Hydrogenaudio Knowledgebase - User contributions [en]

Recommended Ogg Vorbis

2006-01-12T04:05:00Z

Lys: /* Optimized binaries */

[[image:fish_logo.png|right|]]
[[Ogg Vorbis|Vorbis]], being a continuously developed standard, improves all the time. Plus, being an open (i.e. patent-free) standard, it has many 'third-parties' that produces, and most of the time, improves, its utilities.

Here you can find some guidelines on which utilities to use, and what settings will provide you with the best quality.

=History=

(For a highly detailed description of Vorbis history, check out OggZealot's [http://www.hydrogenaudio.org/forums/index.php?showtopic=15274&view=findpost&p=153268 Ogg Vorbis Historic] where Monty also adds a few more details too). Ogg Vorbis reached version 1.0 in '''July 2002'''. It is the official [http://www.xiph.org/ Xiph.org] encoder ''(the one you get from vorbis.com)''. HA codec developer, Garf, did his own tunings, based on version 1.0 to produce GT3b1 and GT3b2. Both encoders showed improved [[pre echo]] handling for ''q'' values of ''5 to 10''. It was later judged in an internal listening test that GT3b1 was the better of the two. There was a minor bugfix update earlier that year in March, which only appeared in the CVS at Xiph.org. This consisted of very minor bug fixes, which do more to correct odd problems that may occur rather than improving quality, including ''(garbled noise output and gaps in streams)''. This was referred to as Post 1.0 CVS. Quality problems that mainly affected low bitrates were later addressed in a new bugfix ''(1.0.1)'' that was released.

Post 1.0.1 CVS was released late '''December 2003''' by Monty at Xiph, and includes a true [[CBR]] template. In order to simplify the situation where we had two encoders ''(1.0.1 and GT3b1)'', OggDropXPd developer John33 merged the sources to give us GT3b2. Once the 128 kbps multiformat test was completed, Aoyumi's [[aoTuV]] Vorbis tuning was determined to be the best Vorbis encoder. After the success of aoTuV beta 2 encoder, Xiph.Org merged their tunings into the official CVS branch to produce the long-awaited Vorbis 1.1. Aoyumi's later release of '''aoTuV beta 4''' encoder as of '''November 2005''' significantly improves Vorbis' quality while enhancing the compression. It is now the recommended encoder by 3rd party developers at Hydrogenaudio. Aoyumi's latest is '''aoTuV beta 4.5''' (later bugfixed with '''aoTuV beta 4.51''') released in '''December 2005''', which improves low bit-rate quality even more. However, it is still undergoing peer review before it becomes the recommended encoder. Near the end of 2005, after aoTuV beta 4.51 is released, Xiph.org released '''Vorbis 1.1.2''', which only provides bugfixes to Vorbis 1.1.1. So, although this is the 'official' version, it is '''not recommended'''.

''Note: Xiph.org does not maintain the binaries, but rather provides newest updates and releases to libogg and libvorbis, etc.''

=Recommended Vorbis Encoders=

(adapted from [http://www.hydrogenaudio.org/forums/index.php?showtopic=15049 Recommended Encoder and Settings] post compiled by QuantumKnot)

==Windows binaries==

John33's oggenc2.8 is a special version of the Ogg Vorbis encoder. "Features include compression from lossless files (Monkeys Audio, LPAC, FLAC, OptimFROG, WavPack and Shorten - requires presence of decoders), and the ability to specify 'padding' in the headers for subsequent insertion of Tags." (from Ogg Vorbis page at rarewares)

* '''[http://www.rarewares.org/dancer/dancer.php?f=59 aoTuV beta 4.51 (MSVC compile)]'''
* '''[http://www.rarewares.org/dancer/dancer.php?f=60 aoTuV beta 4.51 (ICL compile for P3/AMD Athlon)]'''
* '''[http://www.rarewares.org/dancer/dancer.php?f=61 aoTuV beta 4.51 (ICL compile for P4/AMD Sempron, Athlon 64))]'''

