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What Sounds Better, MP3 or Ogg?
I've never been able to make a clear decision on the subject. These days I rip all my CDs to MP3 at 160kbs which means about 80 megs for a longer album. With a 100g drive on order ($220. I remember paying more then that for .1% of that space) disk space isn't really the defining issue, but that doesn't mean I'm gonna rip everything at 300kbs just because I can. I'm curious what people think sounds better, and what bit rates they find to be acceptable for both casual listening, and more picky listening. Don't forget to mention what sort of equipment your listening on so we know where you are
coming from.
Ogg sounds better, but I can't go to walmart and buy a portable Ogg player. Hopefully this will change with some reprogrammable units. Anything like this on the horizon?
Kindness is the language which the deaf can hear and the blind can see. - Mark Twain
I think oggs sound great but i am still ripping mp3s at 192 bits or better because they also sound great and everything i have is geared towards running them, (WinAmp, SoundBlaster Live, Creative Nomad Jukebox, nothing flashy) I think that ogg has what it takes to supplant mp3s in the future (better sounding compression and smaller filesize) and all that it lacks is maturity.
This sounds similar to a previous /. story. Although the tests were apparently run with a variety of people in the musical arena, the tests weren't run blindly (apaprently the panel knew if they were listening to an mp3 or an oog file.)
But, it's still worth a read, imho.
--You will rephrase your request for me to go to hell. Goto statements are not acceptable programming constructs
I like mp3 a lot more than ogg. I have an album or 2 ripped with ogg as well as some randoms songs from compilations. I did them around 200kbps VBR and my mp3s are 192 kpbs CBR. I'm listening on cambridge soundworks 4.1 surround speakers on an MX300.
I found the ogg files really tinny and light, so I'd stick with mp3.
Question for the masses:
Doesn't the quality of the speakers, the noise on the wires, the interference from the monitor and the size of the bass cabinet etc etc etc have a more pertinent effect on sound quality when you get above a certain sample rate.
128 is better than 64, sure, but above that isn;t the difference between monitor mounted speakers and a dolby 5.1 creative surround sound system, say, the most important one?
I dont know - I'm asking you...
Personally I think using r3mix on LAME mp3 encorder makes the mp3 sound exactly like you are listening to the cd. And if you rip the cd with EAC, you have a perfect copy. I never really liked VBR before but it is actually starting to prove itself to be worthy. Check out http://www.r3mix.net for more info.
I actually just did a pretty vigorous test of this the other day. I tested 128, 160, 192, and 256 bitrate mp3s and oggs against the source wav file. At 128 they both sounded similar, but the ogg file did seem a little brighter and clearer than the mp3, and the wav file of course blew them both away. At 160 ogg vorbis really shines... the mp3 remains kind of dull, muddy, and the high end is very "sizzly" compared to the ogg file which sounds brilliant and clear. I barely noticed a difference between the wav file and the ogg at this bitrate. Going up to 192 I found the difference between the ogg and the wav indistinguishable while the mp3 STILL retained some of that annoying high-end sizzle and midrange mud. If you've got the space... 192 oggs amazing... I'm doing mine at 160 because while disc space is cheap, the difference between 160 and 192 is negligible. As for 256... don't bother doing oggs at this level... it's just a waste of disk space. As far as mp3s go... IMO you'd have to encode them at 256 to get the same fidelity as a 160 bitrate ogg vorbis file.(your milage may very... i have been an audio engineer for a while and have picky picky ears.)
Now, if only I could flash my Rio into decoding these files i'd be in digital audio heaven! Also... I'm cannot wait for the 1.0 Ogg encoder to come out... encoding times should be much faster and fidelity even better. Amazing work!
Hope this helps.
-auttie
--->auttie
MPEG is lossy compression. Period. Even if you encode at 320Kbps, you are still losing data. You lose a lot less data at 320Kbps than you do at 128Kbps, but you still lose things.
128Kbps is NOT CD quality. A CD is a 44KHz, 16 bit PCM data stream, uncompressed. It's usually decoded using a 1 bit DAC, IE, via pulse width modulation.
Nothing is "CD Quality" except uncompressed audio or audio compressed with lossless compression like ZIP, RAR, ACE, gzip, bzip, et cetera. "Multimedia" compression is without exception lossy compression - Even our beloved DivX ;-) MPEG-4 High Speed compression is lossy, it's just less lossy than most.
