Vorbis And Musepack Win 128kbps Multiformat Test
technology is sexy writes "After 11 days of collecting results Roberto Amorim today announced the results of his 2nd Multi-Format listening test: Vorbis fork AoTuV scored the highest and ranks as the winner together with open source contender Musepack closely followed by Apple's AAC implementation and LAME MP3, which improved markably since last year thanks to further tunings of its VBR model done by Gabriel Bouvigne. Sony's ATRAC3 format ranks last after WMA on the third place. The suprising success of AoTuV (compared to last year's performance of Xiph.org's reference implementation) shows the potential of Vorbis and possible room for further tuning and improvments. Take a look at the detailed results and their discussion at Hydrogenaudio.org."
So given Microsofts stated goal to bring us innovative technology, they should throw in the towel and ship OggVorbis and derivatives with Windows, right?
And if you thought that was boring you obviously havn't read my Journal ;-)
When everyone gets an iPod, dood, or the WinFooTunes player that you get with your Dell only works with WMA, or your in-dash CD player only groks 128kbps MP3s, whats the practical application of the other codecs? It's nice that we propeller-heads on Slashdot can smirk while we rip everything to FLAC and write custom Perl apps to transcode-on-the-fly to our wireless enabled MythTV box, but for John Q. Drone^H^H^H^H^HConsumer, none of this matters.
So how do we get the word out? How do we start the revolution? Open-Source hardware?
I want to delete my account but Slashdot doesn't allow it.
No matter what researchers find the best format, the best format for users is what they can doubleclick to play, use on their el-cheapo portable mp3 player or whatever music device they own.
This might be of interest to musicians but the proverbial "jane doe" will keep using mp3 for quite a while
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The open source ones don't have the big push amungst the general population. So, number 3 on the list Apple (ACC) can say in independent tests ACC scored higher than WMA or MP3. The top 2 don't have the marketing push to get out and be popular in the general population.
This does give more fuel to Apple. Although I'm not complaining about them having fuel over Microsoft.
Evolution or ID?
Good to know MP3 is still improving. Yes vorbis and others are great, but i know every software and hardware player out there plays MP3. I'll be ripping all my cds to high quality MP3 befor i go to college, not because its the absolute best, but because its a standard. Standards aren't always the most efficient, but their strength lies that you cant change them on a week to week basis. Whatever hologrphic storage based finger sized half terabyte 24th generation iPod i buy ten years from now will probably still play my 128 and 256 MP3s.
"Sic Semper Tyrannosaurus Rex."
So given Microsofts stated goal to bring us innovative technology, they should throw in the towel and ship OggVorbis and derivatives with Windows, right?
Microsoft will never. They will take the code from the #1, put a DRM to it and ship it as the next version of WMA. If they can't make the best they buy or take the best and make it their own. (with some tweaking of course)
Evolution or ID?
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Because this was designed to test various codecs at 128Kbps. You can't make flac do 128Kbps. Besides, flac, being lossless, sounds exactly the same as the source media, so what's the point of testing how it sounds?
Ah. My mistake. WTF? Why are you even listening to lossy codecs then?
i realize the geeks of the populace want the highest quality encoding to win. naturally. and it helps when something such as vorbis is rated so highly; it gives it even more geek cred.
however: as someone who studied music and audio, i am constantly surprised at what people will listen to. my friends (well some of them) have no problem cranking low quality mp3s of 50 cent, while i drop my jaw at the poor audio quality as a result of lost information. one time i even remarked to my dad "oh its an mp3" when he was playing something i had given to him which had been apparently later encoded. he wasnt sure (he didnt do the encoding) but doublechecked and yes it was mp3 (probably 160 kbps). he was impressed, when to me the timbral change in the cymbals was a dead giveaway. another time i asked a friend of mine if he was using aac to import all his cds in to itunes when he had been recently doing so. he looked at me blankly and said "whats aac?". which meant, yes he was.
i apologize for rambling, this is what im arriving at:
despite early adoption influence etc that geeks hold, how much does all of this really matter. most people dont care what format its in as long as they can listen to it. and often they cant discern loss of quality unless its extreme. so while i applaud these efforts, im simply wondering if -- aside from research -- they arent futile.
