Hydrogenaudio Knowledgebase - User contributions [en]

Transcoding

2006-05-08T08:37:35Z

Hankwang: subjective

'''Transcoding''' means converting a file from one encoding method (i.e. file format) to another. Transcoding can be performed from [[lossless]] to lossless, from lossless to [[lossy]], and from lossy to lossy.

Some transcoding tools you can find in the [[Download page#Transcoders|download page]].

==Lossy-to-lossy transcoding==

Every time you encode with a lossy encoder, the quality will decrease. There's no way to gain quality back even if you transcode your 128kbps [[MP3]] into a 320kbps [[MP3]] (or any other high-quality compression).

Transcoding between lossy formats is therefore generally not recommended. The sound quality of the result will always be worse than the (lossy) source file. Reasons to use this approach anyway could be:

*Lowering the bitrate for use in portable players, for which a listener may not care so much about sound quality.
*Saving storage space. Raw CD-audio data is about 1411 kbps (605 MB per hour); lossless encoders achieve around 700 kbps (300 MB/h). Lossy encoders such as [[Vorbis]], [[MPC]], and [[AAC]] achieve transparency for most people at 150--170 kbps (69 MB/h). For [[MP3]] (with the [[LAME]] encoder), transparency is usually achieved around 192 kbps (82 MB/h). For a large music collection, this could make a significant difference in disk space as compared to lossless archiving.

Lossy transcoding has been discussed in a few forum threads:

*[http://www.hydrogenaudio.org/forums/index.php?showtopic=32440&hl= Transcoding to mp3] - Listening test by guruboolez, from 256 kbps lossy to 128 kbps mp3. According to this test, Ogg Vorbis and MPC performed best, followed by WV4 and AAC with a marginal difference. Transcoding from 256 kbps MP3 to 128 mp3 caused a very significant deterioration compared to 128 kbps mp3 directly from the original.
*[http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=1545&view=findpost&p=348892 192 kbps Ogg to 128 kbps Ogg] Remark that this is a bad idea, although apparently without thorough [[ABX]] testing.
*[http://www.hydrogenaudio.org/forums/lofiversion/index.php/t41845.html Transcoding from 192 kbps (preset standard) mp3]. Informal subjective findings (no ABX test) with classic rock music: Ogg 96 or 128 kbps are audibly different from the source mp3, but still acceptable. Mp3 and WMA at 128 kbps had an unacceptable deterioration. AAC 96 kbps was better than Mp3 and WMA, but still had objectionable artifacts.
*[http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=39315&view=findpost&p=347373 Mp3 to Ogg] Ogg -q6 was required to achieve transparency against the (high-quality) mp3 with difficult samples.

==Lossless-to-lossless transcoding==

Unlike the aforementioned lossy transcoding, quality '''will not''' decrease. Thus you may transcode from one lossless format to another as often as you like (e.g. to take advantage of better compression or better error-correction or better hardware support).

==Lossless-to-lossy transcoding==

Keeping lossless archives gives you the opportunity to re-encode music in the future to other lossy formats as encoder technology improves. For example, if currently lossy format X is transparent at 192 kbps, while in three years lossy format Y is transparent at 128 kbps, it is still not likely that transcoding from X@192 to Y@128 will give acceptable results, contrary to transcoding from lossless.

If you are transcoding to lossy encoding from a lossless source, it is strongly recommended to '''keep the lossless source files'''. Thus, if the lossy result is not satisfactory, you can reencode easily.

Note that some transcoding tools have the option to automatically delete source files. Make sure that this option is '''turned off'''.

Transcoding

2006-05-07T13:44:00Z

Hankwang: /* Lossy-to-lossy transcoding */ guruboolez test clarification

'''Transcoding''' means converting a file from one encoding method (i.e. file format) to another. Transcoding can be performed from [[lossless]] to lossless, from lossless to [[lossy]], and from lossy to lossy.

Some transcoding tools you can find in the [[Download page#Transcoders|download page]].