"You're right," Fisheye says. "I should have set it on 'whip' or 'chop.'"
Do me a favour, everyone:
1) Rip your CD to 128kb mp3s.
2) Re-burn it to CD. (use a rewrite if you're a cheapskate)
3) Listen to the two side-by-side.
Big Fscking Difference!
192 is the best bang for your searching efforts, because any higher takes up too much bloody space. But that encode I can burn to a custom CD and it'll sound fine.
Now, back to Ogg... Ogg sounds about the same at 160kb as an mp3 at 190, (debate and argue all you want...) which is why I like its compression system. Still, I wouldn't touch a 128kb ogg either
I don't know much about ogg, as I use mp3 for most of my music encoding. I've played around with various bit rates and finaly settled on what I felt was the best for me in terms of quality vs size.
I now encode all of my music at a variable bit rate 64-256kbps with lame. Lame 3.70 does a really good job of this and produces files (at least for the types of music I listen to) that sound very good. For the most part, they encode smaller than a 192kbps, as the average bit rate used is less. As a check, peeking at John Coletrane's Giant Steps, the average bit rate is right around 150. The bulk of my music averages between 160 and 192kbps.
The cool thing about vbr is that if the file needs more than that, is can use up to 256kbps to help make the harder to encode spots sound better. So I guess the worst case size you could get would be a song completely encoded at 256kbps (but I can't say that has ever happened).
I have a hard time telling these vbr 64-256kbps files apart from the orignal cd. Sometimes I can tell, but it is rare and difficult. However, IANAA (I am not an audiophile), so doing your own tests should help.
All of your standard tools should support vbr files. Xmms does a fine job. I did need to upgrade mpg123 to pre0.59s, however.
Anyway, consider vbr before you go straight to 300kbps.
This sig is false.
I just read this article about Croteam using it for their next game:
"We did a major change in the sound engine between FE and SE. And its name is Ogg Vorbis. Yeah, that's right, we're using ogg for music playing. In case someone hasn't heard of it yet, Ogg Vorbis (http://www.vorbis.com) is a patent-free, open source audio codec project. Or in english: a music compressor that plainly rocks. Make sure you check it out. We've tried encoding all the music for SE with Oggdrop at 64kbps and the quality was perfect even at such low bitrate. In the final version, since we won't need the extra space, we'll be shipping with 128kbps music tracks, for even higher fidelity. The guys there are really helpful and supportive and the whole project is surprisingly functional already. There are plugins for all major music players and other music programs."
Ant(Dude) @ Quality Foraged Links (AQFL.net) & The Ant Farm (antfarm.ma.cx / antfarm.home.dhs.org).
But if you do care about the actual sound, rip some tracks you like from different types of albums. Then, cut out one part of the .WAV file and encode it using different MP3 encoders and different bitrates. (Or, if you want to save time, use only LAME for MP3, because there's a near-consensus that it gets the best sound. Don't forget to try VBR.) Then encode it in OGG format, also at various bitrates.
Now, the important step:
Decode the OGGs/MP3s back to a .WAV file, and make sure you name your files so you know which is which. Now, ask your roommate to burn all these .WAV files on a CD in an order that will not be revealed to you. Also burn the WAV that never went through compression/decompression (see if you can identify it by sound). Now, get your best pair of headphones, go to your stereo with a pad of paper, play the tracks over and over, and take notes on which track sounds the best.
Only after you've decided which tracks sound the best can you ask your roommate which tracks were encoded with which method.
This is not hard to do, and absolutely necessary if you want anyone to take your opinion about encoder quality seriously.
spork
The cake is a pie
Comment removed based on user account deletion
Of course, with a drive that size, you could go all-out and use Monkey's Audio, lossless audio compression (you can decode to get *exactly* the same WAV file that was encoded. Compression ratio of only 2:1 or so, but again...what's the 100 GB drive for?!! Get on Google and search around for some comparisons, and make an educated choice.
What I mean by this is, are you trying to be as true to the original recording as possible, or do you just want decent sound? If the former, you're trying to approach hardcore high-end audio and you don't want ogg or mp3. If the latter, then just go by what your ears tell you -- from everything I've experienced, the two formats are virtually indistinguishable on a standard speaker setup.
