[FAQ] Audio Formats 101 Final

[DreamTactix291]

MP3

MP3's full name is MPEG layer III audio. It is an older format dating back to the early 90s, but nowadays is by far the most prevalant audio format out there. It is not a bad format by any means but has a few limitations that newer formats have corrected. Most common bitrate used is probably 128kbps, but 192 has caught on a lot more than it used to. LAME 3.97b1's -V2 --vbr-new, a VBR preset (formerly --preset fast standard), is considered the best size vs quality tradeoff, by even the golden ears (audiophiles with way too good hearing for their own good) and is the recommended version at Hydrgenaudio.org. Preset standard is also what I recommend to people wanting to make their own mp3s.

Good mp3 encoders: LAME (especially from 3.9 on with the presets), FhG (more optimised than LAME for non 44.1kHz sampling rates)
Decent mp3 encoders: Xing (side note: althought Xing got a bad reputation about being a bad encoder it really isn't all that bad. Also the first encoder able to produce VBR files)
Bad mp3 encoders: Dist10, Blade, Plugger, Shine, and anything else based closely off the Dist10 sources (Dist10 is the sample encoder provided when you purchase the mp3 specifications)

The layer III specifications supports different bitrates and sampling rates for each different MPEG standard.

MPEG-1 (most common, high quality) - Sampling rates: 32kHz, 44.1kHz (44.1kHz is CD's sampling rate), 48kHz. Bitrates: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, and 320
MPEG-2 (made for smaller size) - Sampling rates: 16kHz, 22.05kHz, 24kHz. Bitrates: 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160
MPEG-2.5 (made for smallest size, non-official although widely supported) - Sampling rates: 8kHz, 11.025kHz, 12kHz. Bitrates: 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160

Bitrates can be per frame and VBR is achieved in mp3 by changing the frame size bits on a per frame basis. If you look at the stream it would be like this as an example - 32, 128, 160, 224, 320, 192, 256... You get the idea.

VBR = variable bitrate, constant quality, unpredicatble filesize
CBR = constant bitrate, variable quality, predictable filesize
ABR = average bitrate, constant to slightly variable quality, mostly predictable filesize. I like to think of ABR as "smart CBR".

MP3 tags are techincally a hack since the mp3's stream format does not specify tags. Common tags include

ID3v1: Fixed tag field sizes and fixed tags. The first tag format made for mp3. Also can be used for some other formats as well. Always takes up 128bytes exactly IIRC. Unofficial standard and widely supported. ID3v1.0 = no track number and ID3v1.1 = track number present.
ID3v2: Non-fixed tags field sizes and non-fixed tags. Most likely the second tag format made for mp3. Pretty much just mp3 uses this one. Very hard for programmers to work with due to the way this tag format stores its data from my understanding. Unofficial standard and widely supported. ID3v2.3 is the most common and best supported. ID3v2.4 has some problems with support or did the last time I checked. Replaygain values can be stored in ID3v2.
APEv1&2: Non-fixed tags field sizes and non-fixed tags. Like ID3v2 but much easier for programmers to work with. Actually structurally it looks a lot more similar to Vorbis comments IMO. Not nearly as widely supported as ID3v1&2. Replaygain values can be stored in APEv2.

General rule is if the player comes across tags it doesn't understand it just skips them. So don't worry too much about corrupting a file with too many tags. I now use ID3v1.1 and ID3v2.3 for my mp3 files since APEv2 borked tag support on my H120 for some files. Just something to be aware of.

MP3 can encode stereo information in several different ways

Mono: Not encoding stereo information at all. Only one channel is encoded.
Dual-channel/Dual-mono: Each channel is encoded independantly with no coupling at all. Not recommended for the reason that a 128kbps file with always have both channels with 64kbps a piece even though one channel might silent and the other is not. 128kbps full stereo is even smarter than that.
Stereo: Each channel is encoded independantly but can share bits due to being coupled. So a 128kbps file might have 80/48 or 64/64; whatever is more efficient.
Mid/Side Intensity Stereo: Used for low bitrates. Channels can be encoded mid/side or left/right. Throws away stereo information to save space though. Recommended only for low bitrates. Shows up simply as joint stereo.
Mid/Side Stereo: Recommended for all bitrates except for the really really low. Channels can be encoded mid/side or left/right depending on which is most efficient on a per frame basis. The smartest stereo mode mp3 has and yields the highest quality. Also shows up as joint stereo, but M/S IS is rarely ever used over 96kbps I believe. Regardless of what has been spread over the years a good MS implementation is far more efficient than plain, simple L/R stereo.