If you prefer a nice drag-and-drop interface, then you can try John33's OggDropXPd (Windows only). "Features include compression from lossless files ([[Monkey's Audio]], [[LPAC]], [[FLAC]] and [[OptimFROG]]), auto-tagging, renaming of encoded files, setting of advanced encoder parameters, use of VorbisGain tags on decode, Playlist (.m3u) creation, and others." (from Ogg Vorbis page at rarewares)

* '''[http://www.rarewares.org/dancer/dancer.php?f=50 OggDropXPd aoTuV beta 4.51 (MSVC compile)]'''
* '''[http://www.rarewares.org/dancer/dancer.php?f=51 OggDropXPd aoTuV beta 4.51 (ICL compile for P3/AMD Athlon)]'''
* '''[http://www.rarewares.org/dancer/dancer.php?f=52 OggDropXPd aoTuV beta 4.51 (ICL compile for P4/AMD Sempron, Athlon 64)]'''

''(work in progress, guide will be completed soon)''

==Linux binaries==

* '''[http://www.rarewares.org/quantumknot/oggenc-aotuv451.gz oggenc aoTuV beta 4.51 and libogg 1.1.3 (static gcc 4 compile)]''' '''with impulse_trigger_profile'''

The static GCC 4 binary was compiled by QuantumKnot.

==3rd party source code==

* '''[http://www.geocities.jp/aoyoume/aotuv/ aoTuV Homepage]''' -- contains modified source code written in C and downloads of binaries.

The source-code contains modifications mostly to the psychacoustics model and bitrate allocation, i.e ''psy.c'' and ''bitrate.c''

These are other Ogg Vorbis encoders that were tuned by 3rd party developers (outside of Xiph.Org).

'''aoTuV beta 4''' '''Now the recommended encoder'''

Developed by Aoyumi and based on libvorbis 1.1.1, many people have reported this encoder to give better quality at low to medium bitrates. It includes a -q -2 option for the lowest bitrate. According to guruboolez' latest listening test on classical music, aoTuV beta 4 performed magnificently well at -q 6!! ''(see Aoyumi's website for information above)''

'''aoTuV beta 4.51''' '''The latest encoder'''

Many HydrogenAudio enthusiasts report that this version gives even better quality for low bit-rates. -q 1.5 becomes acceptable, even good enough, and -q 4 is now near-transparent! However, more peer review is needed before this version becomes the recommended encoder.

==Optimized binaries==

These are highly optimized encoders developed by the '''[http://homepage3.nifty.com/blacksword/index_e.htm Ogg Vorbis Acceleration Project]''' that are much faster than the standard binary builds with negligible to nearly no effects on audio quality, including sped-up routines, i.e ''mdct.c''

* '''[http://homepage3.nifty.com/blacksword/oggenc2_lancer20050709.zip oggenc2.6 aoTuV beta 4 (Lancer 20050709) (Win32 Build)]'''
* '''[http://homepage3.nifty.com/blacksword/oggenc2_lancer20051121.zip oggenc2.6 aoTuV beta 4.51 (Lancer 20051121) (Win32 Build)]''' -- ''Latest version''

See [[Lancer]] Information and downloads page above.

''Note: Output of Lancer may be slightly different from output of 'standard' aoTuV. This is due to the difference of floating-point rounding: Lancer uses 64-bit SSE, while standard aoTuV uses 80-bit FP. This difference, however, should not be audible at all. In fact, tests have proven that playback of Lancer's output is indistinguishable from playback of standard aoTuV output.''

=Recommended Encoder Settings=

'''Refer to the table below. For best results, start at -q 4 and''' [[ABX]] '''your way up.''' 
''Ultimately, the best -q setting will depend on your specific needs. Feel free to experiment.'' 