Second, you're playing said file from a computer or some kind of mp3 player. How good are your speakers/headphones? Do they have the range, presence, crispness, etc. that you want? How good is your player's line out and D/A converter? How noisy is your sound card? Hell, how much RF interference does your computer produce or induce in the sound card? If you want to be really anal, what kind of cables are you using to run to the speakers (or stereo)?
Ultimately, since you know that you're going with something that's not going to be totally true to the original, you just have to go with what you think sounds good. You have to remember, not all ears are created equal. Go by what's good for you.
Having said all that (and at the risk of contradicting myself), with -specific- songs I've noticed a difference between encoding at 128k and, say, 192k. This is especially true when listening with quality headphones. Classical music in general or music like Orbital in specific seem to sound better to me at 192k. After 192, I personally can't tell a difference. Your mileage may vary. I've listened to two identical classical pieces, one compressed at 128k and one at 192k, over a friend's hifi stereo and there was a difference in hearable elements and sense of presence. Over my lofi stereo there's no discernable difference.
So, of course make sure you take this with however much salt you desire. It all comes down to what sounds good to you, and what kind of sound setup you're using. As the question was stated, it's difficult to give an accurate answer -- and of course, even a "correct" answer may not necessarily apply to you.
Including this one.
- Jonathan
I sent them an email asking for ogg support and they said if there was enough interest they would implement it.
course. someone'll have to mod this up first. ;).
I just saw a nice quote today: "audiophiles are people who listen to the audio equipment, not the music"
You get almost the same quality at half the size with WMA
Sadly, you can't. Listening tests have shown that WMA 8 has sacrified sound quality at medium/high bitrates over WMA 7 to improve quality at the low end. So it's great for music over a modem, but at 96k and above it is no better than Ogg Vorbis.
-- Help Digitise the Public Domain at DP.
No, it's not, not if you're playing music originally from CD. CD's are stereo. Not 4-channel, not 5.1. Do you expect your surround system to magically figure out what speaker to send a signal to?
With that said, four-speaker stereo can significantly increase the size of your room's "sweet spot" and reduce the stereo distortion effect you hear when turning your head. Add a subwoofer for deep bass response, and that's about the most you'll need for accurate playback of any two-channel source.
I have done a >REAL3000$ stereo equipment (Van den Hul
cables, atacama stands, gold plated connectors
etc) to play 2 tracks in :
a) vorbis, 192
b) mp3, notlame, high quality VBR, stereo, 128-320, 195 kbps average
c) original wav file
The tracks were ripped from a superb quality
classical recording (I play the piano), from
DECCA.
I then had 3 of my friends compare the track
quality "blindly".
The difference between vorbis and mp3 is
immediately noticeable. Vorbis was found superior
by all the listeners. Some people had difficulty
telling vorbis from wav but they generally
tended to prefer the wav. (each one was
questioned individually)
Personally I find the difference quite striking
and was truly amazed!
This was an important finding for me, because
I make amateur recordings at home and I need
an easy means of archival (we are talking many
GB here, and I don't intend to fill my HD).
I decided to use vorbis at 350 for all my
archived recordings. (I also keep
cds).
I cannot say whether vorbis is also superior
in lower bit rates such as 128kbps.
Petros
What difference does it make if your receiver does Dolby Digital? Your MP3s aren't an AC3 source. Receivers with "all the bells and whistles" are often of LOWER quality than those dedicated to doing one task well. Dolby Digital is for movies with earth-shattering-kabooms.
Are there really people here that think a "16 bit" sound card can't reproduce full CD audio? How do you think they play WAV files?
It's amazing the number of completely irrelevant factors people are bringing up here. Is there a word for the phenomenon that occurs when someone shells out money for something and then feels the need to factor its presence into anything remotely related to it?
It's also amazing that nobody is bringing up some REAL issues:
The quality of your connectors is more important than that of your sound card. Bring the audio to your receiver over SPDIF or TOSLINK, not over analog RCA cables! Sound cards --- ALL of them --- have really awful RCA connectors.
Even SPDIF and TOSLINK aren't lossless --- but the conveyance of waveform audio in your computer to your audio peripherals is. Since the inside of your computer has lots of interferance (hard drives, power supplies), it logically makes sense to deliver your audio as far away from your computer as possible before converting it to send to your receiver.
So USB audio makes a *lot* of sense for setups that simply want to do faithful MP3 playback --- a cheap Roland UA-30 will do SPDIF, TOSLINK, powers itself off the bus, and can sit yards away from your computer.