Assuming FLAC is truly LOSSLESS:
.zip .tar etc., not an encoder... technically.
1) Is it really a codec? Seems to me it is a compression method for media, like
2) It should sound exactly like the original. LOSSLESS = no loss. No point in comparing it to lossy codecs, unless it's not truly lossless.
3) The stored file sizes although smaller than the raw music are still way to big to be portable IMO.
Post: Sigged, for your pleasure.
It is interesting that the note that they used the AAC encoder in iTunes 4.2 instead of the newer 4.5 because of "quality" concerns.
Apparently there's some "high frequency ringing" going on.
Better stick to something else for now, if planning to rip to AAC.
I read some of the results, and I'm not a Vorbis hater or anything, but how much of this is open source fans voting for their favorite codec? I looked at the test just now, but can't tell if it was blind or not.
*ducks*
it isn't everything. Microsoft still has enough cash to fight this. Our local radio (Switzerland) still broadcasts in .asx. I sent them an e-mail asking them why. They said because their server is sponsored by Microsoft.
Now I listen to virgin radio, they broadcast in broadband ogg
1) Is it really a codec? Seems to me it is a compression method for media, like .zip .tar etc., not an encoder... technically.
Umm... technically, by definition, yes it is a codec. enCOder/DECoder. It encodes a WAV file into something else, which happens to shrink the file size, and can then decode the something else back into a WAV file, restoring the WAV file losslessly.
That really doesn't look very fair to me! MPC and Vorbis using about 20% more bits than Lame and iTunes AAC.
Lalala
Compare this with radio. There are a lot of popular AM and LW radio stations here in the UK even though FM is a superior format. MP3 will be around for almost ever due to the popularity and level of takeup.
Omnis amans amens
What we need is open source marketing. The open source community may be great at producing programs but at marketing them, well that's another story. We need open source marketing.
Evolution or ID?
FLAC compresses to what, 5-600k? The only way you could make it fit the competition would be to mutilate the original WAV, and trust me, you wouldn't want to do that.
Certainly, it's a good reference if you want to compare across bitrates (what's 128k vs 256k vs lossless like in quality/size) but it has no place here.
Kjella
Live today, because you never know what tomorrow brings
1) Is it really a codec? Seems to me it is a compression method for media, like .zip .tar etc., not an encoder... technically.
"Codec" means "coder-decoder". FLAC sounds encoded to me, if you need a FLAC library to enable a piece of music-playing software to read it, then I'd say the FLAC library is a codec.
2) It should sound exactly like the original. LOSSLESS = no loss. No point in comparing it to lossy codecs, unless it's not truly lossless.
Actually, it's interesting to compare lossless and lossy compressions because, these days, there's a fair chance that very good lossy compression sound so good it's almost impossible to tell the difference with the lossless compression.
3) The stored file sizes although smaller than the raw music are still way to big to be portable IMO.
Depends how much smaller. I'd say anything that doesn't produce at least 5x compression is worthless in any music player. You can zip a wav file and despite being much smaller than the original, it will still feel worthless to you in a compactflash card in terms of size.
There is a Debian repository at rarewares where you can get the aotuv version of vorbis enconder. Grab yours today!
One way I see Vorbis making it into the mainstream is if there were high availability of Vorbis content on the net. This includes P2P channels as well. If music releasers in the underworld start adopting vorbis, then Joe 'I own the original CD' Downloader will get a far wider familiarity with the codec, same as to what happened IMHO with xvid. More content will eventually lead consumers to start demanding vorbis compatibility in their hardware.
What is it? I followed the link, but there was no info about it. It appears to be open source though!
No, they won't. Their definition of innovation is making the same thing in an incompatible way.
Please correct me if I got my facts wrong.
Because I want to fit more than two songs on my MP3 player. If the encoding is good enough, then it is indistiguishable from the original are close enough. I don't want to dedicate an entire 80GB drive just to house my music collection.
Just a Tuna in the Sea of Life
Well, sure; I don't want to devote 90% of my 512M CF card to a single album, but for my desktop it's perfect. I just transcoded an album from it to Vorbis (this is where it's nice that FLAC is so effecient to decode) -q-1 (that's right, quality minus one) and it sounds fine on my iPAQ. FLAC reduced the filesize from 683M to 466M, which is fine for my desktop; Vorbis reduced the filesize from that to 23M. It's not the sort of thing I want anywhere near my desktop with it's 90UKP soundcard and Sennheiser headphones, but for a portable device it's fine.