'''General Command Line Usage:'''

'''oggenc -q n'' inputfile.wav'' '''

:'''where n is a number from -1 to 10''' e.g. '''oggenc -q 5 ''inputfile.wav'' ''' '' '''Note:''' Most standard oggenc binaries can input FLACs as well ! ''


''It should be noted that Vorbis supports fractional numbers, e.g. 5.50 (or 5,50, depending on where the tool is compiled.) ''

{| border="0" style="margin-left:50px; color:green; background-color:white; border:1px dotted green; text-align:center;" cellpadding="5" cellspacing="0"
|- style="background-color: #cccccc;"
! Switch !! VBR target (kbit/s) !! VBR range (kbit/s) !! [[Channel coupling|Channel Coupling]] !! [[Noise normalization|Noise Normalization]] !! Note
|-
| -q -2 || ~32 || ~32 - ~64 || yes || yes || (1)
|- style="background-color: #cccccc;"
| -q -1 || ~48 || ~48 - ~64 || yes || yes || (2)
|-
| -q 0 || ~64 || ~64 - ~80 || yes || yes ||
|- style="background-color: #cccccc;"
| -q 1 || ~80 || ~80 - ~96 || yes || yes || (3)
|-
| -q 2 || ~96 || ~96 - ~112 || yes || yes ||
|- style="background-color: #cccccc;"
| -q 3 || ~112 || ~112 - ~128 || yes || yes ||
|-
| -q 4 || ~128 || ~128 - ~160 || yes || no ||
|- style="background-color: #cccccc;"
| -q 5 || ~160 || ~160 - ~192 || yes || no || (4)
|-
| -q 6 || ~192 || ~192 - ~224 || no || no ||
|- style="background-color: #cccccc;"
| -q 7 || ~224 || ~224 - ~256 || no || no ||
|-
| -q 8 || ~256 || ~256 - ~320 || no || no ||
|- style="background-color: #cccccc;"
| -q 9 || ~320 || ~320 - ~500 || no || no ||
|-
| -q 10 || ~500 || ~500 - ~1000 || no || no ||
|}

'' '''(1)''' Only supported on aoTuVb4 and newer '' 
'' '''(2)''' Bitrate of 48 kbit/s is only for aoTuVb4 and newer. Earlier versions and Xiph.org versions use a bitrate of 45 kbit/s '' 
'' '''(3)''' Reports seem to indicate that aoTuVb4.51 -q 1.5 provides good quality for streaming '' 
'' '''(4)''' Most people agree -q 5 achieves transparency''

=Advanced Encoder Settings=

===Reducing Pre-echo===
''This is applicable with Vorbis 1.1; it may not be applicable to aoTuVb4 or aoTuVb4.51.''

Generally, using the recommended settings above will give the best quality. However, there may be cases where Vorbis 1.1 will fail to reproduce sharp attacks or transients in your music, causing [[pre echo]]. In which case, you can use the '''impulse_noisetune''' advanced encode switch:

'''oggenc -q '''n''' --advanced-encode-option impulse_noisetune=p inputfile.wav'''
:'''where '''p''' is a number from 0 to -15'''

Note that the lower the number (toward -15) for impulse_noisetune, the higher the bitrate will fluctuate in passages of music filled with transient attacks (and the final average bitrate may be much higher than the nominal). Therefore, you should try a small value to start off (say -5) and see if you get acceptable quality. If not, tweak it lower.

===Reducing noise due to microattacks===
''This is only valid for some Vorbis encoders that are marked as having '''''impulse_trigger_profile'''.

There are certain parts of some types of music, called "microattacks", where Vorbis will produce a noise (sort of like a puff of steam), which is due to inaccuracies in the block-switching algorithm. Because the attacks are so fine and close together, Vorbis doesn't switch to short blocks enough which "smears" the reproduction.

By default (with no additional switches), Vorbis selects a different profile for block switching (lower means less switching, higher means higher switching) and the default values are shown in the table below:

{| border="0" style="margin-left: 150px; color:green; background-color:white; border:1px dotted green; text-align:center;" cellpadding="5" cellspacing="0"
|- style="background-color: #cccccc;"
! Quality !! Profile !!   !! Quality !! Profile
|-
| -1 || 0 || || 5 || 2.5
|- style="background-color: #cccccc;"
| 0 || 1 || || 6 || 2.7
|-
| 1 || 1 || || 7 || 3
|- style="background-color: #cccccc;"
| 2 || 1.5 || || 8 || 3.7
|-
| 3 || 2 || || 9 || 4
|- style="background-color: #cccccc;"
| 4 || 2 || || 10 || 4
|}

If you encounter this problem on microattacks, you may try the impulse_trigger_profile advanced encode switch, which will change (increase) the profile to your desired value.

oggenc -q n --advanced-encode-option '''impulse_trigger_profile=r''' inputfile.wav
:''where ''r'' is a number from 0 to 4, fractions accepted.''