I don't understand the original question or some of the responses regarding bit rates. I encoded my entire CD collection at 192kbs MP3. I'm not an audiophile by ANY means (and I don't want to be: I'd rather not TRAIN myself not to like my sound system!!!) --- but I *regret* doing this; guitar and (real) drum driven music sounds awful in a good car stereo (Pioneer+JL+DynAudio) at 192, and tolerable at 256.
Even 2 years ago disk space was cheap enough to make 256 the reasonable choice. But when you can get a 75G stackable firewire drive/enclosure for less than $200, what possible incentive could you have for encoding at less than 256?
I can't tell the difference between 256 and anything above. VBR improves sound quality when you set a floor of 256 and a ceiling of infinity; otherwise, it's just a silly hack to save disk space at the expense of your MP3 files. It may not noticeably damage audio quality, but it sure as hell makes your MP3 files more complex, harder to analyze and play with/sort/etc. MP3 is just a poor file format for what VBR asks it to do.
Another big gotcha with MP3 is joint-stereo, the "reasonable default" in many encoders. Joint stereo is another psychoacoustic hack that saves an inconsequential amount of disk space at the expense of noticeable degradation in sound quality. It "spoofs" stereo for frequency ranges that its model believes is hard to localize in human ears. Make sure you nail your encoder at real stereo.
The most painful gotcha of all, fortunately, is one that most people have managed to avoid, and that is that codec quality is a HUGE factor. My original batch of 600 CDs was done with bladeenc (mass groan!); bladeenc is/was completely broken. People aren't kidding when they say that Fraunhofer sounds better than random other encoders. Fortunately LAME is a great choice.
As for Ogg: it's great that we have an open source codec. This will come in very handy for streaming audio delivery and for the cores of sound engines in games or other random programs. Because of this it's also great that Ogg is (apparently) more efficient than MP3. One hopes it will continue to become more and more efficient so it can give Microsoft's compromised but extremely efficient format a run for its money.
But since disk space isn't an issue, if you don't trust MP3 (putting you squarely in the minority), I'd say use Shorten or some other lossless format before making the irrevocable decision to put all your music into young Ogg Vorbis. It takes a *long time* to re-encode all of your CDs (*sob*).
Remember this: your time is far more valuable than disk drive space. Don't encode your music to the weak sound system you may have now: encode it to the ideal, even if you can't exploit it now, so that you'll be able to listen to your music without wasting time re-encoding it later on.
I have no idea where you got the idea that 128/44 is standard CD quality. I'm not even sure what 128/44 means.
Let's figure out what the bitrate of CD-quality audio is:
1. 44100 Hz (i.e. 44 kHz)
2. Two channels
3. 16 bits per sample
44100*2*16 = 1411200 bits per second, or 1411 kbps. That's the bitrate of CD audio.
Note that these are bits, not bytes. A CD takes up 1411/8 = 176 kB per second.
So the fact that an MP3 sounds pretty good at 192 kbps (which is 24 kB per second - the capital B for Bytes instead of bits) is actually quite impressive. It's compressing by about a factor of 7.
Luckily, most rippers don't even give you a choice. They just rip the raw bytes and stick a WAV header on each track. Good rippers verify that they're reading the CD correctly, of course, but they don't do any compression or re-encoding.
I avoided WMA for years, since I was afraid of all the horrible things people were saying about it. I finally tried it, and at least to my ears, a 96K WMA sounds as good as a 160K MP3. OGG is about a wash vs. MP3, and it's not supported nearly as well, which gives me about zero reason to use it. I don't spend much time in Linux, which is pretty much the only area where OGG is better supported than WMA.
Which listening tests are you referring to? I'm pretty darned picky, and I can hear a difference. 96K is pretty bad on OGG and MP3, and very good on WMA, at least for rock music.
And I would urge everyone to do their own listening tests - I took the pepsi challenge, and WMA won hands down.
+5:offtopic,but anti-American
I haven't directly compared OGG and mp3, mostly because I'm very happy with the quality of the mp3 encoding.
In my own testing, the r3mix.net settings were pretty much indistinguishable from the original in terms of frequency response. I did notice some changes in spatial effects. One of my CDs in particular was affected, Deepforest 2. With the original CD playing, the sound tended to bounce all around your head when wearing headphones. After being encoded by LAME, the sound still moved some, but it was much more granular. Most of the effect was lost. However, the actual FREQUENCY RESPONSE was awesome, and the only way I could really tell the difference was by listening very intensely. It is more than adequate for normal listening.
