Audio Format Listening Tests Concluded
Pointing to the conclusions of this listening study, nullity writes: "The results are interesting, and show a high variation in the performance of the various codecs on different musical styles. Ogg seems to work well on dance music, WMA8 on chamber music, etc."
Ogg seems to work well on dance music, WMA8 on chamber music, etc.
Like requiem...
While mp3s encoded at lower bit rates seem to have a tingly sound sometimes, almost every wma file I've gotten or encoded ALWAYS sounds like someone is blowing windchimes in my music. I can't stand it.
In my opinion, 192kbps MP3 is the way to go, but then what do I know? I can't hear about 13khz.
Chris
These tests are all at 64 kbps and most people use much higher bitrates for real music. I'd like to see comparisons at 128k bits minimum, and preferably 160k or 192k, which is what most quality mp3's are at, for direct comparison.
Tests confirmed that attempting to encode "Aphex Twin" with any of these codecs caused the PC to tremble at a frequency that, when connected to a refracting laser stuck up Bill Gates' ass, had it spell out "we're all dead" on the nearest wall.
Invoicing, Time Tracking, Reporting
Well....not quite. There's a different frequency distribution between electronic, pop acoustic and classical music.
Specifically, electronic music, which most dance stuff is, has a very flat frequency distribution. See this for yourself - load your favourite media player, siwtch on the graphic equaliser graph and watch how basically nothing happens except in the mid-range.
Now try again with an orchestral piece. There will be much more variation, though in most it will tend towards the top end.
Now try again with rock. Tends towards the bottom and top, with middle frequencies missing.
Keep going with any format you feel like mentioning...you'll get the same.
Actually, this is a striking example of how recording techniques can ruin sound as well. Take a look at the Apollo 440 album - Gettin' High on Your Own Supply. A good mixture of guitars and electronics, right? Well, look at the frequency graph again. See how virtually every guitar frequency variation has been cut out: this music was recorded digitally, mostly using samples by the looks of it. The normal variations you'd associate with having guitars play live are all filtered out, and the graph goes back to the flat digital sound again.
Cheers,
Ian
This isn't terribly surprising, as it was already known that the different formats have different frequency responses. More specifically, the way they compress the music dictates what frequencies are cut out. MP3s, for example, are notorious for removing high and low frequencies from music - not a big deal for casual listeners, but those with high-end stereo systems will definitely notice the lack of high overtones, and the "flat" low end bass response. WMA sort of sounds like certain frequencies are cut from the raw audio, leaving the rest to fill in as sort of an approximation of the original full sound - it sounds hollow and "chime-y". Ogg has its defining sound characteristics as well. Thus, it isn't surprising that different styles of music sound better encoded in different formats, as different styles take advantage of different frequencies. Rock music has high frequency cymbals and low frequency bass drums and guitars, as well as a very full mid-range, so a well-rounded encoding system works well. Classical is somewhat more compressed, as a result of the physical limitations in terms of sound reproduction of the instruments, so to the undiscerning ear, a format with especially good mids will suffice. The examples go on and on, but the point is that different tools are needed for different jobs - if nothing else, this study shows that having having a number of encoding tools on hand is actually a good thing. When you look in your tool box, you've got more than a couple Phillips-head screwdrivers - you should have enough tools to deal adequately with any job. The same applies to music.
Let's assume that anyone who likes Ogg and is seriously into music will compress their music with both Ogg variants and use the best variant for each file.
Therefore we should also consider taking the best of the two results and comparing it to mp3.
From a quick look at the results it appears that Ogg will still be edged out by mp3 when analysed in such a fashion, but it's much closer.
Also a test on several bit rates would be useful.
See http://etbe.coker.com.au/ for my blog.
I guess grip will have to use Genre info from CDDB to decide what to encode the the files as now. I wonder if you coudl set up something to optimize individual tracks. Like scan a wav and pick the best codec for the frequencies used in the audio.
Why not fork?
I'd be interested to know how these codecs perform when streaming things like news or talk radio or foreign language lessons. Clarity at a low bandwidth would support a lot of simultaneous listeners from a low-end server. Clarity at medium bandwidth could provide the extra sound quality needed for something like language learning/practice, again from relatively modest servers.
"Those who have never entered upon scientific pursuits know not a tithe of the poetry by which they are surrounded."