Starting with LAME 3.95 LAME now incorporates a VBR table to supercede the presets. This is similar to the quality settings of Ogg Vorbis and Musepack; just in reverse.

LAME VBR table for 3.95 through 3.96.1

Switch          Equals               Target  -Y  -b  Lowpass Resample
-V 0            --preset extreme       240   No  128 19500
-V 0 --vbr-new  --preset fast extreme  240   No  128 19500
-V 1                                   210   No  128 19000
-V 1 --vbr-new                         210   No  128 19000
-V 2            --preset standard      190   No  128 19000
-V 2 --vbr-new  --preset fast standard 190   No  128 19000
-V 3                                   175   Yes     18000
-V 3 --vbr-new                         175   Yes     18000
-V 4            --preset medium        165   Yes     18000
-V 4 --vbr-new  --preset fast medium   165   Yes     18000
-V 5                                   130   Yes     17000
-V 5 --vbr-new                         130   Yes     17000
-V 6                                   115   Yes     16000
-V 6 --vbr-new                         115   Yes     16000
-V 7                                   100   Yes     14900   32000
-V 7 --vbr-new                         100   Yes     14900   32000
-V 8                                    85   Yes     12500   32000
-V 8 --vbr-new                          85   Yes     12500   32000
-V 9                                    65   Yes     10000   24000
-V 9 --vbr-new                          65   Yes     10000   24000

Current VBR table for 3.97 branch

Switch         Equals                 Target  -Y  Lowpass Resample
-V 0           --preset extreme       210-270 No  19500
-V 0 --vbr-new --preset fast extreme  210-270 No  19500
-V 1                                  200-250 No  19000
-V 1 --vbr-new                        200-250 No  19000
-V 2           --preset standard      170-230 No  18600
-V 2 --vbr-new --preset fast standard 170-230 No  18600
-V 3                                  140-210 Yes 18000
-V 3 --vbr-new                        140-210 Yes 18000
-V 4           --preset medium        130-190 Yes 17500
-V 4 --vbr-new --preset fast medium   130-190 Yes 17500
-V 5                                  110-170 Yes 16000
-V 5 --vbr-new                        110-170 Yes 16000
-V 6                                  100-150 Yes 15600
-V 6 --vbr-new                        100-150 Yes 15600
-V 7                                   90-130 Yes 14900   32000
-V 7 --vbr-new                         90-130 Yes 14900   32000
-V 8                                   70-110 Yes 12500   32000
-V 8 --vbr-new                         70-110 Yes 12500   32000
-V 9                                   50-80  Yes 10000   24000
-V 9 --vbr-new                         50-80  Yes 10000   24000

[DreamTactix291]

Ogg Vorbis

Ogg Vorbis is an open source lossy audio codec created by the non-profit organization Xiph.Org (formerly Xiphophorus). While it is still used far less than mp3 by the masses it is gaining ground steadily and has hardware support on several DAPs including iRiver ones. Vorbis is inherently VBR, but it can have its bitrate managed to make an ABR mode. 1.1 is the official version currently out and it is mostly aoTuV b2 (a third party tweaked encoder merged with bugfixes). Most recent third party version is aoTuV b3, which is 1.1 with improvements. Ever since aoTuV b2 Vorbis's quality has been tied at the top.

Good Vorbis encoders: Most of them really. Everything except what I list below.
Decent Vorbis encoders: 1.0, 1.0 Post CVS, 1.0.1, 1.0.1 Post CVS (these are all slightly outdated)
Bad Vorbis encoders: The old betas and RCs probably, but then again Vorbis wasn't even 1.0 yet so they weren't intended to be great yet. Once 1.0 came out these were all obsolete.
Newest Vorbis encoder: aoTuV b4.51 (technically Lancer but it's based off of aoTuV b4.51)
Recommended Vorbis encoder: aoTuV b4.51, but I see no harm in using a few others as well. aoTuV b2, aoTuV b3, 1.1.0, GTune3 b1, GTune3 b2, Megamix, and Megamix II, 1.1.1, aoTuV b4, aoTuV b4.5, and 1.1.2,. Megamix and Megamix II bloat the bitrate a lot though, but they throw every quality tweak out without much concern for bitrate. I currently use aoTuV b4.51.

