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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
They sound about the same, really. What encoder you use has a lot more to do with the end sound quality then the format.
ReadThe ReflectionEngine, a cyberpunk style n
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...
LOL good one!
If you were an audiophile, you wouldn't be listening to compressed music at all..
even though i also use 192kbps on my mixed song, i found that 128kbps on mp3 is pretty good enough for everyday listening.
check out http://ff123.net/cbr128.html
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.
all my files are ripped to ogg 192 kbits. the sound is excellent, and it takes about 60-65 MB per album, depending on number of songs and song length, etc. i listen on some fairly nice stereo headphones at work, and a medium-range stereo at home with a nice 'whoofer', and the sound is great.
-sam
The REAL sam_at_caveman_dot_org is user ID 13833.
128 kbps MP3 isn't CD quality at all. You can easily hear the difference. Use some classical music, which is generally harder to compress, rip and encode it and compare for yourself. If you use headphones it's even easier.
I have no idea why people call 128 kbps CD quality. You can never get to CD quality with MP3, since no matter how many kbps, the encoding process still removes data.
Score:-1, Wrong
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
ugh,
first, I doubt that a cd is 128kbps, it's more like 150kBps (notice the _big_ b, *Byte*).
The measured data rate (150 kB/s, 128kb/s) is always about the "streaming" rate, i.e. the packed mp3/ogg stream, which makes this difference clear (otherwise we wouldn't really compress, would we?).
Perhaps you confused it with 44,1kHz, which is a cd fixum and indeed sets an upper limit on frequencies which can encoded on a audio cd?
The first time I heard ogg I could hear it clipping from time to time and I thought it sucked, but later on I downloaded a bunch of different encoders and did some tests to see how stuff came out and compare file sizes and honestly, the tests I did they sounded the same and the file sizes weren't significantly different.
What is the best Ogg encoder?
The man who trades freedom for security does not deserve nor will he ever receive either. - Benjamin Franklin
Unless you are an audio nut, you'll have trouble telling the difference at any high bitrates. The real reason to use Ogg is that it's not encumbered by patents :-)
Gerv
I would agree with the general sentiment that -- despite any quality difference -- MP3 is certainly going to be easier to use because portable players and software have been built to use it.
Remember that ultimately, the "best" any format is going to get will simply be as good as the original CD. So as long as the audio quality you're getting is indistinguishable from the original, it won't matter what format you're using.
That said, I think 160 is something you'll regret if you're doing a large number of songs. I originally used 160 on my 800 CDs, and it sounded fine -- until I hooked the digital out on my soundcard to my dolby digital 5.1 system. On good speakers with a clean connection you can definitely hear the compression artifacts. I went up to Xing VBR 192-320 and have been very pleased with the results. As you said, disk space is no longer an issue, so I'm comfortable using what i think were probably overkill settings.
These source files are good enough, BTW, to re-encode into WMA at 64k for use on my portable (a NEX II -- highly recommended). With a 256MB compactflash card, I have about 150 songs with me for running (this can use a microdrive too, but it skips when running). The WMA 64 quality is perfectly acceptable for cheap headphones, but it would be total crap on good speakers. This (and streaming) are the only places where files size really matters anymore...
Recursive: Adj. See Recursive.
I rip mp3 at 256k. Long ago when I started ripping with lame I compared 128, 192, and 256. The difference from 192 to 256 was noticable on my material. In particular, the stereo imaging was off much worse at 192 than 256. There is still a noticable difference from 256 to CD, but I can live with it.
I listen with a set of Boston Acoustics speakers with an external subwoofer. It has a tiny sweetspot, about head sized, but for a single listener they are quite good, especially the stereo imaging. (I also have a set of their less expensive model that I got at compusa, these are not so good. You want the ones with the bigger speakers.)
Although I listen mostly at my linux machines, I also use a Mac for portable work. As soon as there is an ogg plugin for itunes I will switch to ogg and re-encode all my CDs. I'll redo the bitrate selection exercise at that time.
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
Wellnow, there ARE a few issues here that you neglect to touch upon. The ARGUMENT has come up before about what is the best format, granted, but it has not really been asked as this specific question, with a purpose in mind.