I did these tests about a year and a half ago, on LAME 3.81, and apparently it has improved quite a bit since. That team respects the r3mix site enough that they actually added in an '--r3mix' command line switch to implement all of their suggested settings at once. Apparently LAME now keeps more of the original signal; it's not quite so enthusiastic about assuming you can't hear certain kinds of noise. I'm hopeful this may have fixed the encoding issues I had with the earlier version.
Basically, given the fact that he has tons of space available, and given that there's all sorts of portable MP3 players in the world, I think he may still be happiest with MP3. I certainly am.
Equipment used: Non-golden ears, but decent ones. Soundblaster Live Platinum 5.1 (which has some frequency response issues with REAL audiophiles), Sennheiser HD 580 headphones for 'real' listening, Midiland S2 4100s (the older 2 speaker model) for casual music and gaming.
Aside: The 580s are AWESOME headphones, and you can often get them very cheap at auction. I got mine about two years ago for about $125. They have a reputation of having flaky connections. Mine did indeed have a problem when I first got them, which I solved simply by removing and replugging the wire in the bottom of the headphone. They are fully modular, easy to disassemble and clean, and sound INCREDIBLE. Two downsides: they really need an amplified headphone jack to reach their true potential, and they are big headphones. They're very comfortable but large.
Aside on the early model Midilands: great quality speakers, dismal amp. Hissy at any volume. Someday I'll move the way-cool little satellites onto a real amplifier, and will toss the subwoofer/amp in the trash.
JPEG is a bad example here. Our most developed sense is eyesight; The eye is a very complex piece of equipment, and we have more brain dedicated to eyesight than any other sense.
Also, a lot can happen to sound before it reaches your ear. A lot less happens to light (especially at close range.)
With that said, I can definitely tell the difference between a JPEG and the original uncompressed image, even at fairly high quality settings.
The idea behing JPEG's loss being acceptable is that photographic-type images, the kind JPEG is intended to be used for, are already grainy, due to the nature of the universe, which is also grainy. Therefore the grainyness (is that a word?) of JPEG does not cause a problem, ostensibly. In reality, you can't control HOW JPEG makes things grainy, so you may lose detail you were counting on to get a high-quality image out.
The audio information to which you are referring is known as "psychoacoustic" audio information. While you cannot actually hear the frequencies which MP3 is supposed to be dropping, those frequencies when combined with other frequencies, the resonance of your eardrum and associated mechanisms, and so on, become audible. Sometimes it's only perceptible as a slight pressure on the eardrum, but it changes the way all other sounds are perceived at the same time. This is what the vinylcentric audiophiles are talking about when they try to explain why they prefer vinyl over a CD. When you play a very good piece of vinyl on a very good turntable, using a very good needle, going into a very good analog amplifier, and using very good speakers, headphones, or whatever, there is definitely a difference between vinyl and a compact disc.
As you say, whether or not this difference is important is entirely up to the individual listener. But MP3 does not in fact only lose frequencies that are ostensibly not important to you, as you seem to believe; It creates QUITE perceptible differences, especially with heavy bass, as I have previously mentioned. Even a person with partial hearing loss should be able to detect the difference between the original CD source and a 128Kbps MP3 in most cases, again, especially with regards to heavy bass.
This is true. If 128 (or lower!) Kbps bitrate mp3s are suitable for you, then go on with your badself. Me, I discard mp3s with a lower-than-192Kbps bitrate, unless it's some exceptionally rare material, or it's something where the quality doesn't matter so much, like plain speech.
"You're right," Fisheye says. "I should have set it on 'whip' or 'chop.'"
I ripped the Playstation Descent soundtrack to .wav, and proceeded to encode it to mp3. Problem was, there was one track with a particular instrumental arrangement that my normal 160K MP3 (LAME) just mangled. I tried various mp3 codecs, all the way up to the max of 320Kbps, and couldn't get it to sound correct. Then I tried Ogg Vorbis just for fun. Even 96K Ogg reproduces it correctly.
