give a masking threshold produced by a single masker for neighboring frequencies. The simplest function would be a triangular function with slopes of +25 and -10 dB / [[Bark]], but a more sophisticated one is highly nonlinear and depends on both frequency and amplitude of masker.

;Simultaneous masking
: Simultaneous [[masking]] is a frequency domain phenomenon where a low level signal, e.g, a smallband noise (the maskee) can be made inaudible by simultaneously occurring stronger signal (the masker), e.g, a pure tone, if masker and maskee are close enough to each other in frequency. A masking threshold can be measured below which any signal will not be audible. The masking threshold depends on the sound pressure level (SPL) and the frequency of the masker, and on the characteristics of the masker and maskee. The slope of the masking threshold is steeper towards lower frequencies,i.e., higher frequencies are more easily masked.

: Without a masker, a signal is inaudible if its SPL is below the threshold of quiet, which depends on frequency and covers a dynamic range of more than 60 dB. We have just described masking by only one masker. If the source signal consists of many simultaneous maskers, a global masking threshold can be computed that describes the threshold of just noticeable distortions as a function of frequency. The calculation of the global masking threshold is based on the high resolution short term [[frequency|amplitude]] spectrum of the audio or speech signal, sufficient for critical band based analysis, and is determined in audio coding via 512 or 1024 point FFT. In a first step all individual masking thresholds are calculated, depending on signal level, type of masker(noise or tone), and frequency range. Next the global masking threshold is determined by adding all individual thresholds and the threshold in quiet (adding this later threshold ensures that the computed global masking threshold is not below the threshold in quiet). The effects of masking reaching over [[critical band]] bounds must be included in the calculation. Finally the global signal-to-mask ratio (SMR) is determined as the ratio of the maximum of signal power and global masking threshold.

;Temporal masking
: In addition to simultaneous [[masking]] two [[time domain]] phenomena also play an important role in human auditory perception, pre-masking and post-masking. The temporal masking effects occur before and after a masking signal has been switched on and off, respectively. The duration when pre-masking applies is less than -or as newer results indicate, significantly less than-one tenth that of the post-masking, which is in the order of 50 to 200 msec. Both pre and post-masking are being exploited in the ISO/MPEG audio coding algorithm.

: It uses either a separate [[filterbank]] or combines the calculation of energy values (for the masking calculations) and the main filter bank. The output of the perceptual model consists of values for the masking threshold or the allowed noise for each coder partition. If the quantization noise can be kept below the masking threshold, then the compression results should be indistinguishable from the original signal.

;[[ATH]]

;[[Masking]] threshold
: Masking raises the threshold of hearing, and compressors take advantage of this effect by raising the noise floor, which allows the audio waveform to be expressed with fewer bits. The noise floor can only be raised at [[frequency|frequencies]] at which there is effective masking.

: The equal widths of the [[subband]]s do not accurately reflect the human auditory system's frequency dependent behavior. The width of a "[[critical band]]" as a function of frequency is a good indicator of this behavior. Many psychoacoustic effects are consistent with a critical band frequency scaling. For example, both the perceived loudness of a signal and its audibility in the presence of a masking signal is different for signals within one critical band than for signals that extend over more than one critical band. Figure 2 compares the polyphase filter [[bandwidth]]s with the width of these critical bands. At lower frequencies a single subband covers several critical bands.

==== Simplified overview of the psychoacoustic model ====
* Perform a 1024-sample [[FFT]]s on each half of a frame (1152 samples) of the input signal, selecting the lower of the two masking thresholds to use for that subband.
* Each frequency bin is mapped to its corresponding critical band.
* Calculate a tonality index, a measure of whether a signal is more tone-like or noise-like.
* Use a defined spreading function to calculate the masking effect of the signal on neighbouring [[critical band]]s.
* Calculate the final masking threshold for each subband, using the tonality index, the output of the spreading function, and the [[ATH]].
* Calculate the signal-to-mask ratio for each [[subband]], and passes information on to the [[quantize|quantizer]].