1. As someone notes above, there is no commercially available Ogg Vorbis player - however, if you're solely ripping for computer use, either format is fine.
2. To correct everyone above, you RIP at 128/44, which is standard CD quality, but if you ENCODE at a higher bitrate, you get cleaner ultimate quality.
3. Most importantly, it all goes back to purpose. If you're just setting up a jukebox for the home, stick with MP3, stick with 128/44, it's more than fine for good quality (with a decent encoder) and if you aren't listening for artifcating it's easy enough to ignore.
"I'm not even supposed to BE here today!"
Well, I'd have to say that with my 16 bit sound card and my 1 speaker hooked up to it (that's right one, we've all done it, I don't have the cash flow at the moment to go and buy a Soundblaster 5.1 and 4 speakers and an amp from boston, if I did then I wouldn't put down that I had 1 speaker hooked up to the really old isa sound card that is 16 bit, now would I?) and I can listen to both pretty fine. I think the problem will be three fold. The compression rate, the hardware issue, and the software issue.
:)
1)The compression rate- By this, I do not just mean 64 kbs or 192 kbs, but also what you decoded it with. If you were to do it with software a versus software b, that software may compress it differently, causing tiny bits of saturation in the bass or the higher octanes. Of course, people of the art of music have been using isdn for the longest to do compilations together across great distances. I assume they would know the best way to encode.
2)The Hardware Issue- Do you have surround sound? That would be a major question. I mean, if you are worried about different kinds of files playing the same music, you probably would need surround sound to tell the difference. It's the honest truth. Someone said in an earlier comment that distortion from monitors, your server (they're not workstations, they're servers. Look at the stats, p3 1.7 ghz with 2 gis of ram... what else could it be?) the phone lines, electric cabling, anything. And everything. If you were really into this, you would make a sound room like in music halls. No distortion, sound proof walls, etc. They're pretty cool to have too.
3) The software- I mean this as an os and as the software you listen to as well. If you use real player, winamp, freeamp, would that sound better than other said software? What about the os? What services are bogging it down so that you cannot use those resources to power your music.
In conclusion, do what a friend of mine did. Make yourself a "napster box". Hook it up, you only need a 2 gig hard drive. Put on a 4 speed cd-rw, and you don't need anything above a k6-2 for processor. More ram the better though. (Of course). But put on every kind of sound hardware you can. Also only put on a 10 inch monochrome monitor, if any kind of monitor at all. Put it in a closet, and just administer it through the network, voila, sound system. Later
The cake is a pie
There are over a dozen lossless audio compression packages available. They all sound the same. I'll just note that FLAC is open source (GPL & LGPL), patent free, and has WinAmp and XMMS plugins available.
Doug Moen.
I have written a truly remarkable program which this sig is too small to contain.
I believe ogg sounds better at the same bitrate as a mp3.
.wavs to mp3s (lame@128, no VBR) and to oggs, both at 128kbit. I came back a day later and loaded up each group of songs (the mp3, the ogg and the wav) into xmms and winamp. I turned on shuffle and began switching back and forth between songs (so I wouldn't know which format I was listening too) with my eyes closed. Obviously the .wav sounded the best as it was an exact copy of the CD, then I found that the ogg sounded better than the mp3. Ogg seems to sound crisper and bring out the little details of a song much better.
I took some of my favorite CD's (ones that I've listened to over and over again and know very well) and did a little comparison. I ripped one track off each of the discs (usually my favorite track), encoded the resulting
For some fun, take your headphone or speaker connection (as long as its a barrel connector) and pull it about halfway out. Now, if you do it right you can hear some kind of fuzzy noise on mp3's, maybe encoding artifacts but it sounds like noise. The higher the bitrate, the less prominent it is. I'm imagining I can hear that when I'm listening normally but it's mixed correctly so it's very low-key. Anyways, it sounds strange and ogg doesnt seem to produce this "noise".
Just my OPINION,
Geoffeg
CD audio is 16 bits per channel, 44khz stereo. Uncompressed this is far, far more than 128kbps. 128kbps was/is the standard encoding rate because this is the bandwidth of an ISDN2 line, which was/is a commons means of transmission of audio, particularly for things like outside radio broadcasts.