'''Notes:'''
* There is the possibility that relaxed block switching (higher profiles) may cause other quality problems and create suboptimal Vorbis files. So use sparingly and with caution. If in doubt, leave impulse_trigger_profile on default (that is, don't use it at all)
* Setting the profile too high will make Vorbis switch to short blocks more often, which will lead to higher bitrate fluctuations. So be conservative.
* This settings has no effect on reducing the level of pre-echo. It can be said to only reduce the ''likelihood'' of pre-echo, but the amount of pre-echo is tuned using the '''impulse_noisetune''' switch instead.
* You may try profiles 5 and 6 as substitutes for 3 and 4. Profile 5 and 6 came from 3 and 4 in GT3b2.

You can use both impulse_noisetune and impulse_trigger_profile at the same time, but you will need separate switches, e.g.

oggenc -q n --advanced-encode-option impulse_noisetune=p
--advanced-encode-option impulse_trigger_profile=r inputfile.wav

Lossless comparison

2005-12-24T04:51:26Z

Lys: /* WV CONS */ capitalization

Given the enormous amount of [[lossless]] audio compressor choices available, it is a very difficult task to choose the one most suited for each person's needs.

Several people only take into consideration 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 that 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.

==Comparison Table==



{| border="0" cellspacing="0"
| '''Features''' || {{Turquoise}} ALAC || {{Turquoise}} FLAC || {{Turquoise}} Monkey's || {{Turquoise}} OptimFROG || {{Turquoise}} Real Lossless || {{Turquoise}} Shorten
|-
| {{yellow}} Encoding speed || {{white}} average || {{light-green}} fast || {{light-green}} fast || {{orange}} slow || {{white}} average || {{green}} very fast
|-|
| {{yellow}} Decoding speed || {{green}} very fast || {{green}} very fast || {{light-green}} fast || {{orange}} slow || {{light-green}} fast || {{green}} very fast
|-|
| {{yellow}} Compression* || {{light-green}} 58,50% || {{light-green}} 58,70% || {{green}} 55,50% || {{green}} 54,70% || {{white}}   || {{orange}} 63,50%
|-|
| {{yellow}} Flexibility** || {{orange}} bad || {{green}} very good || {{green}} very good || {{green}} very good || {{orange}} bad || {{orange}} bad
|-
| {{white}}  
|-
| {{yellow}} Error handling || {{white}}   || {{green}} yes || {{orange}} no || {{green}} yes || {{white}}   || {{orange}} no
|-|
| {{yellow}} Seeking || {{green}} yes || {{green}} yes || {{green}} yes || {{green}} yes || {{green}} yes || {{green}} yes
|-|
| {{yellow}} Tagging || {{green}} QT tags || {{green}} FLAC tags || {{green}} ID3/APE || {{green}} ID3/APE || {{light-green}} Proprietary || {{orange}} no
|-|
| {{yellow}} Hardware support || {{green}} yes || {{green}} yes || {{orange}} no || {{orange}} no || {{orange}} no || {{orange}} no
|-|
| {{yellow}} Software support || {{orange}} bad || {{green}} very good || {{light-green}} good || {{white}} average || {{orange}} bad || {{green}} very good
|-|
| {{yellow}} Hybrid/lossy || {{orange}} no || {{orange}} no || {{orange}} no || {{green}} yes || {{orange}} no || {{orange}} no
|-|
| {{yellow}} ReplayGain || {{white}} sort of || {{green}} yes || {{orange}} no || {{green}} yes || {{orange}} no || {{orange}} no
|-|
| {{yellow}} RIFF chunks || {{white}}   || {{orange}} no || {{green}} yes || {{green}} yes || {{white}}   || {{green}} yes
|-|