In future I hope portable storage scales up to the point at which I don't need to worry about encoding to lossy formats though. In the mean time, having my source as FLAC means I can transcode to whatever format best suits the available technology; from burning an exact CDDA copy for my CD player to a bunch of MP3's for my DVD player to a bunch of ultra low bitrate Ogg Vorbis files for my iPAQ. That'll do me fine until I can get a 256G CF card
FLAC has an extensive testing suite if you're not convinced it's lossless, btw.
"I don't want to dedicate an entire 80GB drive just to house my music collection."
Glad to hear it. But i sure don't want to have saved a small (think future with me) amount of space at the cost of crappy audio. Your call.
It would have been nice to have an original unencoded piece and rate it against the masses. That way we'd be sure the listeners weren't picking up on a mastering problem that is muffled by an encoder.
IMHO, the best way to test is to provide an uncompressed source and a variety of compressed files, and ask "which most closely matches the uncompressed source" -- and NOT "which sounds best."
Years ago, I did an a/b switch test with a high-end audio engineer between a CD and a 128kbit/s MP3. Though we could both clearly hear a difference, he actually guessed wrong.
My point is: the test needs to be blind, and the test should be looking for compressed files that most closely sound like the uncompressed original -- and not the ones that "sound best."
Here's what I do: Bitty Browser & Andromeda
It could also be marketing.
MP3 players got *heavily* marketed after Napster and friends got press and serious college use. "MP3" became associated with "free music". They took off.
The iPod, a decent but not earth-shattering MP3 player, sold *much* better than other MP3 players out there. Why? Marketing. Lots of ads -- the only significant difference to cause such a change.
Vorbis doesn't have a lot of ad money behind it pushing it.
I'd also like to point out that:
* People still use CBR MP3s. CBR was designed for exactly one reason -- allowing constant-rate streaming. It's *stupid* to use CBR for locally stored files -- it gets significantly worse quality for the size -- I've generally found that on the music I listen to, using VBR is equivalent to at least a 30% increase in bitrate in terms of my ability to distinguish between a master an an MP3. If people cared about quality, CBR MP3s would not exist. They wouldn't even have to switch their hardware/software around, since it's the same format, but they won't even go that far.
I *really* get a kick out of it when people buy an MP3 player and a pair of high-end earbuds. It's just plain inane. They just purchased a low-quality audio playback device and then spent a huge amount of money on an expensive pair of earbuds that don't let them hear the now missing nuances of the audio. It's the ultimate in trendiness -- like buying Nike or Banana Republic clothing. iPod + expensive earbuds is not "the ultimate in sound reproduction" even if you really, honestly gave a lot of retailers a whole lot of money for the combo.
May we never see th
What if a lossless codec were included in the test - and it came in dead last?
That would provide useful information: either the listeners weren't up to the job or the lossy codecs at ~128 kbps were truly indistinguishable from the source material.
"there's a fair chance that very good lossy compression sound so good it's almost impossible to tell the difference with the lossless compression."
Simply untrue.
What you might be saying is "using my earbuds, my ears can't distinguish the difference between the CD and 128kb AAC".
Really, that's pretty thin justification for cheerleading for iTMS (which sounds pretty thin).
Having designed and written a mp3 decoder and now working on a vorbis decoder I can't say I'm that suprised by vorbis coming out on top.
From a technological standpoint the Vorbis codec has 10 years of audio compression R&D in it since MP3 was invented.
MP3 is a subband DCT based codec using fixed window length. Vorbis is also DCT based but encodes an approximation to the orginal frame's spectral curve and also uses variable length window length.
In using the source from the vorbis library and the decoder specification to help guide its development I have to say it is a real joy to code. The people at xiph.org have really done a first class job and have approached some of the problems of audio codec design with some of the best lateral thinking that I have ever seen.
Believe me! Coming from me that is very rare praise.
Testing FLAC is useless, as most of you say.