Blaming GW Bush for the Iraq war is like blaming Ronald McDonald for the poor quality of food.
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
As much as I like ogg, MP3 is the standard format right now. If you, for instance, encode your files as mp3... and you later buy one of those cd-players that can read mp3 cd's, you won't have to re-encode. Ogg is a new format so support for it is not high, so if you have portable devices in mind, go for mp3.
GoatPigSheep, the 3 most important food groups
If ultimate best sound is the issue, then what you need is something lossless. That rules out MP3 and OGG. And WAV is a waste of space. But there is FLAC, which is lossless sound compression. I haven't tried it yet, as the sound quality is not yet the defining issue for me. But as soon as I get beyond these tiny speakers and this cheap sound card, and have a laboratory grade ultra-linear DAC doing my analog conversion, feeding speakers with more watts than my PC power supply knows about, then quality certainly will be an issue.
now we need to go OSS in diesel cars
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"
If you'll be listening on your home stereo, I'd recommend using MP3 at 192kbps, Ogg at around 160kbps, or better. But if you're ripping primarily for your car or portable, lower bitrates will probably be acceptable, since in both cases there'll be enough ambient noise to wash out the fine details anyway. Of course, if you use your portable in a quiet room, or listen to the stereo in your parked car with the windows closed, you might again consider higher bitrates.
The reason for encoding CDs into digital formats are size, archiving, convenience, portability.
Size - some say Ogg is better at smaller sizes, but it's debatable. Storage has never been cheaper and is getting cheaper still. Why would anyone encode at 128k anyway? MP3 with VBR and the right options is about the same size as 192k and without some very high end playback equipment is indistinguishable from CD.
Archiving - there is no real difference in quality if you know how to use Ogg & MP3 encoders properly. Archival encoding means you want to have it forever, so you're not going to be caring much about size differences between formats, minor as they are. Quality matters most here. Can you tell the highest quality encodings in both formats from CD? Day to day use, no. Again you need some very high-end gear to hear the differences, i.e. not your soundcard or your portable.
Convenience - both formats give you the ability to playback what you want without reaching over to the CD rack, just open the player. No difference there.
Portability - MP3 has it all. Ogg has virtually none. Come on, someone reply with a link to some tiny Korean company that promises to make an Ogg player Real Soon Now.
Why would you bother with Ogg? Maybe if you absolutely will not use something that anyone has a patent on, but if that's the case you're going to have a difficult life.
Dast: Yeah man VBR is excellent.
lame --r3mix -b112 source.wav out.mp3
That's a ready-made VBR setting that I (and many others) find remarkable. But lame has made an amazing number of possibilities for modifying how the mp3 is going to sound in subtle ways. Check out r3mix.net, and their forums.
Never understimate the power of the placebo-effect! I used to think all formats sounded like crap at 128kbit, but I guess I was fooling myself.
The other day I did a blind test comparing wav, mp3, mp3PRO, ogg, wma and acc. Since my speakers aren't all that good, and the acuostics of my room are less then perfect, I use a set of nice headphones connected to my EWS64XL soundcard. I tested with a few different songs, both classical and "modern". I converted the a wav to all the formats at 64, 96, 128 and 160kbit (except for mp3pro which I could only encode at 64 and 96kbit and acc at 64, 96 and 128), and then back again to wavs (so buffering delays won't reveal what I'm listening to). Then I made a playlist of them in winamp and randomized it. I put pieces of paper on my monitor so I could only see which number in the playlist I was listening to and then tried to guess what I was listening to.
My conclusion was that if a good encoder was used for MP3 (I used LAME at highest quality settings) I could tell that it was compressed about half of the times at 128kbit. At 96 and 64kbit it always sounded awful, and at 160kbit I could never tell it from the original.
I was really impressed with ogg. It has been tremendously improved with rc2! I could actually not tell which was wich at either 128 or 160 kbit, and about 50/50 at 96kbit! Ogg was also the format that took the crown at 64kbit. I would say 64kbit ogg is really enjoyablem, at least with less than perfect equipment. The default bitrate in the oggdrop encoder seems to be 80kbit. I guess that's a good choice.