==Lossy-to-lossy transcoding==

Every time you encode with a lossy encoder, the quality will decrease. There's no way to gain quality back even if you transcode your 128kbps [[MP3]] into a 320kbps [[MP3]] (or any other high-quality compression).

Transcoding between lossy formats is therefore generally not recommended. The sound quality of the result will always be worse than the (lossy) source file. Reasons to use this approach anyway could be:

*Lowering the bitrate for use in portable players, for which a listener may not care so much about sound quality.
*Saving storage space. Raw CD-audio data is about 1411 kbps (605 MB per hour); lossless encoders achieve around 700 kbps (300 MB/h). Lossy encoders such as [[Vorbis]], [[MPC]], and [[AAC]] achieve transparency for most people at 150--170 kbps (69 MB/h). For [[MP3]] (with the [[LAME]] encoder), transparency is usually achieved around 192 kbps (82 MB/h). For a large music collection, this could make a significant difference in disk space as compared to lossless archiving.

Lossy transcoding has been discussed in a few forum threads:

*[http://www.hydrogenaudio.org/forums/index.php?showtopic=32440&hl= Transcoding to mp3] - Listening test by guruboolez, from 256 kbps lossy to 128 kbps mp3. According to this test, Ogg Vorbis and MPC performed best, followed by WV4 and AAC with a marginal difference. Transcoding from 256 kbps MP3 to 128 mp3 caused a very significant deterioration compared to 128 kbps mp3 directly from the original.
*[http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=1545&view=findpost&p=348892 192 kbps Ogg to 128 kbps Ogg] Remark that this is a bad idea, although apparently without thorough [[ABX]] testing.
*[http://www.hydrogenaudio.org/forums/lofiversion/index.php/t41845.html Transcoding from 192 kbps (preset standard) mp3]. Informal findings (no ABX test) with classic rock music: Ogg 96 or 128 kbps are audibly different from the source mp3, but still acceptable. Mp3 and WMA at 128 kbps had an unacceptable deterioration. AAC 96 kbps was better than Mp3 and WMA, but still had objectionable artifacts.
*[http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=39315&view=findpost&p=347373 Mp3 to Ogg] Ogg -q6 was required to achieve transparency against the (high-quality) mp3 with difficult samples.

==Lossless-to-lossless transcoding==

Unlike the aforementioned lossy transcoding, quality '''will not''' decrease. Thus you may transcode from one lossless format to another as often as you like (e.g. to take advantage of better compression or better error-correction or better hardware support).

==Lossless-to-lossy transcoding==

Keeping lossless archives gives you the opportunity to re-encode music in the future to other lossy formats as encoder technology improves. For example, if currently lossy format X is transparent at 192 kbps, while in three years lossy format Y is transparent at 128 kbps, it is still not likely that transcoding from X@192 to Y@128 will give acceptable results, contrary to transcoding from lossless.

If you are transcoding to lossy encoding from a lossless source, it is strongly recommended to '''keep the lossless source files'''. Thus, if the lossy result is not satisfactory, you can reencode easily.

Note that some transcoding tools have the option to automatically delete source files. Make sure that this option is '''turned off'''.

Transcoding

2006-05-07T13:41:19Z

Hankwang: Lame -> LAME

'''Transcoding''' means converting a file from one encoding method (i.e. file format) to another. Transcoding can be performed from [[lossless]] to lossless, from lossless to [[lossy]], and from lossy to lossy.

Some transcoding tools you can find in the [[Download page#Transcoders|download page]].

==Lossy-to-lossy transcoding==

Every time you encode with a lossy encoder, the quality will decrease. There's no way to gain quality back even if you transcode your 128kbps [[MP3]] into a 320kbps [[MP3]] (or any other high-quality compression).