Vorbis isn't limited like mp3 is and supports just about any sample rate (though some are more optimised) and frame bitrate sizes can be anything from 0 to inifinity in theory. Vorbis also has tags built into the specification, so they are not hacks. Actually Vorbis Comments as they are called are probably some of the most powerful tags there are. Either them or APEv2.

Technically, Ogg is the container format and Vorbis is the lossy audio codec. Other things can and are muxed into Ogg: FLAC, a lossless audio codec; Speex, a voice codec; and Theora, an alpha-stage video codec. Others exist too but I'm sure you get the idea.

Vorbis is encoded usually by selecting a quality setting AKA a -q setting. -q 5 or -q 6 is usually the point where transparency happens in my experience. All of the -q settings are VBR, but they correspond with a nominal bitrate that encoder aims at. Sometimes it doesn't get too close but about 90% of the time it does.

-q -2 = 32 (aoTuV b2, b3, and b4 only)
-q -1 = 45
-q 0 = 64
-q 1 = 80
-q 2 = 96
-q 3 = 112
-q 4 = 128
-q 5 = 160 / 180
-q 6 = 192 / 212
-q 7 = 224 / 244
-q 8 = 256 / 276
-q 9 = 320 / 340
-q 10 = 499.9 / 519.9

The second numbers apply to all versions of GTune3 and Megamix I only. Decimal qualities are entirely possible as well. A setting of -q 4.78, for example, is allowed and will probably net a nominal bitrate of 150 or so. -q 10 is entirely overkill and probably shouldn't ever be used. If you have enough space for 500kbps then you'd probably be better with lossless. IMO over -q 7 is overkill but if you want to use more go right ahead.

Under -q 6 Vorbis can use lossy channel coupling to compress the file better, but over -q 6 channel coupling is lossless only. In my experience -q 4 and -q 5 don't tend to have problems with this making the soundstage sound narrow but below -q 4 I've heard it before.

The easiest to use Vorbis encoder is oggdropXPd at rarewares.org. You simply slide the slider to the quality setting you want and drag & drop the source wav files into the encoder and it outputs Ogg Vorbis. Commandline versions exist as well but they're not as friendly to newbies.

You can see what encoder made a file by looking at the vendor tag stored in every Vorbis file. The ones with 8 digit numbers in them are actually build dates in the order of Year/Month/Date. 20020717 is July 17, 2002.