WMA sounded better than mp3 at 64 and 96kbit, but I could actually tell more wma 128kbit's from the original than mp3s. I couldn't tell the original from 160kbit wmas though. The encoding scheme of wma seems to be quite different from the others. There seems to be less "compression-sounds" but it is as bad as some others at buchering the comes through. When few sounds are heard (a single violin for example) it sound really good at 96 (and quite nice at 64kbit too), but as soon as lots of sounds at a wide frequency range appears (such as big symphony orchestra chord), it sounds as if it is doing rough low-pass filtering or something. Really nasty.
MP3Pro sounded worse than wma with the violin but better with the orchestra, at both 64 and 96 kbit. I could always tell them from the original though.
ACC is as good as (possibly slightly better than) MP3 at 128kbit, about as good as mp3pro at 96kbit, but really bad at 64kbit. This could have been due to a bad encoder though. Sounded like it did lots of low-pass filtering.
Overall, I'd say ogg is the winner. I now encode all my music with ogg at 128kbit. I'm eagerly awating ogg 1.0!
Well, just my thoughts.
Regards / ushac
There is a very good comparison of lossless audio encoders here. The 'shorten' format has some problems, such as seeking when playing. I'd advise you to use the FLAC format. It has all the plugins that Shorten has, but has better compression, and inbuilt support for seeking and streaming.
-- Help Digitise the Public Domain at DP.
AFAIK, every piece of mp3 encoding software out there defaults to CBR. Most home users don't know what VBR is or why they would want to use it, so they just leave things at the defaults. Only the most technical users who take the time to learn about these things well check that VBR box or add that additional command line parameter.
This is why you find so many CBR files out there. You're not missing some wonderful CBR advantage.
[Side note: Whenever I encode, I make VBR .ogg files.]
unless you like your music to sound like it's synthesised by a computer.
:)
My music already IS that way, no encoder is ever going to make it sound natural
Use Flac, or shorten, or any lossless compression codec with (at least) source available.
This gives you approx 2:1 compression on clean CD rips, much better on quiet stuff and (of course) closer to 1:1 on noise. So we're talking about maybe 300Mb or so per album.
For $1 per album or less (prices always falling) you will kick yourself if you rip to 128Kbit MP3 now and then waste another week of your life re-ripping the CDs (if they haven't died) at higher quality later in your life.
Think you're never going to want higher quality? If you're at 192Kbit/s or lower think again, the artifacts in MP3 and Vorbis get more obvious the more you listen to compressed sound. Originals sound eerily "more lifelike" and eventually you will go back to your CDs "just to check" and find that the difference was there after all.
Why use Flac rather than just leaving WAVs on the disk? Well for one thing, disk is cheap but it isn't free. I save up to 50% just by running some free software == bargain.
Also, please for the love of all that is good use paranoid / seamless CD ripping. If I hear one more person playing their "CD quality" rips with obvious jumps in them I will scream.
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.
thanks to both of you -- when I began my jukebox projects, I didn't know about any lossless compressors, and admittedly disk space was a bit more of an issue a year ago (when 40 gig drives were the decent bang/buck)
Recursive: Adj. See Recursive.
128 kbps is nowhere *near* CD quality. It is simply 'acceptable' quality for casual listening for most people. ie: It's not bad enough to make them say 'This sucks'. SO they don't notice.
There is a definite, easy to hear diffrence if you compare them though.
I can't detect any real difference between a bladeenc-encoded mp3 at 192 and a Vorbis file encoded with VBR.
So I use Vorbis, because it's unrestricted and my distribution of choice ships with the tools.
I don't think space is really an issue these days. With today's 100Gb drives, you can fit literally thousands of MP3s on, and the exact bitrate doesn't really matter. I'm in half a mind to encode everything with lossless compressor like FLAC (which average about 15Mb per song) and be done with the quality debate for good.
First thing's first. I listen through a Yamaha SW1000XG sound card, a mid-range Phonic mixer, and a decent pair of Technics headphones (no, they're not stunning; I'm about to order a decent Sennheiser pair). I do a bit of sound engineering here and there, and have had much better things to listen to.
I can tell the difference between some MP3s encoded at 256kbps, and at 320kbps. Personally, my MP3s are LAME VBR encoded, with a maximum bandwidth of 320kbps, although it rarely reaches that.