Transcoding between lossy formats is therefore generally not recommended. The sound quality of the result will always be worse than the (lossy) source file. Reasons to use this approach anyway could be:

*Lowering the bitrate for use in portable players, for which a listener may not care so much about sound quality.
*Saving storage space. Raw CD-audio data is about 1411 kbps (605 MB per hour); lossless encoders achieve around 700 kbps (300 MB/h). Lossy encoders such as [[Vorbis]], [[MPC]], and [[AAC]] achieve transparency for most people at 150--170 kbps (69 MB/h). For [[MP3]] (with the [[LAME]] encoder), transparency is usually achieved around 192 kbps (82 MB/h). For a large music collection, this could make a significant difference in disk space as compared to lossless archiving.

Lossy transcoding has been discussed in a few forum threads:

*[http://www.hydrogenaudio.org/forums/index.php?showtopic=32440&hl= Transcoding to mp3] - Listening test by guruboolez, from 256 kbps lossy to 128 kbps mp3. According to this test, Ogg Vorbis and MPC performed best, followed by WV4 and AAC with a marginal difference. Transcoding from 256 kbps MP3 to 128 mp3 caused a very significant deterioration.
*[http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=1545&view=findpost&p=348892 192 kbps Ogg to 128 kbps Ogg] Remark that this is a bad idea, although apparently without thorough [[ABX]] testing.
*[http://www.hydrogenaudio.org/forums/lofiversion/index.php/t41845.html Transcoding from 192 kbps (preset standard) mp3]. Informal findings (no ABX test) with classic rock music: Ogg 96 or 128 kbps are audibly different from the source mp3, but still acceptable. Mp3 and WMA at 128 kbps had an unacceptable deterioration. AAC 96 kbps was better than Mp3 and WMA, but still had objectionable artifacts.
*[http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=39315&view=findpost&p=347373 Mp3 to Ogg] Ogg -q6 was required to achieve transparency against the (high-quality) mp3 with difficult samples.

==Lossless-to-lossless transcoding==

Unlike the aforementioned lossy transcoding, quality '''will not''' decrease. Thus you may transcode from one lossless format to another as often as you like (e.g. to take advantage of better compression or better error-correction or better hardware support).

==Lossless-to-lossy transcoding==

Keeping lossless archives gives you the opportunity to re-encode music in the future to other lossy formats as encoder technology improves. For example, if currently lossy format X is transparent at 192 kbps, while in three years lossy format Y is transparent at 128 kbps, it is still not likely that transcoding from X@192 to Y@128 will give acceptable results, contrary to transcoding from lossless.

If you are transcoding to lossy encoding from a lossless source, it is strongly recommended to '''keep the lossless source files'''. Thus, if the lossy result is not satisfactory, you can reencode easily.

Note that some transcoding tools have the option to automatically delete source files. Make sure that this option is '''turned off'''.

Transcoding

2006-05-07T11:41:24Z

Hankwang: Add references to forum, rm superfluous wiki links, etc.

'''Transcoding''' means converting a file from one encoding method (i.e. file format) to another. Transcoding can be performed from [[lossless]] to lossless, from lossless to [[lossy]], and from lossy to lossy.

Some transcoding tools you can find in the [[Download page#Transcoders|download page]].

==Lossy-to-lossy transcoding==

Every time you encode with a lossy encoder, the quality will decrease. There's no way to gain quality back even if you transcode your 128kbps [[MP3]] into a 320kbps [[MP3]] (or any other high-quality compression).

Transcoding between lossy formats is therefore generally not recommended. The sound quality of the result will always be worse than the (lossy) source file. Reasons to use this approach anyway could be:

*Lowering the bitrate for use in portable players, for which a listener may not care so much about sound quality.
*Saving storage space. Raw CD-audio data is about 1411 kbps (605 MB per hour); lossless encoders achieve around 700 kbps (300 MB/h). Lossy encoders such as [[Vorbis]], [[MPC]], and [[AAC]] achieve transparency for most people at 150--170 kbps (69 MB/h). For [[MP3]] (with the [[Lame]] encoder), transparency is usually achieved around 192 kbps (82 MB/h). For a large music collection, this could make a significant difference in disk space as compared to lossless archiving.

