Some questions about various audio codecs

If I go to Control Panel -> System -> Audio and video controllers -> Audio codecs (or “Compression” in Audio options- VirtualDub) there are a bunch of codecs! For example, CCITT A-Law, u-Law,… what do they serve me ? (I wonder what programs actually use them). I have some questions about them:

  1. The Fraunhofer IIS MPEG-1 Layer 3 (MP3) codec can compress from 1 to 320 kbps. But MPEG-4 codec already exists! We see it in DivX, XviD, Microsoft ASF (which was the first to use it)… why don’t we see any .mp4 files instead of the “old” .mp3 ones then :confused: ? Does the MPEG-4 codec use a psychoacoustic/compression algorithm that has little in common with the MP3 one ? Or does it serve video only ?

  2. What is the latest version of the Fraunhofer IIS MP3 codec ? Are there better ones on the net ?

  3. What bitrate does the Dolby AC3 codec use ?

I was wondering the same. And more. What is a good “break down” of codecs, like what to use when? I am just trying to find the highest quality with smallest disk space. I’ve only tried mp3 and wma though. Mp3 has horrible distortion (imo) at anythign below 256kbs. And then WMA im just leary of because of encryption. I want to re-rip all my fav cds to a 40gb ipod. Whats the suggestion? or where can I find more info about each codec?

The place to be for this, is probably - if a codec matters, it’s probably got a dedicated forum there.

MP3 survives for the same reason as ZIP - it ain’t great, but everyone knows it.
Fraunhofer is better (though MP3 is a poor choice) at low bitrates, while LAME tends to be better at mid-high bitrate - alt-preset-standard (often shortened to APS in discussion) is a highly tested and tuned VBR preset, that should average 220k and be transparent (not possible to distiguish from original) on all but the most demanding of material and exeptional ears - alt-preset-extreme is the next step above that.

If I understand it correctly, the audio part of Mpeg4 is AAC, and the encoder in Nero seems to do quite well.

Others favour a move away from patented/proprietary codecs, and champion OGG Vorbis or MPC.

MP3Pro does better than plain MP3, and WindowsMedia (WMA) also wipes the floor with MP3 at low bitrates - a recent HA test was the “dialup streaming” test, at 32k, had LAME MP3 as the low anchor, and it certainly sank.
If I remenber the scale, 5 is perfect, 4 is defects found when trying, 3 is defects audible, 2, is defects disturbing, and 1 is - completely unlistenable - well something like that.

My ears must be pretty wrecked, as 32k MP3 sounds ok to me - mind you, that is 32k MONO, so a bit less demanding than stereo, even with JS.

My guess was right, Nero AAC did pretty well!
The 128k MP3 results, other than a drop on the odd sample, LAME pretty well sweeps the board
But in this 128k multiformat test, the honours go to Vorbis AoTuv - a low bitrate tuning of the VORBIS open source codec.

A few more
MPC wins another of them

MPC is the best codec so far but since there is very little software support and no hardware suppord it will die. The next best is mp3, but mp3 has been tunes to the max and most likely oggvorbis(sp?) will take over if the creators work on it.

So I am stuck with mp3 if I want it portable right now. And everyone is probably going to laugh at this but I am going to “ask” anyways…It does matter what program you use to rip/encode then? EVEN IF it uses the same codec as another??

First of all that isn’t a stupid question. It is a tru audiophile question. In my humble life I have discovered that most people obtian the best results if they rip with EAC and then encode with the audiophile standard of lame(3.90.3), with APS.

Rip/encode on the fly, needs good handling of overruns, unless ripping below the maximum encode speed - eg “Accurate stream” in EAC.
Normally, to use the presets, you have to be using LAME command line (which EAC can, but not many others).
Tweaked DLLS, at one time mapping q-value to presets, but now one preset locked seems to be the way they do them, allows any LAME DLL capable application to use a preset.

I’ll just point out one more of the “128k” tests…
Bladeenc MP3, low anchor - based on the old sources, it’s hopelessly out of date, and hopelessly bad.
LAME, predictably, was last of the serious contentders - at 128, FHG may be better.
Of the rest, MPC was top, Vorbis just on the edge of the accuracy band at the bottom, but BOTH cheated on the bitrate.
In truth, AAC should really be the winner.

While the rest of you are discussing other compressions, I thought I would answer the last question in the original post. Dolby Digital bitrates are as follows:

192kbps for 2 channel material (lower bitrates may be possible but this is the standard)
384 or 448kbps for 5.1 channel material

I believe that movies (in the theaters) have to use 384kpbs because that is what the older theater equipment supports. 448kbps was added somewhere around the invetion of the DVD. Newer theater equipment may support 448, and it may be used in theaters now.

MPEG-4 is not only audio, it’s video and the mp4 extension is a container just as AVI, OGM, Matroska etc. What you want is a muxer that creates mp4 streams (take a look at doom9’s forum for more information). Regarding audio quality and low bitrates I’d recommend AAC perferably encoded using Ahead’s library/encoder. As for MPC it’s not very well supported and there are better formats out there and it’s not all that great to begin with.