But I think it was convenient testing any kind of lossless (FLAC or WAV).
The audience may have giving slanted scores because they know the clips suffered lossy codecs, they were hearing artifacts the way people hear "voices".
It would have been interesting to know if lossless scores 5.
And a codec wich really improves music would help many artists today XD
First -- it's only crappy if the difference is noticeable. Icode my music in a lossy format, but of course enough so it isn't noticeable when using headphones. That's good enough for me, and as a bonus I've saved the disk space. In the future 80 GB might be little, but it isn't today and that's what matters as you buy hard drives.
Beware: In C++, your friends can see your privates!
what's the point of testing how it sounds?
To see how much worse the lossy codecs are in comparison?
Beware: In C++, your friends can see your privates!
1) Is it really a codec? Seems to me it is a compression method for media, like .zip .tar etc., not an encoder... technically.
.zip/.gzip are very poor at compressing audio recordings. FLAC and similar lossless audio compression look for things like (I would imagine) relatively small deltas from each sample point to the next, since this is a common characteristic of audio data.
Compressers are encoders of a particular variety. They just choose a different data representation as an encoder does, but make an effort to take advantage of specific known characteristics of the data they are compressing to get a smaller, reasonable representation..
ZIP and gzip (tar does not do compression, just file joining) do very poorly at compressing audio. They do things like look for patterns of repeating (or at least commonly seen) sequences of data, and simply say something like "every time you see "z1", I really mean ";lt&a href="". This approach often works very well in computer-generated files.
However, it's very unlikely that you will get exactly the same sequence of bits in an audio recording, so
FLAC is indeed lossless.
May we never see th
LRC, the best-read libertarian site on the web
If a sound was perfectly accurate except for an instantaneous annoying pop every few seconds, it would probably average as the best codec, but it would be useless as a consumer standard. I remember a codec shootout years ago where Mp3Pro sounded "tinny," WMA sounded "flat," and MP3 sounded "fuzzy." Was being objectively closer to the source material more important than the type of distortion introduced? Not at all.
When dealing with sound equipment, from pre-amps to encoders, the tone of the introduced distortion is very important. Everything introduces distortion, in some way or another. You just want it to make the sound better, not worse.
The ______ Agenda
Well, I can only comment from a technical point of view, but firstly it is very good news that we are progressing in the right direction in terms of quality. Secondly, compared with the other codecs (esp. the proprietary ones), Vorbis is quite simple and minimalistic and lacks a lot of advanced tools and profiles, yet we've been able to extract quite competitive performance from some adjustments here and there. There is more to do in Vorbis and Monty has some new ideas that he wants to implement in the next major version like a better stereo model, noise normalization (which in its current form is mostly experimental), and support for 5.1 stereo. Given the success of aoTuV and the fact that Monty is fully aware of these third-party tunings, I think Vorbis development is looking ever-more exciting. :)
(Note I don't work for Xiph.Org but just one of those third party Vorbis tuners)
An 80 GB drive should hold nearly 200 (or more) FLAC encoded CDs. That's CDs, not tracks. This is assuming that the CD is nearly full, and the FLAC compression averages out at its usual 60% of the original.
Lossless is a more viable solution as we get larger hard drives and faster Internet connections.
I guess that if you are like many people and you have a ridiculous collection of pirated songs though, FLAC may not be a good solution.
I certainly cannot tell a difference between FLAC encodings and the original WAVS. The raw output looks identical if you examine the waveform.
1) It is a codec. You encode/decode from raw PCM/WAV to another format.
2) Sounds lossless to my ears. This is because it uses a compression format that is similar to ZIP/TAR.GZ/ETC., as you suggest. Nothing is cut out in the process. MP3 and other lossy formats make it a point to strip the highs and lows that the human ear cannot normally hear.
3) An average of 40-50% savings on a fully compressed FLAC file are very much worth it, in my opinion. Most people these days have many (hundreds?) of gigs of storage space. As I noted above, 200 (or more) full albums in FLAC format should fit on an 80 GB drive. I can normally fit at least two whole FLAC albums on a single 700 MB CDR for archival and backups. It's really hard to tell though. Certain types of music, like classical, compress better in FLAC format than noisy rock n' roll. You never really know what you are going to get, but on average - it is about 50%-60% of the original size for most of my music collection.
according to this, 54 results was discarded because they ranked the reference file, instead of the encoded file. If flac was to be included in such a test, I'm sure it would have won, but im also sure it wouldn't have scored a perfect 5, even if it should have.