Lossy transcoding has been discussed in a few forum threads:

*[http://www.hydrogenaudio.org/forums/index.php?showtopic=32440&hl= Transcoding to mp3] - Listening test by guruboolez, from 256 kbps lossy to 128 kbps mp3. According to this test, Ogg Vorbis and MPC performed best, followed by WV4 and AAC with a marginal difference. Transcoding from 256 kbps MP3 to 128 mp3 caused a very significant deterioration.
*[http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=1545&view=findpost&p=348892 192 kbps Ogg to 128 kbps Ogg] Remark that this is a bad idea, although apparently without thorough [[ABX]] testing.
*[http://www.hydrogenaudio.org/forums/lofiversion/index.php/t41845.html Transcoding from 192 kbps (preset standard) mp3]. Informal findings (no ABX test) with classic rock music: Ogg 96 or 128 kbps are audibly different from the source mp3, but still acceptable. Mp3 and WMA at 128 kbps had an unacceptable deterioration. AAC 96 kbps was better than Mp3 and WMA, but still had objectionable artifacts.
*[http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=39315&view=findpost&p=347373 Mp3 to Ogg] Ogg -q6 was required to achieve transparency against the (high-quality) mp3 with difficult samples.

==Lossless-to-lossless transcoding==

Unlike the aforementioned lossy transcoding, quality '''will not''' decrease. Thus you may transcode from one lossless format to another as often as you like (e.g. to take advantage of better compression or better error-correction or better hardware support).

==Lossless-to-lossy transcoding==

Keeping lossless archives gives you the opportunity to re-encode music in the future to other lossy formats as encoder technology improves. For example, if currently lossy format X is transparent at 192 kbps, while in three years lossy format Y is transparent at 128 kbps, it is still not likely that transcoding from X@192 to Y@128 will give acceptable results, contrary to transcoding from lossless.

If you are transcoding to lossy encoding from a lossless source, it is strongly recommended to '''keep the lossless source files'''. Thus, if the lossy result is not satisfactory, you can reencode easily.

Note that some transcoding tools have the option to automatically delete source files. Make sure that this option is '''turned off'''.

Recommended Ogg Vorbis

2006-03-27T07:34:19Z

Hankwang: /* 3rd party source code */ - link to compiling aoTuV

[[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 contribute, discuss, and work to improve the project.

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 was released, Xiph.org released '''Vorbis 1.1.2'''. Unfortunatly it only provides bugfixes to Vorbis 1.1.1. Although this is the 'official' version, it is '''not recommended''' by HA. If you are not interested in the latest "bleeding edge" improvements, you are welcome to use the latest release from Xiph.org '''Vorbis 1.1.2''' library.

''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 oggenc2.8 aoTuV beta 4.51 (MSVC compile)]'''
* '''[http://www.rarewares.org/dancer/dancer.php?f=60 oggenc2.8 aoTuV beta 4.51 (ICL compile for P3/AMD Athlon)]'''
* '''[http://www.rarewares.org/dancer/dancer.php?f=61 oggenc2.8 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 v1.8.7 aoTuV beta 4.51 (MSVC compile)]'''
* '''[http://www.rarewares.org/dancer/dancer.php?f=51 OggDropXPd v1.8.7 aoTuV beta 4.51 (ICL compile for P3/AMD Athlon)]'''
* '''[http://www.rarewares.org/dancer/dancer.php?f=52 OggDropXPd v1.8.7 aoTuV beta 4.51 (ICL compile for P4/AMD Sempron, Athlon 64)]'''

''(Work In Progress: Guide will be completed soon. Check [[OggDropXPd|here]])''

==OS/X binaries==
* '''[http://www.rarewares.org/files/ogg/vorbis_tools_1.1.1_aoTuVb4.51.zip Vorbis Tools (oggenc, oggdec, etc.) version 1.1.1]''' using aoTuVb4.51 for MacOS X.
: These tools were compiled by krmathis for OS/X.

==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'''
: This static GCC 4 binary was compiled by QuantumKnot.

''Note: There is no Linux Oggenc2.x that has been compiled with Lancer Optimizations as of yet (third-parties with a access to a unix environment are encouraged to try and compile one themselves if they have access to the sources).''