Xiphophorus libVorbis I 20000508 = 1.0 beta 1 or beta 2
Xiphophorus libVorbis I 20001031 = 1.0 beta 3
Xiphophorus libVorbis I 20010225 = 1.0 beta 4
Xiphophorus libVorbis I 20010615 = 1.0 rc1
Xiphophorus libVorbis I 20010813 = 1.0 rc2
Xiphophorus libVorbis I 20010816 (gtune 1) = 1.0 RC2 GT1
Xiphophorus libVorbis I 20011014 (GTune 2) =1.0 RC2 GT2
Xiphophorus libVorbis I 20011217 = 1.0 rc3
Xiphophorus libVorbis I 20011231 = 1.0 rc3
Xiphophorus libVorbis I 20020307 = Post 1.0 RC3 CVS
Xiphophorus libVorbis I 20020406 = Post 1.0 RC3 CVS
Xiph.Org libVorbis I 20020717 = 1.0
Xiph.Org/Sjeng.Org libVorbis I 20020717 (GTune 3, beta 1) = GT3b1
Xiph.Org libVorbis I 20030308 = Post 1.0 CVS
Xiph.Org libVorbis I 20030909 (1.0.1) = 1.0.1
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
Xiph.Org/Sjeng.Org libVorbis I 20031230 (GTune 3, beta 2) = GT3b2
AO; aoTuV a1 (based on Xiph.Org's 1.0.1) = aoTuV alpha 1
AO; aoTuV a2 (based on Xiph.Org's 1.0.1) = aoTuV alpha 2
AO; aoTuV a3 (based on Xiph.Org's 1.0.1) = aoTuV alpha 3
AO; aoTuV b1 (based on Xiph.Org's 1.0.1) = aoTuV beta 1
AO; aoTuV b1a (based on Xiph.Org's 1.0.1) = aoTuV beta 1.5
AO; aoTuV b2 [20040420] (based on Xiph.Org's 1.0.1) = aoTuV beta 2
Vorbis Megamix: aoTuV beta 2 + QKTune beta 3.2 + GTune 3 beta 2 = Megamix
Xiph.Org libVorbis I 20040629 = Xiph.Org Vorbis 1.1 and Xiph.Org Vorbis 1.1RC1
Vorbis Megamix II: Xiph.Org 1.1 RC1 [20040629] + QKTune beta 3.2 + GTune 3 beta 2 = Megamix II
Xiph.Org libVorbis I 20040920 = Xiph.Org Vorbis 1.1 with impulse_trigger_prof ile & POST 1.1.0 SVN
AO; aoTuV b3 [20041120] (based on Xiph.Org's libVorbis) = aoTuV beta 3
BS; Archer B09.1 [20041209] (based on aoTuV b3) = Archer Beta 09.1
BS; Archer B10 [20041213] (based on aoTuV b3) = Archer Beta 10
Xiph.Org libVorbis I 20050304 = 1.1.1 & 1.1.2
BS; Archer RC1 [20050312] (based on aoTuV b3) = Archer RC1
BS; Archer RC2 [20050318] (based on aoTuV b3) = Archer RC2
BS; Archer RC3 [20050319] (based on aoTuV b3) = Archer RC3
BS; Archer RC4 [20050320] (based on aoTuV b3) = Archer RC4
AO; aoTuV pb4 [20050324] (based on Xiph.Org's libVorbis) = aoTuV prebeta 4 [20050324]
AO; aoTuV pb4 [20050403] (based on Xiph.Org's libVorbis) = aoTuV prebeta 4 [20050403]
AO; aoTuV pb4 [20050412] (based on Xiph.Org's libVorbis) = aoTuV prebeta 4 [20050412]
AO; aoTuV b4 [20050617] (based on Xiph.Org's libVorbis) = aoTuV b4 (pre- and post-1.1.1 merge)
BS; Lancer [20050621] (based on aoTuV b4 [20050617]) = Lancer based on aoTuV b4 pre-1.1.1 merge
BS; Lancer [20050709] (based on aoTuV b4 [20050617] 111merged) = Lancer based on aoTuV b4 post-1.1.1 merge
AO; aoTuV b4a [20051105] (based on Xiph.Org's libVorbis) = aoTuV b4.5
AO; aoTuV b4b [20051117] (based on Xiph.Org's libVorbis) = aoTuV b4.51
BS; Lancer [20051118] (based on aoTuV b4a [20051105]) = Lancer based on aoTuV b4.5
BS; Lancer [20051121] (based on aoTuV b4b [20051117]) = Lancer based on aoTuV b4.51

[DreamTactix291]

AAC

AAC stands for Advanced Audio Coding. It is an MPEG standard that was designed to be the successor to layers I-III. AAC has some big businesses backing it and it is also gaining momentum. The biggest reason for this is that Apple supports it on their iPod and their iTunes store sells it's music in DRM'ed AAC only. AAC is both and MPEG-2 and MPEG-4 standard. There are many flavours of AAC which I'll get to a little farther down. You almost always see AAC muxed in an .mp4 container (Apple's .m4a is the same thing)

Good AAC encoders: Nero, Quicktime/iTunes (CBR and ABR only though)
Decent AAC encoders: Coding Technologies, FAAC (only open source one I know of)
Bad AAC encoders: Too many old bad AAC encoders to remember. A lot of the original ones were really simple and the quality left much to be desired. This has changed however.

The two major competitors for best AAC encoder are the Nero and Apple's Quicktime encoders. The Nero encoder supports LC-AAC and HE-AAC, while the Quicktime encoder supports only LC-AAC at this time. Apple has announced HE-AAC support will be added soon, but the question is how soon. Nero has several VBR presets while Apple's is CBR only to my knowledge.

Like Vorbis, AAC does not have fixed frame bitrates. I believe there is a lower and a higer limit but I'm not sure. I've never seen an AAC file fall below 3kbps during silence, so I'd guess that to be the lower limit.