I noticed a number of confused posters here... The tested codecs were AAC/MP3PRO/OGG/WMA, not MP3. Had mp3 been tested, it would have lost every round as all of the tested codecs are vastly superior to plain MP3 at this bitrate.
It also should be noted that the only two samples that WMA beat OGG at (indeed the only ones that it didn't totally flop on) were two very simple samples that are demonstrations of two differnt weaknesses in the current revision of vorbis. Orignally the results page had some very interesting commentary from Monty on this, but it looks like it got pulled.
With the exception of those two samples, OGG clearly won. Even including those, it was only beat out by MP3PRO by a small margin. When you factor in that MP3PRO isn't available at anything but such low bitrates and that it's substantially more propritary then MP3, it seems like pretty much a no-contest.
that these codecs are lossy, and take advantage of the fact that the human ear is better at hearing certain things than others to pair out extraneous info and improve compression. IOW, it doesn't matter how technically different the new files are as long as they still sound the same to the human ear.
BlackGriffen
Considering that different codecs do better at different music w/ different frequency spreads, who else thinks that the next generation of audio codecs will be multi-modal; in effect, be several codecs in one. Then have each codec specialize on certain types of music. Perhaps even have them run in an advanced mode where they do a frequency analysis of whole songs, rather than just using genre, to automatically select the best codec for the job. Perhaps even use different codecs for different sections of the song. That would definitely help songs like Bohemian Rhapsody and orchestas with movements, etc.
Would this be too time consuming to implement or what?
BlackGriffen
Its not diferent formats. Just diferent encoder options.
The oggq0 entries are for music enconded in a Variable Bit Rate mode (oggenc -q0) -- the encoder defines a quality treshold and uses whatever bitrate necessary to keep it there.
The ogg64 entries are for music encoded with a nominal bitrate (oggenc -b 64 --manage) -- it atemts to keep the bitrate around 64 kbps without looking at sound quality.
Why did only Ogg Vorbis got to show these two modes? Because though the test focuses on 64kbps (nominal bit rate) encoding, its likely than most Ogg Vorbis users will use variable bit rate encoding with it. I know I do.
I'd like to know if there are and codecs that support constant quality but a variable bit rate?
A codec with a target bitrate of 64k but maintains quality by channing between say 1k(for silence) and 100k through the streem would be nice.
thank God the internet isn't a human right.
But is their some fundamental reason why nobody else insists on VBR?
Do you mind, your karma has just run over my dogma.
My particular problem is that I hate the noise PC CD-ROMS make while spinning at 1x. Since I like to listen to music while working, I make recordings that play without that noise in the background (my PC fan is very quiet). Plus, I can play from a large subset of my CD collection from either machine at home or from my machine at the office.
TWW
"Encyclopedia" is to "Wikipedia" what "Library" is to "Some people at a bus stop"
Just a side note about the frequency distribution of different styles of music:
The reason why classical music generally compresses better is because the frequency distribution of the sound of natural instruments like for instance string instruments (including the human voice) is harmonic. This means that the sound spectrum consists mainly of a superposition of peaks at the base frequencies of the instruments played and their corresponding harmonics at higher frequencies.
If you were to make a two dimensional spectral analysis of a such sound recording with the time axis to the right, the frequency to the top and the amplitude as the color intensity of the point you would see a lot of wiggling lines at
regular distances. (BTW: this would make a great visualization plugin for xmms)
Since audio compression algorithms also make such a spectral analysis and after that discard some of the information below a threshold they can
reproduce a mainly harmonic spectrum easier than that found in pop or rock music, which is much more complex and more "noisy" because of the
use of distorting amplification and all kinds of
percussion.
Holger
Looking at the data, it looks the two samples where Ogg performed poorly ended up being encoded at a significantly smaller average bitrate than any of the other encoders.
The table at the end lists LiszBMinor with an average ogg bitrate of 45 and BachS1007 with an average bitrate of 47. Since the other codecs encoded those samples at a bitrate 64 or higher, this may explain the results.
The results may point to a flaw in Ogg's VBR login rather than in the lossy compression scheme.
OVERALL RANKINGS (12 SAMPLES)
mp3pro 49.00
oggq0 44.00
ogg64 40.00
wm8 24.00
aac 23.00
The AC above me speaks the truth. mp3PRO has no hope of gaining enough market share to become a worthy competitor. It's a very proprietary extention to MP3. OGG being open source and free (as in beer) has clear advantages for hardware vendors (where it really counts). Lets hope the codec is easy to embed into portable products.