Recently I decided to use lame with cbr 192kbps after comparing to the preset vbr settings (including extreme). I use the settings: --cbr -b 192 -h -q0
Using vbr I can hear the noise floor being modulated e.g. by a large amplitude low pass filtered bass sound. I contribute this to vbr changing bitrate. Maybe the psychoacoustic model just doesn't fit my ears:-)
The vbr files average around 200kbps anyway, so they're not smaller than 192kbps cbr.
It would have been nice if the test included cbr as well.
I use good headphones: Sennheiser HD-25, and my mp3 player: archos jukebox recorder running the open source firmware Rockbox
"what if people liked the 128kbs mp3 version better than the cd recording!"
People like to pierce all parts of their body too. All we can learn from this practice is that 50% of the population is below average.
And people listening to 128kb lossy music and saying "they sound just like the CD" are in the lower part of that bell curve.
That's hardly insulting, its just reflects the statistical model here.
"You can critise the 'timbral change in the cymbals' but why do we worry so much about this when, in many cases came off a synthesizer anyway."
Not everybody listens to pop music.
Some people like orchestral works, or baroque music or choral music, or bop, or dixieland, or a whole host of genres where synthesizers are never (or rarely) used.
If a lossy codec sounds fine, then you should be glad; you've found something inexpensive that works well for you. For other types of music, even CD's struggle to maintain good sonic quality and so my frustration is that we're currently undercutting quality for more convenience.
You were mistaken. Which is odd, since memory shouldn't be a problem for you
libvorbis-aotuv-dev_0.b2-1_i386.deb
libvorbis-aotuv-0_0.b2-1_i386.deb
libvorbisenc-aotuv-2_0.b2-1_i386.deb
libvorbis-aotuv-0_0.b2-1_i386.deb
libvorbisfile-aotuv-3_0.b2-1_i386.deb
oggenc-aotuv_1.0.1+aotuv-2_i386.deb
You don't even need to uninstall your existing vorbis packages.
Calling it "Vorbis" in the title sort of makes it seem like you're pimping a certain format, rather than offering an unbiased pointer to some information.
If the creators of AoTuV hadn't considered their work important enough to require distinguishing from Ogg, they probably wouldn't have gone to the trouble of forking a new project from it. If you're going to put their work on a pedestal as exemplar, at least afford them the respect they deserve by acknowledging the real name rather than referring to its catchword roots for whatever reason.
a post: the reply:Disclaimer: I am NOT new here
I think you are way off here.
Firstly, a number of portable players support Ogg Vorbis. There is a list of four here, I'm sure the number will increase.
Secondly, I'd doubt that many of the public know about Ogg Vorbis, let alone consider it to "reek[s] of piracy and hackers".
Furthermore, the "success" of P2P music sharing indicates that the public are the last group of people to have morals about the source or the format of the music they listen to.
Ogg isn't as widely used by the public, because it is not known by the public, it is as simple as that. That will change, as more and more players support it, and the public find out that it is a DRM free alternative to the flexibility restricted formats such as AAC.
The Internet's nature is peer to peer - 20050301_cs_profs.pdf
I'd wonder why songs are encoded from 16 bits / 44100 hz, while our ear can hear a difference with 96 khz/32bits and consumer sound cards can often do 48khz ? (if you can do better than rip from a CD, that is)
I mean, would 160 kbps sound better from 44100 or 48 / 96 khz ?
It seems this WMA9 encoding type has been omitted, so it's not a complete research to me.
The MPC codec was neck-and-neck with Vorbis most of the time, except for one song by Debussy. What is interesting though is that it only encoded at 91kbps for that song---suggesting that perhaps if it were forced to use more bits it might have scored higher. It seems the heuristics it uses to determine how many bits it needs didn't quite work for that song.
Ive listened to mmp3, ogg and wma versions of the same song and can't tell the difference. They all sound the same, so I just encode into the smallest format which is wma
Apple's AAC implementation uses ABR rather than a purely VBR implementation. Look at the bitrate distributions. Ogg's bitrate was significantly higher than iTunes' in the majority of listening tests.