AAC has many different variations created for different uses

Profiles(MPEG-2)/Object Types(MPEG-4)
LC / Low Complexity: The most common version of AAC to be used with mid to high bitrates.
Main: Slightly more complex than LC. Much more rare though.
SSR / Scalable Sampling Rate: Able to scale down for streaming. Of lower quality than LC though.
LTP / Long Term Prediction - only in MPEG-4 AAC: LC with predictors. More rare than LC.
LD / Low Delay - only in MPEG-4 AAC: Designed for use in telecommunications. Lower quality than LC.
HE / High-Efficiency - only in MPEG-4 AAC: Designed for low bitrate usage. LC AAC with Spectral Band Replication.
HE + PS / High-Efficiency + Parametric Stereo - only in MPEG-4 AAC: Same as HE but with parametric stereo. Designed for extremely low bitrates, i.e. 24kbps - 32kbps.

AAC muxed in an .mp4 (or .m4a) container uses tags that are taken from Apple's .mov container. Unmuxed AAC can be tagged like mp3 with ID3v1, ID3v2, or APEv2. MPEG-7 I believe defined a set of standard tags for use with AAC but I don't believe they are in use yet.

AAC isn't Apple's proprietary format like many people believe, but they have a lot of money in it so they back it a lot. My personal preference is the Nero encoder, but the others are worth mentioning.

Currently AAC encoders at best match LAME at the same settings. AAC has a long way to go before it's optimised like mp3 has been.

[DreamTactix291]

Musepack

Musepack, formerly known as MPEGPlus, is a lossy audio format that is strongly based on MPEG-1 layer II (mp2) but has evolved far past it. It was created by Andree Buschmann for one purpose: quality, and it has one of the finest tuned psymodels for lossy audio in the world. Musepack is a VBR only encoder like Vorbis and has absolutely no way of forcing an average bitrate on it. Musepack also has the fewest problem cases among all of the modern audio codecs. While Musepack is pretty much a dormant project right now (although mppenc 1.15s, 1.15t, 1.15u, and 1.15v have been released recently) there is hope that it will continue improving and hopefully fix its biggest flaws of being unable to be muxed with video and its slow seeking (it seeks about as fast as mp3 though). Musepack files get an .mpc extension but the old .mp+ extension is still valid. Musepack is now open source unlike in the past, but also as of now has no portable hardware support. It is on the Rockbox list of formats for the H100 series, and can be played back at realtime now.

Good Musepack encoders: Pretty much any of the SV7 encoders
Bad Musepack encoders: None that I know of

The current StreamVersion is 7, which is the container. SV8 is hoped for in the not too distant future. A tool that will allow you to remux current SV7 files without loss into an SV8 container has been promised once SV8 is complete. As of now Musepack only supports 1 or 2 channels so it is not for use with multichannel.

Supported input formats (SV7):

Channels: 1 or 2
Bit depths: 1 to 32 bit linear PCM
Sample rates: 32 kHz, 37.8 kHz, 44.1 kHz, 48 kHz (44.1 and 48 are highly tuned)

Musepack like Ogg Vorbis has a quality slider. Qualities 1 through 10 are valid. --quality 5 AKA --standard is the setting that was created for transparency. Going over --quality 6 AKA --xtreme is considered overkill. The --xlevel switch was once recommended to reduce clipping artifacts induced by encoding but as of 1.15s it is hardcoded into the encoder. It will be completely unneccessary once SV8 is done.

Profile Options (Quality Presets):

--telephone or --quality 1 lowest quality,       (typ.  32... 48 kbps)
--thumb     or --quality 2 low quality/internet, (typ.  58... 86 kbps)
--radio     or --quality 4 medium (MP3) quality, (typ. 112...152 kbps)
--standard  or --quality 5 high quality (dflt),  (typ. 142...184 kbps)
--xtreme    or --quality 6 extreme high quality, (typ. 168...212 kbps)
--insane    or --quality 7 extreme high quality, (typ. 232...268 kbps)
--braindead or --quality 8 extreme high quality, (typ. 232...278 kbps)

I personally recommend --standard or --xtreme. --insane and --braindead are completely overkill as well as --quality 9 and --quality 10. Decimal qualities are allowed like --quality 5.25 like Ogg Vorbis. Also Musepack is the fastest of the the modern lossy codecs on both the encode and decode side of things.