;)
Not exactly a scientific comparison, but a valuable example none the less. I've found that mp3's biggest problem is that it will mangle certain patterns in certain songs. Chances are, if you picked a random song out of my 1000+ playlist, it would sound reasonbly good at 128, or even 112 or 96. But there's a few in there, just a handful, that require 160 to sound ok, and a few (as above) that even 320 can't save. Try encoding Metallica's (heh, irony) "Until It Sleeps" at 128 or lower. When the main riff kicks in, you should be moved to vomit by how awful it sounds. Try again at 160 and it should be ok. If you can't hear it, consider yourself VERY lucky
BTW, for more in depth discussion that has been ongoing, have a look at the forums at r3mix.net and the Ogg-specific forums at Hydrogen Audio. I keep up with both forums, and the folks there tend to make prerelease build binaries available for people to play with. For up-to-date detailed information without the overhead of the Vorbis-dev list, those are the places to go.
One more link for folks who want to know more: The beginning of the document describing Vorbis stereo discusses good terminology and qualification of subjective fidelity. It's nothing new to most posters I expect, but it might help keep the discussion consistent.
Happy hacking,
Monty
xiph.org
i became a fan of .ogg this summer, just because i thought it sounded better on my altec lansings. so when i came to school this fall, i couldn't resist challenging my audiophile next door neighbor/old roomate/good friend to test it.
.ogg, he made a 256k .mp3 with whatever encoder it is he prefers, and then we both decoded them back to .wav, and made a 3-track cd (the 3rd track being the song uncompressed).
.ogg.
i'd just gotten a wynton marsalis cd from amazon, so _carnival of venice_ was used as the testing track. i made a 256k
we did a blind test, kinda. put the cd in his player and set it on random. it was obvious that one track was better than the others (cd) and one was a lot worse than the others (mp3). the ogg sounded remarkably like the cd track, though there were some small things that allowed us to differentiate.
i'm not sure i'd be able to do so well on the same test using my computer speakers, of course. but the difference is certainly there.
test stereo setup:
CD Player: NAD 512
Integrated Amplifier: NAD 314
Speakers: Acoustic Energy Aesprit 300
Interconnects: Kimber Kable PBJ
Speaker Cables: Kimber Kable 4VS
of course, there are problems in the test in that we only tested one track, so the findings are only representative for the wynton marsalis genre. but it made me a fan of
i encourage everyone to try something similar and draw your own conclusions.
I'm the guy who wrote up a 'sonogram encoder study' using a pathologically impossible waveform to encode, and then measuring how much different mp3 encoders fell apart, and in what ways. Like r3mix.net, I wound up supporting LAME, but with some explanations for what people find compelling about Blade and Fraunhofer, respectively.
You also should know that people have been pestering me to add Ogg comparisons for _ages_, even wanting to send me the files I couldn't encode myself on an OS 8.1 Mac.
Well, there have been some changes at Airwindows:
And so, _yesterday_, I set about getting a preliminary look at Ogg Vorbis using sonogram analysis on my Encoder Hell test sound- put in half a day on it, and updated my site to include the new information. And today, guess what turns up on Slashdot? Spooky.
Now, I need to emphasise that the process wasn't exactly the same as last time- I had to include some 'control' sonograms using the same mp3s that I used last time (Frau 128 and Blade 320, strong but idiosyncratic performers of known characteristics) for comparison. It's preliminary, and I don't want to immediately go into a complete shootout again because (a) it's such an undertaking and (b) I'm not at all sure I'm using a current Ogg version here. That said...
Here is the result of this early look at Ogg Vorbis, and I think I managed to sort of exactly what Ogg is relative to mp3. Quotes from the final report:
That is, to my mind, a pretty strong endorsement, requiring only that high bit rates be used (as is intended) As such, I think Ogg will only become more relevant as bandwidth and storage space inevitably expand. It also is, in my professional opinion, very well positioned to keep mp3 in check- mp3 can only maintain its dominance by not getting carried away with licensing and IP abuses, because Ogg is sonically superior enough to be able to take over _if_ given the opportunity of a situation involving harsh mp3 licensing, given widespread use of higher bit rates rather than low ones. (This is why I dismiss WMA- it belongs to yesterday, an era of limited storage space and harsh licensing restrictions)
Now, about iTunes? I have some observations that I'd love to learn more about. Basically, I picked up iTunes because there's a patch making it possible to install on system 8.6, and I did that- only to be startled by a distinct difference in sound quality which I have the background to interpret. Briefly, it sounds like iTunes dithers its mp3 output to 16 bit, instead of truncating it.