| {{yellow}} Streaming || {{green}} yes || {{green}} yes || {{orange}} no || {{green}} yes || {{green}} yes || {{orange}} no
|-|
| {{yellow}} Pipe support || {{orange}} no || {{green}} yes || {{green}} yes || {{green}} yes || {{orange}} no || {{green}} yes
|-|
| {{yellow}} Open source || {{light-green}} Yes (decoding) || {{green}} yes || {{green}} yes || {{orange}} no || {{orange}} no || {{green}} yes
|-|
| {{yellow}} Multichannel || {{green}} yes || {{green}} yes || {{orange}} no || {{orange}} no || {{orange}} no || {{orange}} no
|-|
| {{yellow}} High resolution || {{green}} yes || {{green}} yes || {{green}} yes || {{green}} yes || {{orange}} no || {{orange}} no
|-|
| {{yellow}} OS support || {{light-green}} Win/Mac || {{green}} All || {{green}} All || {{light-green}} Win/Linux || {{green}} Win/Mac/Linux || {{green}} All
|-|
| {{gray}}   || {{gray}}   || {{gray}}   || {{gray}}   || {{gray}}   || {{gray}}   || {{gray}}  
|-|
| '''Features''' || {{Turquoise}} WavPack || {{Turquoise}} WMA || {{Turquoise}} LA || {{Turquoise}} TTA || {{Turquoise}} LPAC
|-|
| {{yellow}} Encoding speed || {{green}} very fast || {{white}} average || {{orange}} slow || {{green}} very fast || {{white}} average
|-|
| {{yellow}} Decoding speed || {{green}} very fast || {{white}} average || {{orange}} slow || {{green}} very fast || {{green}} very fast
|-|
| {{yellow}} Compression* || {{light-green}} 58% || {{green}} 56,30% || {{green}} 53,50% || {{light-green}} 57,10% || {{light-green}} 57,20%
|-|
| {{yellow}} Flexibility** || {{green}} very good || {{orange}} bad || {{white}} average || {{orange}} bad || {{orange}} bad
|-
| {{white}}  
|-
| {{yellow}} Error handling || {{green}} yes || {{green}} yes || {{orange}} no || {{green}} yes || {{orange}} no
|-|
| {{yellow}} Seeking || {{green}} yes || {{green}} yes || {{green}} yes || {{green}} yes || {{orange}} slow
|-|
| {{yellow}} Tagging || {{green}} ID3/APE || {{light-green}} Proprietary || {{light-green}} ID3v1 || {{light-green}} ID3 || {{orange}} no
|-|
| {{yellow}} Hardware support || {{green}} yes || {{orange}} no || {{orange}} no || {{green}} yes || {{orange}} no
|-|
| {{yellow}} Software support || {{light-green}} good || {{light-green}} good || {{orange}} bad || {{white}} average || {{white}} average
|-|
| {{yellow}} Hybrid/lossy || {{green}} yes || {{orange}} no || {{orange}} no || {{orange}} no || {{orange}} no
|-|
| {{yellow}} ReplayGain || {{green}} yes || {{orange}} no || {{orange}} no || {{green}} yes || {{orange}} no
|-|
| {{yellow}} RIFF chunks || {{green}} yes || {{orange}} no || {{green}} yes || {{orange}} no || {{green}} yes
|-|
| {{yellow}} Streaming || {{green}} yes || {{green}} yes || {{white}}   || {{orange}} no || {{orange}} no
|-|
| {{yellow}} Pipe support || {{green}} yes || {{green}} yes || {{green}} yes || {{orange}} no || {{white}}  
|-|
| {{yellow}} Open source || {{green}} yes || {{orange}} no || {{orange}} no || {{green}} yes || {{orange}} no
|-|
| {{yellow}} Multichannel || {{green}} yes || {{green}} yes || {{orange}} no || {{green}} yes || {{orange}} no
|-|
| {{yellow}} High resolution || {{green}} yes || {{green}} yes || {{orange}} no || {{green}} yes || {{green}} yes
|-|
| {{yellow}} OS support || {{green}} All || {{light-green}} Win/Mac || {{light-green}} Win/Linux || {{green}} All || {{light-green}} Win/Linux/Sol
|}

<nowiki>*</nowiki> Encoding speed, Decoding speed and Compression ratio are based on each encoder's default settings.