The way I understand the testing methodology (and I could be wrong):
.wav format. There are several reasons for this: 1) it prevents the testers from deducing anything from the file sizes; 2) it eliminates the need for the testers to have access to all the codecs; and 3) it simplifies the design of the ABC/HR+ testing tool.
--- The files that the testers received were all uncompressed (or compressed with a lossless codec). In other words, they each had been run through their respective lossy encoders and then decoded back to
--- The tests weren't pitting the codecs against each other per se, but rather each against the uncompressed original. So there is no sense in including a lossless codec in the test because, well, it's already there in the form of the control file.
Again I could be wrong about this, so please correct me if I am wrong. In any case, there is no sense in testing FLAC here.
All these codecs work on audio data that has to come from somewhere, typically CDs. Xiph offers the standard CDDA->WAV converter that feeds LAME and the other codecs. Sure, CDParanoia III v9.8 works very well, but it's slow: typically the best error recovery cuts performance by at least 65%. The Paranoia news page says
;). So it's hard to demand a new version, even after 2 years. But a lot has happened, especially in performance opportunities, since that release. So where's Xiph's committment to the expectations they created with that announcement? Where's the community effort to advance this essential tech?
"March 27, 2001
Things on hold for now: No, that doesn't mean the project is dead, just that active development is on hold while we throw all the time we have available to get OggVorbis to 1.0 in a reasonable amount of time. Once Vorbis hits 1.0, we'll get back to Paranoia."
Vorbis hit 1.0 22 months ago:
"README 1.14 22 months xiphmont That's it. Full 1.0 libVorbis code handoff to release engineering."
Of course, this is OSS, and free, to boot (pun(s) intended
Hacking the paranoia lib for device performance tweaks isn't my bag, baby. I would help another, lead, developer, by testing, or higher level hacking, or project organization support, etc. This library is the bottleneck for media format freedom - everyone on unix/linux has to use it, in one form or another. That widespread use usually drives progress in OSS. Where's the love?
--
make install -not war
But you can just as well compare lossy compression against the original file...same thing.
For example, the final sample "Waiting" the winning codec MPC was actually 153kbps, followed by LAME and Vorbis at 144 and 148kbps. Unless I don't understand his algorithm, it doesn't seem as though he is taking these bitrate differences into account
The report claims to be a 128kbps Multiformat Test, and it's really a "128kbps standard preset on our encoder" Multiformat Test which may result in higher bitrates than 128kbps. I am sure that if you gave the static bitrate codecs the same bitrate as the resulting VBR average bitrates, they would be much more competitive.
Yawn.
The way it works is, you listen to a given music clip. You have three streams to choose from. One is the uncompressed .wav, and is labeled as such. The other two are not identified, and consist of the compressed source and the original source. You then rate the two unidentified sources based on how closely they approximate the original. Then you repeat the process five times for each of the codecs. When you're performing the experiment, you don't even know which codec you're testing at any given time.
I'm a little surprised that they didn't including HE AAC, a more recent MPEG-4 audio codec. It's best known in its AAC+ implementation from Coding Technologies. It's definitely entertainment quality at 48 Kbps at 44.1 stereo.
Anyone know why it got left off?
We also have parametric stereo AAC coming as well, which should be able to do entertainment quality at 24 Kbps.
My video compression blog
Bear in mind that WMA9 "Standard" is a legacy codec for Microsoft these days. They haven't revved the bitstream in years, in order to keep backwards compatibility.
There are new WMA9 codecs, though. WMA9 Professional, which goes up to 96 KHz 24-bit 7.1. It does have a 2-pass VBR 128 Kbps 44.1 stereo mode, and it'd be interesting to see that included in a future version of this test.
There is also WMA9 voice, which is better thought of as "WMA9 narrowband" since it includes a music mode as well. Only up to 22 KHz mono @ 20 Kbps, though, so it's really meant for modem streaming kinds of applications.
My video compression blog
If you don't understand a post, don't moderate it.