==3rd party source code==

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

* '''[http://www.geocities.jp/aoyoume/aotuv/ aoTuV (Aoyumi's Tuned Vorbis)]''' -- contains modified source code written in C and downloads of binaries.

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

: '''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 above for more information)''

: '''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 works for streaming, even good enough. However, more peer review is needed before this version becomes the recommended encoder.

: See [[Compiling aoTuV]] for information on how to compile it for Linux.

==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)]'''
* '''[http://homepage3.nifty.com/blacksword/oggenc2_lancer20060131.zip oggenc2.8 aoTuV beta 4.51 (Lancer 20060131) (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.'' 

=Released Binaries=

How do I know which encoder was used to make this particular Ogg Vorbis file?

Using either the ''ogginfo'' program or file info in your player, you can tell from the vendor tag:

If you aren't interested in the latest compiles feel free to use Vorbis 1.1.2 libraries.

{| border="0" style="margin-left:2px; color:black; background-color:white; border:1px dotted black; text-align:left;" cellpadding="4" cellspacing="0"
|- style="background-color: #eeeeee;"
! Vendor Tag !! Encoder !! Note
|-
| Xiphophorus libVorbis I 20000508
| '''1.0 Beta 1 or Beta 2 '''
|
|- style="background-color: #eeeeee;"
| Xiphophorus libVorbis I 20001031
| '''1.0 Beta 3 
|
|-
| Xiphophorus libVorbis I 20010225
| '''1.0 Beta 4 
|
|- style="background-color: #eeeeee;"
| Xiphophorus libVorbis I 20010615
| '''1.0 RC1 
|
|-
| Xiphophorus libVorbis I 20010813
| '''1.0 RC2 
|
|- style="background-color: #eeeeee;"
| Xiphophorus libVorbis I 20010816 (gtune 1)
| '''1.0 RC2 GT1 
|
|-
| Xiphophorus libVorbis I 20011014 (GTune 2)
| '''1.0 RC2 GT2 
|
|- style="background-color: #eeeeee;"
| Xiphophorus libVorbis I 20011217
| '''1.0 RC3 
|
|-
| Xiphophorus libVorbis I 20011231
| '''1.0 RC3 
|
|- style="background-color: #eeeeee;"
| Xiph.Org libVorbis I 20020717
| '''1.0 
|
|-
| Xiph.Org/Sjeng.Org libVorbis I 20020717 (GTune 3, beta 1)
| '''GT3b1 
|
|- style="background-color: #eeeeee;"
| Xiph.Org libVorbis I 20030308
| '''Post 1.0 CVS 
|
|-
| Xiph.Org libVorbis I 20030909
| '''1.0.1 
|
|- style="background-color: #eeeeee;"
| Xiph.Org/Sjeng.Org libVorbis I 20030909 (GTune 3, beta 2) EXPERIMENTAL
| '''Experimental GT3b2 
|
|-
| Xiph.Org libVorbis I 20031230 (1.0.1)
| '''Post 1.0.1 CVS 
|
|- style="background-color: #eeeeee;"
| Xiph.Org/Sjeng.Org libVorbis I 20031230 (GTune 3, beta 2)
| '''GT3b2 
|
|-
| AO; aoTuV b2 [20040420] (based on Xiph.Org's 1.0.1)
| '''aoTuV Beta 2 
|
|- style="background-color: #eeeeee;"
| Xiph.Org libVorbis I 20040629
| '''Xiph.Org Vorbis 1.1 or Xiph.Org Vorbis 1.1 RC1 
|
|-
| Xiph.Org libVorbis I 20040920
| '''Xiph.Org Vorbis 1.1 with impulse_trigger_profile 
|
|- style="background-color: #eeeeee;"
| AO; aoTuV b3 [20041120] (based on Xiph.Org's libVorbis)
| ''' aoTuV Beta 3 
|
|-
| Xiph.Org libVorbis I 20050304
| ''' Xiph.Org Vorbis 1.1.1 or Xiph.Org Vorbis 1.1.2 
|
|- style="background-color: #eeeeee;"
| AO; aoTuV b4 [20050617] (based on Xiph.Org's libVorbis)
| '''aoTuV Beta 4 
|
|-
| BS; Lancer [20050709] (based on aoTuV b4 [20050617])
| '''Lancer based on aoTuV Beta 4
|
|- style="background-color: #eeeeee;"
| AO; aoTuV b4a [20051105] (based on Xiph.Org's libVorbis)
| '''aoTuV Beta 4.5 
|
|-
| AO; aoTuV b4b [20051117] (based on Xiph.Org's libVorbis)
| '''aoTuV Beta 4.51 
|
|- style="background-color: #eeeeee;"
| BS; Lancer [20051121] (based on aoTuV b4b [20051117])
| '''Lancer based on aoTuV Beta 4.51
|
|-
| AO; aoTuV pre-beta5 [20060321] (based on Xiph.Org's libVorbis)
| '''AoTuV Beta 5
|}