<nowiki>**</nowiki> Flexibility refers to the amount of encoding choices offered to the users (Fast/low compression, Slow/high compression and everything inbetween)

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

==ALAC - Apple Lossless Audio Codec== http://www.apple.com/itunes/import.html

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

===ALAC PROS===
* Very fast decoding
* [[Open source]] ([http://craz.net/programs/itunes/alac.html decoding only])
* Hardware support ([[Apple iPod|iPod]], AirPort Express)
* Streaming support
* Tagging support (QT tags)
* Excellent hardware-software-lossy format integration with [[iTunes]]/iPod.
* Supports [[multichannel]] audio and [[high resolution]]s

===ALAC CONS===
* Closed source (encoding)
* Limited software support
* Compression efficiency not on par with other lossless codecs
* No hybrid/lossy mode

===ALAC Other features===
* Fits in the [[MP4]] container
* Can be used with the AirPort Express gadget

==FLAC - Free Lossless Audio Codec== http://flac.sourceforge.net/

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

===FLAC PROS===
* [[Open source]]
* Very fast decoding
* Hardware support ([[Rio Karma|Karma]], Phatbox, etc.)
* Very good software support
* Error robustness
* Streaming support
* Supports [[multichannel]] audio and [[high resolution]]s
* Tagging support (FLAC tags)
* Pipe support
* [[ReplayGain]] compatible

===FLAC CONS===
* Relatively slow encoding
* No hybrid/lossy mode
* Doesn't support [[RIFF]] chunks

===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
* Fits the [[Ogg]] and [[Matroska]] containers

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

[[LA]] is a lossless codec developed by Michael Bevin.

===LA PROS===
* Very high compression
* Tagging support ([[ID3v1]])
* Supports [[RIFF]] chunks
* Pipe support

===LA CONS===
* Closed source
* Very slow encoding and decoding
* Doesn't support [[multichannel]] audio and [[high resolution]]s
* No hardware support
* No hybrid/lossy mode
* Bad software support
* Doesn't support [[ReplayGain]]

'''''It's important to mention that the LA foobar plugin is buggy and doesn't produce lossless streams!'''''

==LPAC== http://www.nue.tu-berlin.de/wer/liebchen/lpac.html

Lossless Predictive Audio Coder ([[LPAC]]) is a lossless codec developed by Tilman Liebchen. Development of it has been halted in favour of development of [[MPEG-4]] ALS.

===LPAC PROS===
* Reasonable compression ratios
* [[High resolution]] audio support
* Supports [[RIFF]] chunks

===LPAC CONS===
* Closed source
* No error robustness
* Slow seeking
* No tagging
* No [[multichannel]] support
* No hybrid/lossy mode
* No hardware support
* Doesn't support [[ReplayGain]]

==Monkey's Audio== http://www.monkeysaudio.com/

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

===APE PROS===
* [[Open source]]
* High efficiency
* Good software support
* Java version (multiplatform)
* Tagging support ([[ID3v1]], [[APE tags]])
* [[High resolution]] audio support
* Supports [[RIFF]] chunks (only in the GUI encoder)
* Pipe support (only in a [http://www.etree.org/shnutils/shntool/ special] version)

===APE CONS===
* No [[multichannel]] support
* No hybrid/lossy mode
* No hardware support
* No error robustness
* Doesn't support [[ReplayGain]]

===APE Other features===
* Supports APL image link files (similar to CUE sheets)

==OptimFROG== http://losslessaudiocompression.com/

[[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
* [[ReplayGain]] compatible

===OFR CONS===
* Closed source
* No [[multichannel]] audio support
* No hardware support
* Quite slow decoding

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

==RealAudio Lossless== http://www.realnetworks.com/products/codecs/realaudio.html

RealAudio lossless is the lossless codec developed by Real Networks for their multimedia portfolio

===RAL PROS===
* Very fast decoding
* Streaming support
* Tagging support (proprietary)

===RAL CONS===
* Closed source
* No [[multichannel]] and [[high resolution]] audio support
* Little software support (Real Player)
* No hardware support
* Compression efficiency not on par with other lossless codecs
* No hybrid/lossy mode
* No pipe support
* Doesn't support [[ReplayGain]]

==Shorten== http://www.etree.org/shnutils/shorten/

[[Shorten]] is a very old and featureless lossless codec developed by Tony Robinson at SoftSound.