I've tried ogg before, which is probably what's stopping me from trying it again. The version I tried quite substantially chewed up the treble. It's probably got better now, but I don't see a vast amount of advantage in it.
If you really want to do reliable tests on wav files, then visit PCABX to get the PCABX program and to read more about the testing methodology. The program takes in two wav files, and then chooses one of the two randomly and lets the user decide which of the two is the one chosen randomly. Basically, once this done a good number of times (say, 20) the program can then tell whether the user can actually tell the difference between the two files.
Also, a wonderful website dedicated to the task of creating archival quality encoded audio (which is indistinguishable from the original) is r3mix. Lame even has an optimized parameter that comes from the work at the site, --r3mix! This VBR parameter gives incredible quality at a fairly low bitrate. Check out too a listening test carried out at r3mix that showed the blind preferences of 42 users over a month of time.
The riovolt has upgradable codecs, and is (imo) the best cd-mp3 player out there even despite that fact. Go to rio, http://www.rio.com and send them some e-mail begging for an ogg upgrade. I have one of the first generation mp3-cd players, but as soon as the Riovolt supports ogg, I'm buying one, and re-ripping my cds into ogg.
but I encode all my albums into Mp3s using VBR at "100%" quality
:) (although it will be able to be *more* constant than currently in RC3).
This doesn't make any sense -- "100% quality"? What encoder are you using?
does Ogg have variable bit rate capability?
Yes in a big way -- Ogg doesn't really have *constant* bit rate capability
I think we've decided that "size doesn't matter".
Size still does matter. With FLAC, I'll get a lossless file about 60% of the original. With Ogg, I can get a file that sounds almost perfect about 15% of the size of the original. This means I can fit 4 more albums into the same size -- useful for storing all 400 albums on the hard disc so I can random play them.
-- Help Digitise the Public Domain at DP.
No, go with FLAC (or at least 'Shorten').
Monkey's Audio is Windows only, and it is a very stupid idea to archive data into a Windows-only format.
-- Help Digitise the Public Domain at DP.
Red Book Audio is 150 kiloBYTES per second. MP3's are measured in BIT rates.
I don't know if it's just me, but I'm reading the forum and seeing people say they are using 80-250 Megs per album on high bitrate lousy formats. There are several lossless audio compression projects out there that are getting pretty decent compression rates.
Of course the result is never going to be near as small as MPG or OGG, but it does get rid of all the tweaking disscusions (i.e. Which VBR/CBR, CODECs, Bitrates, is best) that seem to be big time wasters.
I look at it like this. HD Space is cheap these days, 60 Gig drives are starting to dip below $100. A pair of 60 gig drives could store 300+ Lossless Albums.
I've heard a lot of people whine about how they wish that OGG would be supported on their Rio or Nomad or whatever. As somewhat of an insider in the portable MP3 player industry, I can say that the people who code the player applications for these devices wish that they could get their hands on a fixed-point algorithm for decoding Vorbis. If someone were to write a proof-of-concept library or application and put it up on Freshmeat or Sourceforge, I'd personally insure that it gets in the hands of the right people.
Why fixed point? All of the portable, mobile, and stereo component MP3 players are based around microprocessors that don't have any hardware floating point coprocessors. Since software FP is too sluggish, an efficient way of doing the Vorbis decode with integer operations alone is necessary.
If anyone is interested, don't hesistate to email me at the address above. No promises, but I might be able to get some development hardware for whoever is interested...
you can even get a streaming shorten plugin for xmms.
in the tape trading network (such as the grateful dead tapers), shn is the preferred format.
while it initially wasn't designed for realtime playback (it was meant for batch compression and file transfer transmission, like ftp), its now considered a streamable format as well.
--
"It is now safe to switch off your computer."
These questions really gets down to how the music is going to be used. I have two ways I listen to music - at home, and on the road.
First a little background on where I'm coming from. I listen to music through Paradigm Active 20 studio monitors, which are professional studio speakers with internal amplifiers, or through medium/high end Sony MDR-V900 headphones. I'm also a video professional, so I've very attuned to quality (I don't watch movies on VHS, for example, as the poor quality is too distracting). I'm not quite so fussy about audio, but am still pretty fussy.