=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, fractions accepted (using comma or period, depending on where the tool is compiled)'' e.g. '''oggenc -q 5 inputfile.wav '''

'''Notes:'''
* Most standard oggenc binaries can input lossless FLAC, WavPack, etc files as well (depending upon oggenc version)
* ''oggenc2.7'' and later are the only versions of oggenc that conform to the '''WAVEFORMATEXTENSIBLE''' header for outputing and decoding multi-channel Vorbis files (3 to 6 channels) in compliance with the Vorbis I decoding specification. Encoding multi-channel content might still produce sub-optimal files if lossless channel coupling is not used, due to modifications that need to be performed in libvorbis for proper channel coupling. Encoding assumes source material is CD audio content or PCM equivalent. ''
* The current oggenc and libvorbis API don't support the capabilities of full "bitrate peeling" yet. Experimental peelers exist, but only for testings purposes. ''i.e (Vinjey Systems)''

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

<div style="margin-left:3px;">
(1)'' Only supported on aoTuVb3 and newer '' 
(2)'' Bitrate of 48 kbit/s is only for aoTuVb3 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 provides good quality for streaming '' 
(4)'' Most users agree -q 5 achieves transparency''
</div>

=Advanced Encoder Settings=

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

Generally, using the recommended settings above will give the best quality. 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'''
:e.g. '''oggenc -q '''5''' --advanced-encode-option impulse_noisetune=-5 inputfile.wav'''

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 ''(which can't be corrected)''. Due to the fact the attacks are so fine and close together, Vorbis doesn't switch to impulse short blocks enough, thus "smearing" the reproduction.

By default (with no additional switches), Vorbis selects a different profile for block switching (lower means less switching, higher means more 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: #eeeeee;"
! Quality !! Profile !!   !! Quality !! Profile
|-
| -1 || 0 || || 5 || 2.5
|- style="background-color: #eeeeee;"
| 0 || 1 || || 6 || 2.7
|-
| 1 || 1 || || 7 || 3
|- style="background-color: #eeeeee;"
| 2 || 1.5 || || 8 || 3.7
|-
| 3 || 2 || || 9 || 4
|- style="background-color: #eeeeee;"
| 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. Please use it 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 impulse 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|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

Compiling aoTuV

2006-03-26T22:39:44Z

Hankwang: New article

As of March 2006, the [[Recommended Ogg Vorbis|recommended]] encoder for Ogg Vorbis is aoTuV beta 4. Here are notes for compiling aoTuV under Linux in case you do not want to use the binaries from [http://www.rarewares.org/quantumknot/ rarewares.org], for example if you want to optimize for your CPU, want to link other programs, or if you don't use an Intel-compatible CPU.