I want my Portable OGG CD Player! I'll buy the first one that comes out. Could you imagine? Twice the capacity of normal players and it STILL sounds better (or same capacity truly indistinguishable from CD -- at only 128k). Right now I have to encode my mp3's at ~180-220kbit to get something acceptable. =/
I haven't seen anyone else mention this yet. At the end, he gives a table of the bitrates for each song for each codec. The one with the greatest variation appears to be oggq0. I noticed that for the songs where that codec did well, the bitrate was much higher, and where it did poorly, it was much lower. I don't realy understand how the bitrate is chosen, but as I understand it, the encoder chooses it automatically somehow, right? I wonder how effective that really is.
The 60 files (12 songs * 5 formats) were all compressed at between 64 and 74 kbits/sec - except LiszBMinor and BachS1007 for OGG q0. They were actually stored at 45 and 49 kbits/sec respectively. No surprise the testers rated them low.
-
- - You can't take something off the Internet! That's like trying to take pee out of a swimming pool.
ABC/HR.. as in ABC/Hidden Reference... as in, there is a copy of the original track included as a hidden reference on every single trial.
The users are given 2 sliders per sample laid out on a panel. The samples are loaded in random order. On the sliders for each sample, one slider is for the original sample, and one is for the encoded. These are also randomized per sample. The user does not know which is which. If they happen to rate the original sample less than 5.0 (highest rating, meaning it should be transparent), then their results are disregarded entirely for that sample.
Man did you guys even do the test or are you just looking at the results?
Yes, all 12 samples, and my results broadly agree with the general conclusions (particularly on MP3pro doing much better than AAC and WMA on almost all the samples).
The only ones I had trouble with were the two oggs and on some files the mp3pro
In general, the results agree with you, so I don't understand why you're so annoyed by the test. When Vorbis was good, it was *very* good - and when it was bad, it was terrible. Luckily, Monty will use the results to help the 64kbits performance for the next version of Vorbis.
(OT: you *really* need to learn about paragraphs)
-- Help Digitise the Public Domain at DP.
Does anyone else feel it would have been nice to see Red Book CD audio (16-bit 44.1KHz uncompressed) compared as a control? Seeing how "pure" audio compares to these compression standards could make the results seem more objective.
mp3pro is better than wma8 (99.9% confidence)
mp3pro is better than aac (99.9% confidence)
oggq0 is better than aac (99.3% confidence)
oggq0 is better than wma8 (99% confidence)
ogg64 is better than aac (97.2% confidence)
ogg64 is better than wma8 (96.1% confidence)
Is 99% good enough for you? Or perhaps you should just take the two at 99.9%?
Dammit, the lameness filter is kicking in. No, these are *not* junk characters - I'm trying to show the peon some useful statistical information, you worthless piece of software. I've already removed all the hyphens, what the hell more do you want me to do? Is a percentage sign 'junk'? Is a question mark? Is a space? What the *fuck* use is this, when it doesn't stop all the crapflooders in any way whatsoever - they just flood with random gay/incest/beastiality sex stories instead... I've been posting on this site for years, and for my sins haven't crap-flooded once - give me a LITTLE FUCKING LICENCE TO POST MATERIAL.
-- Help Digitise the Public Domain at DP.
Been using Ogg/Vorbis/Squish on Quicktime for a year. The Ogg/Vorbis/Squish codec got much better between 1.0rc2 and 1.0. At 128k it's already better than mp3 and the managed bitrate encoding is faster than the hard drive can read. The real value is of course, the ability to read these encoded files as long as there is UNIX. Mp3 is going to die and when it does there won't be any appliance makers interested in paying the $10,000 royalty to support mp3.
Please moderate the parent up - unlike the grandparent, this anonymous coward actually knows what he is talking about.
-- Help Digitise the Public Domain at DP.