==== More detailed overview the psychoacoustic model====
The MPEG/audio algorithm compresses the audio data in large part by removing the acoustically irrelevant parts of the audio signal. That is, it takes advantage of the human auditory system's inability to hear quantization noise under conditions of auditory masking. This masking is a perceptual property of the human auditory system that occurs whenever the presence of a strong audio signal makes a temporal or spectral neighborhood of weaker audio signals imperceptible. A variety of psychoacoustic experiments corroborate this masking phenomenon.

Empirical results also show that the human auditory system has a limited, [[frequency]] dependent, resolution. This frequency dependency can be expressed in terms of critical band widths which are less than 100Hz for the lowest audible frequencies and more than 4kHz at the highest. The human auditory system blurs the various signal components within a critical band although this system's frequency selectivity is much finer than a critical band.

The psychoacoustic model analyzes the audio signal and computes the amount of noise [[masking]] available as a function of frequency. The masking ability of a given signal component depends on its frequency position and its loudness. The encoder uses this information to decide how best to represent the input audio signal with its limited number of code bits. The MPEG/audio standard provides two example implementations of the psychoacoustic model.

Below is a general outline of the basic steps involved in the psychoacoustic calculations for either model. Differences between the two models will be highlighted.

* Time align audio data. There is one psychoacoustic evaluation per frame. The audio data sent to the psychoacoustic model must be concurrent with the audio data to be coded. The psychoacoustic model must account for both the delay of the audio data through the [[filterbank]] and a data offset so that the relevant data is centered within the psychoacoustic analysis window.
* Convert audio to a [[frequency]] domain representation. The psychoacoustic model should use a separate, independent, time-to-frequency mapping instead of the polyphase filter bank because it needs finer frequency resolution for an accurate calculation of the masking thresholds.

Layer II and III use a 1,152 sample frame size so the 1,024 sample window does not provide complete coverage. While ideally the analysis window should completely cover the samples to be coded, a 1,024 sample window is a reasonable compromise. Samples falling outside the analysis window generally will not have a major impact on the psychoacoustic evaluation.

For Layers II and III, the model computes two 1,024 point psychoacoustic calculations for each frame. The first calculation centers the first half of the 1,152 samples in the analysis window and the second calculation centers the second half. The model combines the results of the two calculations by using the higher of the two signal-to-mask ratios for each [[subband]]. This in effect selects the lower of the two noise masking thresholds for each subband.

* Process spectral values in groupings related to critical band widths. To simplify the psychoacoustic calculations, both models process the frequency values in perceptual quanta.

Psychoacoustic model 2 never actually separates tonal and non-tonal components. Instead, it computes a tonality index as a function of frequency. This index gives a measure of whether the component is more tone-like or noise-like. Model 2 uses this index to interpolate between pure tone-masking-noise and noise-masking-tone values. The tonality index is based on a measure of predictability. Model 2 uses data from the previous two analysis windows to predict, via linear extrapolation, the component values for the current window. Tonal components are more predictable and thus will have higher tonality indices. Because this process relies on more data, it is more likely to better discriminate between tonal and non-tonal components than the model 1 method.

* Apply a spreading function. The [[masking]] ability of a given signal spreads across its surrounding [[critical band]]. The model determines the noise masking thresholds by first applying an empirically determined masking (model 1) or spreading function (model 2) to the signal components.

* Set a lower bound for the threshold values. Both models include an empirically determined absolute masking threshold, the threshold in quiet. This threshold is the lower bound on the audibility of sound.

* Find the masking threshold for each [[subband]]. Model 2 selects the minimum of the masking thresholds covered by the subband only where the band is wide relative to the critical band in that [[frequency]] region. It uses the average of the masking thresholds covered by the subband where the band is narrow relative to the critical band. Model 2 is not less accurate for the higher frequency subbands because it does not concentrate the non-tonal components.