===SHN PROS===
* [[Open source]]
* Fast decoding
* Very good software support
* Supports [[RIFF]] chunks
* Pipe support

===SHN CONS===
* Quite inefficient
* No [[multichannel]] or [[high resolution]] audio support
* No hybrid/lossy mode
* No error robustness
* Not streamable
* No hardware support
* No native tagging
* Doesn't support [[ReplayGain]]

==True Audio (TTA)== http://www.true-audio.com/

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

===TTA PROS===
* [[Open source]]
* Good efficiency
* Hardware support (obscure DVD player)
* Supports [[multichannel]] audio and [[high resolution]]s
* Tagging support ([[ID3]])
* [[ReplayGain]] compatible
* Error robustness

===TTA CONS===
* No streaming support
* No hybrid/lossy mode
* Doesn't support [[RIFF]] chunks
* No pipe support

===TTA Other features===
* Fits the [[Matroska]] container

==WavPack== http://www.wavpack.com/

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

===WV PROS===
* [[Open source]]
* Very fast decoding
* Good efficiency
* Error robustness
* Streaming support
* Hardware support ([http://www.rockbox.org/ RockBox])
* Supports [[multichannel]] audio and [[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
* [[ReplayGain]] compatible

===WV CONS===
* Poor gui encoder support
* Little player support

===WV Other features===
* Supports 32bit float streams
* Supports embedded CUE sheets
* Includes MD5 hashes for quick integrity checking
* Can encode in both symmetrical and assymmetrical modes.
* Fits the [[Matroska]] container

==Windows Media Audio Lossless== http://www.microsoft.com/windows/windowsmedia/9series/codecs/audio.aspx

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

===WMAL PROS===
* Streaming support
* Very good software support
* Supports [[multichannel]] audio and [[high resolution]]s.
* Tagging support (proprietary)
* Pipe support

===WMAL CONS===
* Closed source
* No hybrid/lossy mode
* No hardware support (but it's likely to appear sooner or later)
* Doesn't support [[RIFF]] chunks
* Doesn't support [[ReplayGain]]

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

==Oddball Formats==

There are several old lossless formats that aren't being featured in the article above. Reasons are: lack of widespread support, lack of features, bad efficiency and, most importantly, it seems noone is really interested in them.

Most of those would have disappeared by now, but they are being preserved for posterity at [[User:Rjamorim|rjamorim]]'s [http://www.rjamorim.com/rrw/ ReallyRareWares]

===Advanced Digital Audio (ADA)===

* http://www.rjamorim.com/rrw/ada.html

===Bonk===

* http://www.logarithmic.net/pfh/bonk

===Marian's a-Pac===

* http://www.marian.de/en/downloads#APAC
* http://www.rjamorim.com/rrw/apac.html

===AudioZip===

* http://www.rjamorim.com/rrw/audiozip.html

===Dakx WAV===

* http://www.dakx.com/
* http://www.rjamorim.com/rrw/daxwav.html

===Entis Lab MIO===

* http://www.entis.gr.jp/eri/frame.html
* http://www.rjamorim.com/rrw/mio.html

===LiteWave===

* http://www.clearjump.com/products/LiteWave.html
* http://www.rjamorim.com/rrw/litewave.html

===Pegasus SPS===

* http://www.krishnasoft.com/sps.htm
* http://www.rjamorim.com/rrw/pegasussps.html

===RKaudio===

* http://www.msoftware.co.nz/downloads_page.php
* http://rksoft.virtualave.net/rkau.html