Convenience, convenience, convenience. I've got an MP3 CD in my car that's been in there for nearly a month. It's got all my current favorite tracks/albums on it. When I get tired of it, I'll burn another CD that may last as long. Also, lossy compression is very acceptable for mediums such as speeches, books on ta--MP3, etc. I've got a road trip coming up, and I'm planning on listening to the Lord of the Rings Trilogy that I borrowed from a friend. At 64kbps, I can fit something like 21 hours of audio on one CD. As opposed to the 18 or so CDs that it would take if it were pure CD-Audio. You don't have to get rid of CDs, but MP3s have their place, and they're not going away. Lossy compression or not, you can make an MP3 sound as good as CD Audio. You can. Just turn up the bitrate, play with the settings. Use lossless compression if you want, but I guarantee you can rip an MP3 of almost any track that you won't be able to tell from a ripped wav file from the same disc. For me, the quality of the original sample is usually the smallest factor in the imperfections of the audio. Usually, static from the amplifier, and/or inadequate powering of the speakers, and/or crappy speakers, and/or background noise, are the biggest factors. If I were an audiophile freak, I wouldn't rip MP3, but I'm not. I live in the real world. MP3's imperfections are the least of my audio-related problems out here.
Synergy is your friend
I use flac (lossless compression). What was the problem again ? ;)
Seriously, the reason I use flac, even though it takes up a shit load of space is that in the future, inevitably we will have more space to store everything. When we do, thousands will be cursing their crappy mp3s that they ripped at 128 to save space.
Of course, ahem, if you kept your original CDs to rip from then you can just re-rip them to flac or another lossless compression then, but still, why do it all twice ?
graspee
Personally, I can hardly tell the difference between MP3 and CD and my old vinyl. When I play something on CD, then MP3, then CD and listen carefully for the 'crappy bits' I can hear them - but they don't bother me in the slightest when listening to them.
Are my ears just a bit shite? Are most of you guys able to tell the difference - or are the audiophiles just more vocal?
AAC's performance at 64 kbps is not necessarily indicative of its performance at 128 kbps.
I think, that for anyone who would actually be interested in which codec does best on which kind of music, it's a moot point, since by now they delete anything below 128kbps on sight
If you can't see the value in jet powered ants you should turn in your nerd card. - Dunbal (464142)
For those of you that don't know, Opeth's Blackwater Park is one of the most earth-shattering CDs I've ever been privy to witness. They are my favorite band. Check out the last song, Blackwater Park. Wow.
You can get a taste of them in #mp3_metal or #mp3_death in dalnet. type @locator opeth blackwater park and you'll get plenty of results.
Caution - very harsh grunting vocals. May take some time to get used to, but their musicianship is absolutely brilliant.
Berto
To thsoe of us who just want to listen to music on a PC, the newest greatest best algorithms are always good (mp3pro, oggs, wma8). But for many, the goal is to put that music on a MP3Player and listen to it anywhere. I'll summarize the support of these various codecs by MP3Players, as well as mention whether or not my MP3Player (RioVolt SP100) supports them.
MP3PRO -- little support on MP3Players. Not supported by RioVolt SP100.
Oggs -- little/no support on MP3players. Not supported by RioVolt SP100.
WMA8 -- little support on MP3players, though many support older WMA's. Not supported by RioVolt.
So, in summary, all of these new formats are completely useless to me on my MP3Player. The one option they present is if I want to encode something in two formats -- one for my computer, and another for the MP3Player.
Personally, I think more work should go into fractal endcoding, as most music has fractal patterns in it (especially Bach's music).
social sciences can never use experience to verify their statemen
The parent of all of this was certainly in err, but this still isn't a very good experiment. Look at the number of testers that were used. Most tests numbered in the 30s with respect to the number of subjects. While that number may be sufficient for small sample statistical tests, it is not a sufficient sample to test for such a normative value across the human population, such as judging music quality represents. Having achieved a small variance of opinion must not be determined to prove that the sample size was large enough to account for variance in opinion for the greater population, and while these tests are interesting, they are incomplete IMHO.
I am Karma Man, hear me Whore.
An honest double-blind listening test is extremely difficult to arrange, and there is no evidence whatsoever on such on the site.
This is how the test was conducted.
The test required access to a Windows machine (probably Win95 and up, didn't try with Win3.1) with a sound card. Users were required to download the ABC/HR "practice" Zip file, which includes the ABC/HR program, the Ogg Vorbis 1.0 command-line encoder and decoder, a LAME command-line encoder/decoder (I forget which version), a FLAC command-line decoder program, and a .flac sample file (the instrumental introduction to The Eagles' "New Kid in Town").