* Calculate the signal-to-mask ratio. The psychoacoustic model computes the signal-to-mask ratio as the ratio of the signal energy within the subband (or, for Layer III , a group of bands) to the minimum masking threshold for that subband. The model passes this value to the bit (or noise) allocation section of the encoder.

==== Model 2 technical details ====


The psychoacoustic model calculates just-noticeable distortion (JND) profiles for each band in the [[filterbank]]. This noise level is used to determine the actual quantizers and quantizer levels. There are two psychoacoustic models defined by the standard. They can be applied to any layer of the MPEG/Audio algorithm. In practice however, Model 1 has been used for Layers I and II and Model 2 for Layer III. Both models compute a signal-to-mask ratio (SMR) for each band (Layers I and II) or group of bands (Layer III).

The more sophisticated of the two, Model 2, will be discussed. The steps leading to the computation of the JND profiles is outlined below.

;1. Time-align audio data

The psychoacoustic model must estimate the [[masking]] thresholds for the audio data that are to be [[quantize|quantized]]. So, it must account for both the delay through the filterbank and a data offset so that the relevant data is centered within the psychoacoustic analysis window. For the Layer III algorithm, time-aligning the psychoacoustic model with the filterbank demands that the data fed to the model be delayed by 768 samples.

;2. Spectral analysis and normalization.

A high-resolution spectral estimate of the time-aligned data is essential for an accurate estimation of the masking thresholds in the [[critical band]]s. The low frequency resolution of the filterbank leaves no option but to compute an independent time-to-frequency mapping via a fast Fourier Transform ([[FFT]]). A Hanning window is applied to the data to reduce the edge effects of the transform window.

Layer III operates on 1152-sample data frames. Model 2 uses a 1024- point window for spectral estimation. Ideally, the analysis window should completely cover the samples to be coded. The model computes two 1024-point psychoacoustic calculations. On the first pass, the first 576 samples are centered in the analysis window. The second pass centers the remaining samples. The model combines the results of the two calculations by using the more stringent of the two JND estimates for bit or noise allocation in each [[subband]].

Since playback levels are unknown3, the sound-pressure level (SPL) needs to be normalized. This implies clamping the lowest point in the absolute threshold of hearing curves to +/- 1-bit [[frequency|amplitude]].

;3. Grouping of spectral values into threshold calculation partitions.

The uniform [[frequency]] decomposition and poor selectivity of the filterbank do not reflect the response of the ear's Basilar Membrane. To accurately model the masking phenomenon characteristic of the Basilar Membrane, the spectral values are grouped into a large number of partitions. The exact number of threshold partitions depends on the choice of sampling rate. This transformation provides a resolution of approximately either 1 FFT line or 1/3 critical band, whichever is smaller. At low frequencies, a single line of the FFT will constitute a partition, while at high frequency|frequencies many lines are grouped into one.

;4. Estimation of tonality indices.

It is necessary to identify tonal and non-tonal (noise-like) components because the masking abilities of the two types of signals differ. Model 2 does not explicitly separate tonal and non-tonal components. Instead, it computes a tonality index as a function of frequency. This is an indicator of the tone-like or noise-like nature of the spectral component. The tonality index is based on a measure of predictability. Linear extrapolation is used to predict the component values of the current window from the previous two analysis windows. Model 2 uses this index to interpolate between pure tone-masking-noise and noise-masking-tone values. Tonal components are more predictable and thus have a higher tonality index. As this process has memory, it is more likely to discriminate better between tonal and non-tonal components, unlike psychoacoustic Model 116.

;5. Simulation of the spread of masking on the Basilar Membrane.

A strong signal component affects the audibility of weaker components in the same critical band and the adjacent bands. Model 2 simulates this phenomenon by applying a Spreading function to spread the energy of any critical band into its surrounding bands. On the [[Bark]] scale, the spreading function has a constant shape as a function of partition number, with slopes of +25 and –10 dB per Bark.

;6. Set a lower bound for the threshold values.

An empirically determined absolute [[masking]] threshold, the threshold in quiet, is used as a lower bound on the audibility of sound.