Musepack uses APEv2 tags by default but APEv1 and ID3v1 tags are also valid. ID3v2 is not recommended for use with Musepack.

Musepack also has a different kind of version numbering system.

1 decimal point (i.e. 1.0, 1.1): stable
2 decimal points even (i.e. 1.12, 1.14): beta
2 decimal points odd (i.e. 1.13, 1.15): alpha

[DreamTactix291]

Lossless Codecs

I'm going to link to a thread at Hydrogenaudio.org for this one as this has been discussed pretty well.

Lossless codecs unlike lossy codecs like mp3, Ogg Vorbis, AAC, Musepack, and Windows Media Audio Standard and Professional do not discard any information during compression. Think of them like a zip or rar file only designed for audio and can be played back. Personally I've only used 3 of them and played back 4 of them. So I'll leave the long Pros and Cons to the HA.org page. Instead I'll go over the ones I know about, which are incindentally the ones most used anyhow. Unlike lossy codecs you can transcode from lossless to lossless with no penalty. You can't screw up a lossless encode so they're easier for newbies

FLAC (Free Lossless Audio Codec): Part of the Ogg framework like Vorbis and can be used in its own native container, .flac, or in the ogg container, .ogg. FLAC doesn't compress as much as some of the other codecs, but this is intentional. It has one of the best decode times which allows it to have hardware support in players like the Rio Karma and iAudio M3 and X5 as well as the iRiver H1xx series with Rockbox firmware. I would consider its compression to be average to above average, but it has very good tagging in the form of FLAC tags (which are 100% the same as Vorbis comments). FLAC's current version is 1.1.2 and is one of my two lossless codecs of choice.

Monkey's Audio (APE): This is another open source lossless codec that instead of boasting fast decode times instead boasts some of the best compression for lossless audio. However this pretty much means it will never have hardware support and currently Monkey's Audio can't do multichannel. This isn't an issue for CDs but if someone ever cracks SACD and DVD-A this means Monkey's wouldn't be the codec of choice. I would consider its compression to be far above average and excellent. It uses either ID3v1 or APEv1 or v2 for tagging (APE tags came from this format). Monkey's Audio's current version is 3.99.

Shorten (SHN): This is an older open source lossless codec that is still very popular in trading circles. It has good compression times but has some of the worst compression of the lossless codecs. Seeking was also a hack into it. Shorten could have hardware support but IMO FLAC is a much better choice as it is far improved over Shorten. The trading circles are starting to have FLAC show up more. I would personally consider Shorten outdated and its compression to be below average and has no native tagging.

WavPack (WV): This is another open source lossless codec that has impressed me a lot of late. Almost to the point of switching. It has better compression than FLAC nowadays, and also features a lossless/lossy hybrid mode. While this isn't unheard of WavPack is the only open source codec to do this AFAIK. WavPack's comprssion is slightly better than FLAC's (though it is getting even better) and it's decode is very similar. WavPack could easily have hardware support added one because of this. I consider WavPack's compression to be above average though not as much as Monkey's Audio; still better than quite a few lossless codecs. It uses ID3v1 and APEv1 or v2 for tagging. WavPack's current version is 4.31 and it has really come into its own since 4.0. WavPack is one of my two lossless codecs of choice and has hardware support in the iRiver H1xx series with Rockbox

Which one should you use? I personally would recommend FLAC or WavPack. FLAC is good but WavPack just keeps getting even better. All 4 of these formats have Winamp plugins and all can be played with foobar2000 out of the box I believe. There are also several proprietary formats like WMA Lossless and Apple Lossless but I've chosen not to cover them as I don't know as much about them. That is why there is the link to HydrogenAudio and why I included the graph.

[DreamTactix291]

http://www.misticriver.net/boards/sh...9&page=1&pp=20 This is the link to the old Audio Formats 101 page. I will consider that one deprecated and this one is designed to supercede it. The whole reason I made this was I've noticed that not a lot of places anywhere have a really good simple enough explanation of these common codecs. Because of this I've decided to compile all of my knowledge of these things into one place and hopefully someone will benefit from it.

This concludes Audio Formats 101 barring some great advancement like Vorbis 2 or Musepack SV8.

[DreamTactix291]

Minor cosmetic changes and slight updates. Also uses this as a shameless bump