After unzipping this, the user had to run a batch file (encdec_foobar.bat) which un-FLACced the sample file, then encoded it with Ogg Vorbis and LAME, then decoded both of the resulting files back to .wav.
Then the user executed the ABC/HR program, which is a Win32 GUI application. After loading the sample into the application (pull-down menu and file selector dialog), the interface became a row of double-slider pairs. Below each slider was a "Play" button. Below each slider pair was a "Play Ref" button. Below that was a "Stop" button. There was a pair of sliders for each decoded sample -- so for the practice run, there were two pairs of sliders: one for file #1, and one for file #2. The user did not know which file was Ogg Vorbis, and which was LAME MP3.
The user then listened to the Reference file by clicking any of the "Play Ref" buttons. After hearing the Reference, the user could then click any of the normal "Play" buttons. The first task was to determine, for each pair of sliders, which one was the original and which one was the encoded file. Having determined that, the user used the slider (which went from 1.0 to 5.0 in increments of 0.1) to "score" the sample on the subjective quality of the result. There were also text labels on the slider: 4.0 was "perceptible but not annoying", 3.0 was "slightly annoying", 2.0 was "annoying" and 1.0 was "very annoying".
Finally, there was an ABX button, which launched a different window. In the ABX window, the user could select "Original", "Sample 1", or "Sample 2" for the "A" and "B" samples. Normal ABX testing proceeded from that point. (If you don't know what ABX is, go to pcabx.com.) I found that the ABX window sometimes helped me to focus on a specific sample so that I could find its flaws; armed with that knowledge, I was able to make a determination of which of the two sliders, right or left, was the encoded version.
Once a slider was pulled down from the default 5.0 position, another button became active above that slider. Clicking on it opened a new window with a text box, into which comments could be typed. When the user was finished with the test, the slider positions, the comments, and the ABX results (if any) were written to a plain text file (DOS CR/LF format), which was to be mailed to the test administrator. (Though, of course, you weren't supposed to mail the practice results.)
Now, that was just the practice session, which was a prerequisite for participation in the actual test. For the actual test, the process was similar, but differed in a few details.
The actual test samples included copyrighted, patented codecs for which there are no freely distributable decoders. Therefore, the WMA, AAC and MP3Pro samples were distributed as FLAC files, and decoded by the batch file. Since MP3 did not participate in the listening test, the LAME encoder was not used during the actual test. The Vorbis encoder, of course, was used twice: first with -q 0, and then with -b 64 --managed.
With 5 encodings per audio sample in the actual test, there were 5 pairs of active sliders instead of only 2 pairs. But otherwise, the actual test was exactly like the practice session.
(Personal note: I did 10 of the 12 samples, skipping the two classical ones. Out of 50 encoded versions of the 10 samples, there was only one case where I couldn't tell right from left -- "The Source", encoded with MP3Pro.)
As for theft, since the iPod is only the size of a cigarette pack, there's no reason to leave it in the car to be stolen when you go inside somewhere -- that's what pockets are for, chief.
or 80 and 96, comparable to 160 and 192 for mp3s respectively. Any lossy codec is going to break down at some point. The question is where and how quickly? One way is to design a codec to perform as well as it can at some bitrate, determined by needs. But for music, when your goal is to have audio that is virtually indistinguishable from the original rather than merely audible, the goal is for it to be as indistinguishable as possible for as low a bitrate as possible (which means you have tradeoffs). Below that, who cares? A codec that can do that is not necessarily going to scale properly to very low bitrates. And I have yet to hear of any codec that can handle near-indistinguishable music at 64.
---If you can't trust a nerd, who can you trust?
I have to agree. Not showing the rating of the control in the charts makes the chart pointless. It would be interesting to see how the user ratings for the control line up against the codecs.
It's possible that the data from the control showed that the listener preference had little to do with how the rated the codec reproduced the original.
* As is generally the case, my opinions do not reflect those of my employer.
It looked to me from the results that mp3pro performed about equal with ogg at 64K, but would you really want to play 64K files in your car ?
I know that the thread is about compression formats, but hey - go to a bar/club with "LIVE" music, pay $10 at the door, have a drink and a good time.
Hopefully, the guys playing are getting a percent of the door, and they'll be happy to see you in the audience. Feel that bass!
Here's what I do: Bitty Browser & Andromeda
Check out Hydrogen Audio
Its pretty much the best audio discussion you can find on the 'net.