Actually, you'd get about 2:1 compression with Apple's lossles codec (for whatever fundamental reason, almost all lossless codecs, video and audio, work out to around 2:1 compression on average with natural content).
And since the average audio CD isn't full, I imagine you could get nearly 150 CD's on an iPod in that mode.
Still, I happily use 320 Kbps MP3 on mine, same as I use on my server for SqueezeBox access. I used to convert to 128 Kbps AAC for the iPod, but that was too much management trouble. Now I can just copy the files.
My video compression blog
So now at at the 128K level, the difficulty in perceiving differences is showing up on the graphs. Since portable players are coming out with more and more space these days (40 Gig iPod anyone), why not test at 160 kbps or 192kbps and then see if there is even a perceived difference. The file size from 128kbps to 160kbps is only going to increase by around 25-30% but you'll get a nearly indistinguishable file from the original.
I'd wager that at 192 kbps on any codec (except ATRAC by the looks of the numbers), only the real Golden Ears (TM) can hear the difference.
What exactly do you mean by "Don't touch this button?"
That seems like a questionable methadology to me. If the use couldn't tell the difference, it seems like that should be an automatic 5. Dropping cases where there isn't a perceptible difference woud tend to underrate the quality of the best encoders.
My video compression blog
The two winners had the highest average bitrates, 135 and 136 kbit, giving them an advantage over the encoders that outputted the correct bitrate.
Though I think that this is a great idea for a study, aren't the conclusions found by this research ultimately meaningless? In his overall ratings graph, there are a total of eighteen samples and the other track-based analysis have fewer samples than that (N=17, N=15, etc.). This isn't even close to what is required for a good sampling of the US population! (I believe you need about 1,500 to get even close for most statistical comparison of US population.)
Add onto this the fact that no random sampling was done and this test is even more meaningless because of the self-selectivity of the participants. People who were sampled were most likely those with active Internet connections and those of the audiophile and/or Slashdot-type "Nerd" communities online. Naturally, we are more accustomed to hearing vorbis as a file format. Perhaps "Joe Blow" down the street hears a very different tune, but this method of sampling doesn't account for that, because Joe Blow never even heard about participating in this study.
Granted, that might be okay since I think perhaps the self-selectivity of the participants yield results that will match up with those who actually care, but I still can't see how the insignificant sample sizes used for this research can have any real meaning.
You mean, "Other people just might have different needs and desires from yours when it comes to compressing audio."
That seems like a questionable methadology to me. If the use couldn't tell the difference, it seems like that should be an automatic 5. Dropping cases where there isn't a perceptible difference woud tend to underrate the quality of the best encoders.
The problem is that it's difficult to develop a consistent and fair method of determining what to do with results where the reference was ranked. Maybe just assigning 5 to these cases is one way of dealing with it. I can see doing this for someone who marks 4.9 for one of the references, but what about somebody who scores 3.5 on multiple references (I'm exagerrating, but not too much!)? Is it wise to even keep results from such a listener? I think that as long as there are enough results, which seems to be the case here, the fairest and most consistent way to deal with these cases is to simply discard them.
ff123
I own about 250 CDs. To be able to conviently store that amount, I need better compression. As I also don't want to have to do some on the fly conversion to MP3 so I can load songs onto my RIO, Ineed them to be in MP3.
Just a Tuna in the Sea of Life
Get ripping!
But this is the day and age of the Internet, where WinAmp took off like a rocket and made its author very rich. "Ad money" shouldn't be a factor.
There is something inherently wrong with Ogg Vorbis. Think about it: Despite being free, none of the major manufacturers (Panasonic, Sony, Phillips) are embedding it in their electronics, so you get also-rans -- "iRiver"? "Rio"? Not exactly household names.
Possible problems:
1. Firmware instability. The list at http://wiki.xiph.org/VorbisHardware says stuff like "firmware 1.41 out, fixes some problems with Vorbis files", "iRiver flash players support Vorbis with firmware update. ", "Note that firmware versions prior to 1.25 cause stability problems for some people", and "There are reportedly problems with some versions of the firmware...". Is that a result of the algorithm? Why are they having so many problems writing stable firmware? I read a review of a Rio player where the reviewer complained about lockups and warned readers not to buy it.