===Split2000===

* http://www.rjamorim.com/rrw/split2000.html

===Sonarc===

* http://www.rjamorim.com/rrw/sonarc.html

===VocPack===

* http://www.rjamorim.com/rrw/vocpack.html

===WavArc===

* http://www.rjamorim.com/rrw/wavarc.html

===WaveZip/MUSICompress===

* http://members.aol.com/_ht_a/sndspace/
* http://www.rjamorim.com/rrw/wavezip.html

==Links==

Some links to lossless format compression comparisons:

* [http://web.inter.nl.net/users/hvdh/lossless/lossless.htm Hans Heijden's] (used as reference to build the table);
* [http://members.home.nl/w.speek/comparison.htm Speek's];
* [http://foobar2000.net/lossless/ Guruboolez'] - comparing only classical music;
* [http://uclc.info/lossless_audio_compression_test.htm Johan de Bock's] - aimed only at the maximum compression setting for each codec (based on a somewhat limited set of samples, however).

* Go to the [http://www.hydrogenaudio.org/forums/index.php?showtopic=33226 Hydrogenaudio thread] to discuss this article.

Adobe Audition

2005-12-14T20:41:51Z

Lys:

'''Adobe Audition''' is a Digital Audio Editor released by [http://www.adobe.com Adobe Systems]. Formerly known as Cool Edit Pro, the editor was purchased in [http://www.adobe.com/special/products/audition/syntrillium.html May, 2003.] It is a powerful tool for creating audio for any destination, from CD and DVD, to broadcast television and digital media.

==Features==
Adobe Audition includes many advanced audio editing features.
===Multi-track Editing===
Adobe Audition is a multi-track editing program with support for up to 128 tracks. You can work in multi-track view, or edit individual waveforms. Audition also has tools for adding audio effects, and creating muisc with loops. It can record up to 32 tracks simultaneously with real-time effects and EQ processing on each track.
===Video Soundtrack Editing===
Audition includes features that make it easy to edit video soundtracks. Included is a video viewer and associated timeline for frame accurate audio editing. File format support includes [[AVI]], native DV, [[MPEG]], and [[WMV]].
===Digital Signal Processing===
Audition features approximately built-in audio processors and effects. Some of these include volume, noise and EQ filters, delay effects, reverb effects, and time/pitch shifting. Audition also supports third-party effects via both the DirectX interface and the VST plug-in architecture.
===High Resolution Audio===
Audition offers audio recording and editing at a variety of quality levels. It supports 32-bit audio with a sample rate of up to 10MHz. Standard sample rates (such as 44.1KHz - CD quality, 48KHz, 92KHz) are also available.
===Other Features===
* CD Burning Support
* 5000 Included Music Loops
* Built-In 5.1 Surround Sound Encoder

==System Requirements==
* 400MHz or faster processor (2GHz or faster recommended)
* Microsoft® Windows® 2000 or Windows XP Professional or Home Edition
* 64MB of RAM (512MB or more recommended)
* 75MB of available hard-disk space (700MB recommended for installing optional audio clips)
* 800x600 color display (1024x768 display recommended)
* Stereo sound card (multitrack sound card recommended)
* CD-ROM drive
* CD-RW drive for audio CD creation
* Microsoft DirectX 9.0 software required for video import
* Speakers or headphones recommended
* Microphone (optional)

==External Links==
* [http://www.adobe.com/products/audition/main.html Official Website]
* [http://www.vuplayer.com/other.php FLAC Filter] Allows Audition to save and edit [[FLAC]] files

{{Stub}}

MP2

2005-12-11T17:06:08Z

Lys: working link

In audio context, MP2 stands for MPEG-1/2 Layer II (see also the [[MPEG-2]] FAQ of the [[MPEG]] Audio Subgroup), but this abbreviation is also used for MPEG-2 video files.

The tooLAME implementation of this standard still gets developed nowadays and is regarded as having a better sound quality than the original reference software:

http://sourceforge.net/project/showfiles.php?group_id=19022

[[Musepack]] can be considered as a derivative from this standard, as it is based on the same principles, but developed into a [[subband]] [[codec]] of its own with superior audio quality.

This format is currently used in broadcast like satellite, DVB and DAB; and also in legacy VCD and SVCD discs.
[[Image:Mp2.png|none]]