Apollo 440 is actually a prety damn good test bed for this stuff, especially (in my opinion) the Rapid Racer theme. Vocals, guitars, high frequency synth and ultra-low "theta-bass". Of course, you could have just used one of those Dynamic Frequancy test CDs, but what fun would that have been?
You need a FREE iPod Nano
Running any car over a cliff will destroy it, regardless of safety systems. This is not a valid test of a car safety system.
I'm the stranger...posting to
Depends where he is cycling, around me we have some bike paths that run along old railroad lines. Outside the very popular parks where there are large numbers of parking spots for cars there aren't many people. In fact some of them are remote enough I have had complete 5 miles hikes where I have seen 2-3 people the whole 2 hours. At least around me most people who are serious about hiking or biking do it to get away from people, life etc as much as the exercise.
There are 4 boxes to use in the defense of liberty: soap, ballot, jury, ammo. Use in that order. Starting now.
Whatever. You claim vorbis is an audio codec and ogg a bitstream manager, but you don't tell us what the sound format is. Why is the executable called oggenc (ogg encoder) if ogg is not a sound format? Why are the files name .ogg by default if ogg is not a sound format? Why don't you go troll on the xiph mailing lists, because it appears that they are the ones who have perpetuated this.
This is an interesting and relatively well done test (although it appears that the listeners knew which format they were listening to, so it wasn't truly double-blind, and a anti-MS and pro-Ogg bias can't be ruled out).
However, some discussions seem to be focusing on this saying AAC is bad or WMA is bad, when really it refers to the particular implementations in codecs of those formats.
For example, the Apple MPEG-4 AAC-LC encoder was used for AAC. This is a Low Complexity version of the format. Also, the Apple encoder has a strange limitation where it automatically converts 44.1 stereo to 32 stereo at that data rate. This isn't required by the AAC format. Other AAC encoders yield MUCH better results, and beat MP3 Pro in double-blind testing. I haven't seen any double-blind comparisons between AAC and Ogg.
Also, the WMA8 encoder is due to be replaced by the backwards-compatible WMA9 in early September. Of course, there may well be improved versions of the other encoders by then as well.
My video compression blog
I'm sorry if this has been said before, but I couldn't find it:
Why didn't they also try playing the original, uncompressed music, to see how high it scored...?
RMN
~~~
mmm the warm bass... only type of music i ever noticed a vinyl difference was for music with loud or enhanced bass range... it just seems "warmer" in vinyl.
;-)(
I post, therefore I am.
CAn'T CompreHend SARcaSm?
>Basically, after the first listening, you have
>already instinctively 'decided' how the sample
>sounds.
If this were true, it would never have been possible to make conclusions with 95% certainty, and there would have been no correlation between the results of the listeners.
Both are false, and hence, your unsubtantiated claim.
>How many times were the tests repeated?
One test per listener per sample.
>Was there a control group?
There was a control group in the form of 'trusted' listeners also participating in the tests. The results were similar to the general ones.
>Also, there is no information on the sound
>system used and no measurements on its effect on
>the test.
If you had bothered to actually look, you would see that each tester used the system he generally uses to listen to music on his computer, whatever that was.
>My suspicion is, if the test were repeated, the
>result would be completely different.
There is nothing that supports this claim and past tests contradict it.
--
GCP
>I do not know why Vorbis tried to compress those
>two so much more than the others.
Vorbis with the -q mode tries to maintain a certain quality level. The other codecs try to maintain a certain bitrate.
On an easy clip, Vorbis uses less bits because it can do so and keep reasonable quality. The other codecs use more bits because they try to reach the certain bitrate.
--
GCP
It would not matter if ogg was capable of reproducing music at any bitrate, no body will care with a name like that.
It would not matter if mp3 was capable of reproducing music at any bitrate, no body will care with a name like that. Heck, the middle syllable sounds like something you do in the bathroom.
The name of a product can make or break its success in the consumer world
Coca-Cola sells, even though the first half of its name is the first half of "cocaine". Trust me, any name can be promoted by the entertainment media.
As it is now, [the name of Ogg technology is] based on two bits of extremely nerdy trivia
And "Motion Picture Experts Group Audio Layer 3" isn't?
Will I retire or break 10K?
No integer only codec. This means no portable player support as they generally don't have a FPU.
Nerd: Derogatory term typically directed at anybody with a lower Slashdot ID than you.