# Get the patched ''libvorbis source code'' from the [http://www.geocities.jp/aoyoume/aotuv/ aoTuV website]
# unpack the tarball and change into to the directory aotuv-b4_20050617_111merged/
# If you want to overwrite your existing vorbis libraries:
:: <code>CFLAGS=-fno-strict-aliasing sh ./configure</code>
:: <code>make</code>
:: <code>make install</code>
: If you want to keep your existing libraries:
:: <code>CFLAGS=-fno-strict-aliasing sh ./configure --prefix=/usr/local</code>
:: <code>make</code>
:: <code>make install</code>
:: Call oggenc as
:: <code>LD_PRELOAD=`echo /usr/local/lib/libvorbis*.so` oggenc -q4 foo.wav</code>
The flag <code>-fno-strict-aliasing</code> is necessary if you compile with gcc 4, see the [http://www.hydrogenaudio.org/forums/index.php?showtopic=36504 thread on HydrogenAudio].

Channel coupling

2006-03-26T19:39:49Z

Hankwang: Major rewrite of the Vorbis part.

Channel coupling is a method used to reduce [[bitrate]] demand by coding the stereo channel information more efficiently. there are several channel coupling methods. In mp3 the general term is [[joint stereo]].

[[MP3]] can use 2 different channel coupling methods: mid/side-coding and [[intensity stereo]]. Mid/Side-coding calculates a "mid"-channel by addition of left and right channel (l+r)/2 and a "side"-channel (l-r)/2. With more mono-like signals one can use less [[bitrate]] to encode the side-channel, so that the overall bitrate will be less than encoding the left and right channel. Intensity stereo destroys phase information, so it's used only at low bitrates.

Correctly implemented MP3 mid/side stereo (like in [[LAME]]) does very little or no damage to the stereo-image and increases compression efficiency either by reducing size or increasing overall quality.

[[Ogg Vorbis]] treats stereo information with '''square polar mapping''' which is beneficial when the correlation between the left and right channels are strong (this can also be extended to multichannel coupling as well). In Vorbis, the spectrum of each channel is normalized against a floor function, which is a rough envelope of the actual spectrum. In the square polar mapping, the (stereo) phase is roughly defined as the difference between the normalized left and right amplitude of a given frequency component. If the original left and right channel are the same within a certain frequency band, apart from an overall scaling factor, then the normalized frequency spectrum is the same left and right and the stereo phase is zero over the whole frequency band. Note that in the context of polar mapping, the term 'phase' (here: 'stereo phase') has a very different meaning from the phase of a periodic wave. Unlike in the Fourier transform, the cosine transform used in Vorbis and other encoders only provides amplitudes and no phases of the latter type.

Once the stereo information is represented in polar mapping as a magnitude and stereo phase, Vorbis can use three coupling methods:

* '''Lossless coupling''' is equivalent to independent encoding of the two channels ('dual stereo' in MP3), but with the benefit of additional space-saving. It does polar mapping/channel interleaving using the residue vectors.

* In [[point stereo]], the stereo phase is discarded completely. All the stereo information comes from the difference in the spectral floors for the left and right channels.

* In '''phase stereo'''', the stereo phase is quantized, i.e. stored at a lower resolution. Especially above 4 kHz, the ear is not very sensitive to phase information. Phase stereo is '''not''' currently implemented in reference encoder due to complexity, but will be re-added again later on. Note that phase stereo should not be compared to intensity stereo in MP3 coding.

Ogg Vorbis uses lossless/point stereo coupling below ''-q 6''. Lossless channel coupling is used for high bitrates entirely (''-q 6 and up''). This can be adjusted via an advanced-encode switch, but is not done for simplicity's sake.

See also [http://www.xiph.org/vorbis/doc/stereo.html Ogg Vorbis stereo-specific channel coupling] at xiph.org.

Vorbis

2006-03-26T17:55:02Z

Hankwang: /* Technical details */

[[image:fish_logo.png|right|]]

=Introduction=

'''Ogg Vorbis''' ([http://www.vorbis.com www.vorbis.com]) 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 (8khz-48.0kHz, 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.

Informal listening test suggests Vorbis to be comparable to MPEG-4 [[AAC]] at most bitrates and [[MPC]] 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. Currently Aoyumi is working on AoTuV Beta 4 and future releases. The latest version is AoTuV Beta 4.51, released in December 2005.