A bit of background: any decoder, either mp3 or Ogg or whatever, is effectively synthesising a waveform from limited information. It's adding harmonics together to produce a linear PCM representation that's piped to the sound output hardware.
I suspect everyone making mp3 players has been simply truncating the waveform to 16 bit on the assumption that it's low quality anyway and doesn't matter... until iTunes... which has startlingly better dimensionality and depth than any other player I've heard.
However- there's no patent on the general concept of dithering. Some of the fancier ditherers and noise shaping algorithms are proprietary, but I happen to know many that are actually GPLed...
It's exciting to see the pieces of a truly superior free audio technology come together...
SHN, perhaps?
TO BUY A NEW CAR WOULD MAKE YOU SEXUALLY ATTRACTIVE.
I like MP3. The "EM" is a nice hard sound to start with, and transitions nicely into the rhyming "pee" and "three" to lead into the next word.
"Ogg" just makes me feel like I'm choking on a donut.
:-)
sig fault
Sadly, this often isn't valid for quite a few reasons.
Quite often, encoders will use very different procedures to encode a low bitrates than they would use to encode at high bitrates. They will probably use a hearing model (which models the ATH - absolute threshold of hearing) which is less demanding, for example. They may even automatically low pass the music, or resample it to a lower bitrate.
For example, Ogg Vorbis has different methods of channel coupling. At very high bitrates, no stereo information is lost. At medium bitrates, no stereo information is lost for the sounds we are most sensitive to, but for others the phasing is quantisised. The degree of compression determines the range of lossless coupling, and also the amount of quantisisation -- and each may have its own distinctive artifacts.
MP3 can't encode stereo 44kHz (CD quality) sound at 48kp/s without sounding truely terrible. If you try with LAME, you will find that it automatically resamples, and uses one of the 'extensions' to the official MP3 specification which encode better at low bitrates by resampling the sound.
-- Help Digitise the Public Domain at DP.
Ok, we got many things (using lame style names):
CBR = Constant bit rate = Variable quality
VBR = Variable bit rate = Constant quality
ABR = Target bit rate = Variable but not as much quality
OGG normally uses a form of ABR, but is capable to do true CBR and true VBR as well (not sure which versions enabled for).
Also, even if you are using true CBR, there is little room for flexibility in the form of the "bit reservoir"; you can save some bits in the "easy parts" so they can be better spent in the hard parts.
Second, mp3, being open in some way or another, has the side effect of many encoders available. Different encoders produce different quality. Take 4 192kbps mp3s encoded with 4 different encoders, and you will discover quality differences as day to night.
And to use Lame properly, first, let me suggest that you *at least* use Lame 3.89b. Lame 3.70 is *too old*. If you get Lame 3.90a, even better.
Want to be on the safe side? use this single option:
lame --dm-preset standard
This will produce near 256kbps files, and its the hightest quality you can get out of mp3s.
If you think you can live with 192kbps like files, then use
lame --r3mix
Otherwise stick to the normal, don't apply options you don't know much of. Typically you *always* want -h, and -b for the desired bitrate in case of CBR, or minimun frame bitrate for audio in the case of VBR (usually 112 or 128). ABR is VBR attempting an average bitrate. And no, it is not wise to use option -B at all (let the encoder use up to 320kbps frames when using VBR).
If this topic of lossy compression is of interest for you, then you should visit:
Proyect Mayhem, channel #Project_Mayhem at irc.openprojects.org
and
r3mix.net, channel #r3mix at irc.openprojects.org
Um... on side note, have you seen The Wavelet Tutorial yet? Wavelets are planned for Ogg Vorbis 2.x, stay tuned... :)
Artix
Your Linux, your init.
Anybody have a Dennon Test CD or digital equivilant? Anyone have a distortion anlyzer? Osciliscope? Spectrum display? Take a CD of some of the sine wave tracks (direct digital mastered) and encode them into the various formats. Check the results. I am interested in THD, S/N ratio, Jitter, and ailising frequencies. Anybody up to this and posting repeatable test results? Lets find out what the artifacts are on a 20 HZ bass signal as well a 440 HZ and 3 KHZ. I have part of the test equipment needed to perform the tests. My amp is rated at 0.005% THD which is below the capibilities of my test equipment to measure it.
The truth shall set you free!