;7. Determination of masking threshold per [[subband]].

At low [[frequency|frequencies]], the minimum of the masking thresholds within a subband is chosen as the threshold value. At higher frequencies, the average of the thresholds within the subband is selected as the masking threshold. Model 2 has the same accuracy for the higher subbands as for low frequency ones because it does not concentrate non-tonal components16.

;8. [[Pre echo]] detection and window switching decision.

;9. Calculation of the signal-to-mask ratio (SMR).

SMR is calculated as a ratio of signal energy within the subband (for Layers I and II) or a group of subbands (Layer III) to the minimum threshold for that subband. This is the final output of the psychoacoustic model.

The masking threshold computed from the spread energy and the tonality index.

== Pros and cons ==
=== Pros ===
* Widespread acceptance, support in nearly all hardware audio players and devices
* An [[ISO]] standard, part of MPEG specs
* Fast decoding, lower complexity than [[Advanced Audio Coding|AAC]] or [[Vorbis]]
* Anyone can create their own implementation (Specs and demo sources available)
* Since the patents have expired, MP3 is in the public domain.

=== Cons ===
* Lower performance/efficiency than modern codecs.
* Problem cases that trip out all transform codecs.
* Sometimes, maximum bitrate (320kbps) isn't enough.
* Unusable for high definition audio (sampling rates higher than 48kHz).

== See also ==
=== Techniques used in compression ===
* [[Huffman coding]]
* [[Quantization]]
* [[Joint stereo|M/S matrixing]]
* [[Intensity stereo]]
* [[Channel coupling]]
* Modified discrete cosine transform ([[MDCT]])
* Polyphase filter bank

There is a non-standardized form of MP3 called [[MP3Pro]], which takes advantage of [[SBR]] encoding to provide better quality at low bitrates.

=== Encoders/decoders (supported platforms) ===
* [[LAME]] (Win32/Posix)
* [[Audioactive]] (Win32)
* [[Blade]] (Win32/Posix)
* [[Xing]] (Win32)
* [[Gogo]] (Win32/Posix)

=== Metadata (tags) ===
* [[ID3v1]]
* [[ID3v1.1]]
* [[ID3v2]]

== Further reading and bibliography ==
* [[Best MP3 Decoder]]
* [[High-frequency content in MP3s]]

== External links ==
* <s>Roberto's listening test</s> featuring MP3 encoders
* [http://en.wikipedia.org/wiki/Mp3 MP3 at Wikipedia]

[[Category:Codecs]]
[[Category:Lossy]]
[[Category:MP3]]

Best MP3 Decoder

2020-06-06T05:20:49Z

12.218.147.66: Decoders, not encoders.

Since what the [[MP3]] standard really defines is a decoder, there should be no such thing as a better or a worse MP3 decoder. As long as a decoder is accurate, it'll output the expected quality.

All the most popular decoders these days output accurate enough streams. These include:

* [[Winamp]]'s FhG decoder
* [http://www.mpg123.de mpg123], probably one of the most popular decoders, also used in [[foobar2000]], Otachan's winamp plugin, [[LAME]] and countless other MP3 players
* [http://mpadec.sourceforge.net/ MPAdec], a very accurate decoder based on mpg123
* [http://koti.welho.com/hylinen/apollo/ Apollo] (broken link), the most mathematically accurate decoder
* [http://www.underbit.com/products/mad/ MAD], another very precise GPLd MP3 decoder
** A MAD input-plugin for WinAmp is available in [http://www.hydrogenaudio.org/forums/index.php?showtopic=46824&hl= this HA thread]

Using any of the above should be enough for all purposes. Quality differences among them are insignificant.

Older decoders could provide bad quality, due mostly to encoder implementation errors. Such bad decoders include Winamp before version 2.666 (using Playmedia's low quality AMP engine), decoders based on old XAudio, and Digideck. Also, some older decoders did not work well with VBR streams.

[[Category:MP3]]
[[Category:Software]]