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). Vorbis is also an official part of the [http://www.openal.org/extensions.html OpenAL] extension library.

Before encoding files using Ogg Vorbis, check out the [[Recommended Ogg Vorbis]] page to know what encoder to use and what settings are recommended.

==Pros==
* Free. The Ogg Vorbis specification is in the public domain. It is completely free for commercial or noncommercial use.
* Good all-round performance (>48 kbps - a leading codec at [http://www.rjamorim.com/test/multiformat128/results.html 128 kbps])
* Well written [http://www.xiph.org/ogg/vorbis/docs.html specs]
* Several portable [http://wiki.hydrogenaudio.org/index.php?title=Ogg_Vorbis#Hardware hardware players]
* 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 developement is always encouraged)
* Current implementations are more computationally intensive to encode/decode than MP3 (Vorbis 2.0 seeks to overcome this limitation by slimlining the encoder)
* Multichannel input mappings for 5.1, Ambisonic-B, and other config have no channel coupling and aren't tuned (expect sub-optimal results until code is improved).

=Technical details=
* 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. (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.
* The noise-floor (envelope) is calculated using the global masking curve & piecewise linear approximation divided by spectrum to generate the residue (fine detail).
* [[Noise normalization]] is applied to compensate for energy lost in certain frequency bands due to quantization ''(main cause metallic warbling in MP3)''
* 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)
* [http://www.saunalahti.fi/cse/foobar2000/ foo_vorbisenc] vorbis encoder library for Foobar2000 (Win32)
* [[Lancer]] SSE-optimized vorbis encoder utility and libraries by BlackSword (Win32)

'''Decoders'''
* [[Ogg123]] for Unix systems (GPL), a very simple to use command-line player. (Win32/Posix)
* [http://www.rarewares.org/ogg.html OggDec] for Windows, by John33, a very featureful command line decoder. (Win32)
* [http://qtcomponents.sourceforge.net/ QuickTime Component] (Allows playback in [[QuickTime]]/[[iTunes]]) (Win32)
* [http://corevorbis.corecodec.org/ CoreVorbis] (DirectShow) (Win32)
* [http://www.illiminable.com/ogg/ illiminable Ogg Directshow Filters] (Also plays Speex, Theora and FLAC) (Win32)

=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 still evolving HA thread]
* [http://www.ifreemax.com/ FreeMax] FW-960
* [http://www.iaudiophile.net/ iAudio] [[IAudio M3|M3]], U2, G3, X5, I5
* [[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.yepp.co.kr/ Yepp] YP-T6, YP-T7, YP-C1, YP-F1, YP-53 (Firmware 1.200), other..

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

* [[Ogg]] (Container)
* [http://www.vorbis.com Vorbis official website] (updated)
* [http://en.wikipedia.org/wiki/Vorbis Vorbis at Wikipedia.]
* [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.rjamorim.com/test/ Listening tests comparing Vorbis against MP3, AAC, WMA, etc.]
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=15049 Recommended settings thread for encoding with Vorbis.]
* [http://wiki.hydrogenaudio.org/index.php?title=EAC_and_Ogg_Vorbis Configuring EAC and Vorbis as an external command-line encoder]
* [http://www.geocities.jp/aoyoume/aotuv/ Aoyumi homepage tuned versions of Vorbis encoder and current beta binaries.]
* [http://homepage3.nifty.com/blacksword/index_e.htm Archer/Lancer homepage optimized versions of AoTuV Vorbis encoder and other SSE optmizations. ]
* [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]
* [http://www.free-comp-shop.com/vorbis.pdf Keith Wright rendition of understanding the MDCT in Vorbis by defining it's basic trig properties (good read) (PDF)]
* [http://www.mp3-tech.org/programmer/docs/embedded_vorbis_thesis.pdf Ogg Vorbis decoder for an embedded system. Master Thesis (PDF)]
* [http://www.xiph.org Xiph.org Foundation]
* [http://wiki.xiph.org/index.php/Bounties Xiph.org Vorbis bounties]