2. Floating point format , ie Ogg Vorbis mainpulates everthing in floating point format. While this is no problem for a standalone computer, this is an additional expense for a portable battery operated device, because you have to have an additional floating-point coprocessor. This extra chip
eats up the battery more quickly
adds to the expense of the device.
Alternatively, you convert everything to fixed point, and thereby lose the fidelity of the original Ogg Vorbis result. Indeed, is it possible that the internal floating point is the reason that Ogg Vorbis has a better sound fidelity than MP3 simply because the error quantization is less in floating point numbers than integer discretization?
wow.. atrac3 sucks badly.. but they didnt even bother to rate atrac3plus It is like 48Kbps and sounds better than atrac3 132Kbps.
of course, its very heavy into the DRM (like all sony stuff), which kills it, but the compression is great.
It would be a control codec, then. That being the case, you might as well just use the actual source media.
It's odd to keep hearing this code referred to as a 'fork'. Yes, it's based on our reference code while doing further tuning just like all the free MP3 encoders are based off of the original dist8 or dist10.
Fork seems to imply that they're trying to make something incompatible or doing it without our blessing. Neither is true! We never wanted to have *the* only encoder. Nor did we want to be the only people trying to improve Vorbis's encoding.
AoTuV is a 100% real Vorbis encoder and the results of the test speak for themselves. Aoyumi and crew deserve kudos, and I'm glad to see them working on improving Vorbis encoding.
Monty
The speed of CDparanoia is not limited by the cdparanoia code. Regardless of future improvements, speed gain would never be one of them. If anything, additional error correction would only ever make it slower. Your limit is Linux forcing programmed I/O because the IDE subsystem doesn't know how to use DMA on non-multiple-of-512-byte sector sizes. CDDA is 2532 bytes per sector. Linux 2.6 partially fixes this.
;-)
Also, cdparanoia (III) was finished long ago. It has not bitrotted. As new kernels came out, we+others kept it up to date. The distribution maintainers have added whatever fixes have been necessary for their distros. Nothing that worked in 1999 is broken today.
In summary... paranoia does 100% of what *I* need it to. I write software that I need. I don't have to keep releasing 'improved' versions of software that already works as an ego-trip or to placate a marketing department desperate to sell you the same thing in a new box every six months.
Others have expressed interest in doing new things with paranoia, but no one has followed through... at least not yet. Paranoia isn't all that complicated to use or hack. That speaks to a pretty damned low demand for new versions.
The website: yah, OK, I'm lousy at writing HTML updates. My diary hasn't been updated in three years. There is certainly a website attention span problem
Theora: I'm not one of the primary coders today, I only did the initial code import. Also, the Helix project has required relatively little time; Real has done nearly all the heavy lifting on integration there.But, if 'Theora is dead', why does CIA show 500 commits in the past two months?
DirectShow issues can be summed up as 'ugh, what an awful system'. But we'll make it work. The discussion about mux was proposed changes to spec. Voluminous discussion reveals what we have now is still the best option, as designed five years ago.
Monty
It's nice to hear that, but is cross-polination going to happen?
How much are you willing to delegate to the non-xiph tuning crew?
Monty, I think I remember you mentioning that cdparanoia was designed to work with CD-ROM drives that had 1 meg or less of cache. Most CD-ROM drives have more than 1 meg now.
Will an update for modern drives be released, if cdparanoia is not so difficult to hack..?
Since that is what it essentially is...a highly tuned version of the reference encoder. That is probably why Garf called his encoder GTune (Garf Tuned) and I've followed suit with QKTune. Also, aoTuV stands for "Aoyumi's Tuned Vorbis" IIRC. :)
Vorbis needs more working binaries!
.ACM codec for Vorbis (and for Speex). Let people try out these free codecs using Windows Media Player or Sound Recorder, WITHOUT making them download and install more bloatware or broken freeware crap. (And note that some non-Microsoft media players also support .ACM files.)
On vorbis.com, the only Windows players listed are Winamp 2, Winamp3, foobar2000, and zinf. THAT'S IT?! Only four (three?) Windows players support Vorbis??
There should be a WORKING
I told you it was RIGGED.
.
The parent, GP, GGP, GGGP, etc., are all off-topic. Get off you're asses and do something about it!