At home, I have all my music on an old PowerMac G4 400 I had lying around, which I also use to rip with. I use Maxtor 75 GB external FireWire drives, which are pretty much infinitely daisy-chainable to expand storage, so the only real issue in data rate is balancing quality per cost .
On the road, I listen to music via iTunes on my PowerBook G4. Quality is less important than storage effeciency, since I have a limited amount of space I want to spend on my hard drive for audio (2GB is my budget - I need a lot of room for video files). Also, I'm pretty much only listening the the music while I'm writing. I've found a 128 Kbps with an average data rate around 155 to be good enough that I'm not actively distracted by poor audio quality, although I can hear artifacts if I pay attention. However, I continually add and remove audio from my local storage
I did a bunch of encode tests, and spent quite a while figuring out the best way to go. I found audio sounded "good enough" for high end listening at 192 Kbps MP3. However, given the amount of labor of encoding all my CD's (34 days worth of music so far), I really, really wanted to make sure I wouldn't EVER have to go back to the original discs again. I assume I'm going to be recompressing from these files to new audio codecs for at least a decade to come, so I want the quality to be not only transparent for listening, but not to have a minimum of sub-audible compression artifacts that would make later recompression more difficult. Because of this, I encoded everything at 320 Normal (not joint) stereo, with no filtering. 256 might have worked, but it was worth spending a little more on storage in order to not have to worry about having to rip all those CD's again. Even assuming your time is only worth minimum wage, it's way cheaper to buy more storage and spend less time fussing. Still, it's a little irritating to know all those bonus tracks with 10 minutes of silence in them are still eating up 40 K per second.
For my laptop music, I encode at 128 max VBR, joint stereo, with a 10 Hz filter. These files sound just fine. I reencode all of these from my master array of music as needed. In the future, I'm sure I'll migrate to other audio codecs for this as the technology improves, allowing me to get more music on the laptop, the car stereo, or whatever I wind up doing with the stuff.
-Ben Waggoner
My video compression blog
Whoever mod'd this down to Offtopic doesnt know what they're doing. This is NOT offtopic.
EAC and LAME are the best way to rip and encode mp3s (respectively)
EAC: http://www.ping.be/satcp/eac00.htm Use secure mode when ripping, its slower but you wont get pops, clicks and bleeps.
LAME: http://www.hot.ee/smpman/mp3/
At the moment you should be using at least the LAME 3.89 executable to encode. If you are, use the following command line (EAC -> Compression Options -> External Compression -> Additional command line options):
--r3mix
Yes, that's all, "--r3mix", nothing else. For more information on this, visit http://www.r3mix.net
Remove the Spam to email me.
Taco, 192, 160, none of that. Go to r3mix.net to learn more. Using lame --r3mix, you get quality virtually indistinguishable from the CD, at a size usually lower than 192kbps. 360 is always an option, but the quality of this will equal that. Be careful though, not all VBR encoders are created equal. There's a reason they picked Lame. There are also programs like Exact Audio Copy for Windows (and some Linux equivilent though the name eludes me at the moment) that will double-check your audio extraction to confirm its correctness. Yes, this means I only rip at 1x or so on my slow cd-rom drive and duron 650 box, but it's killer quality. I output straight from my sound blaster live to the stereo on my desk, so I should know (well, it isn't a great stereo, but better than most any computer setup).
'mon, dudes, anything plays VBR these days (even my crappy kenwood in-dash car player). Am I missing some wonderful CBR advantage here?!?!
I've had a friend tell me that you can't stream VBR with icecast or shoutcast or something. He said it gets all choppy and sounds like total crap. He had to rerip his entire collection (and 5GB I gave him) to CBR so he could stream it.
Intelligent Life on Earth
Since the Ogg compression sounded awful...
.wav to check that it had read the track right. The sound was very plastic, it felt like a cheap radio playing inside a box. It wasn't even a question about any snobbish high-end audiophile 'take away information and it sounds like crap' kind of thing. It just sounded plain wrong...
Decided that it was about time I checked out the Ogg Vorbis encoder since, well, many people have said many good things about it.
I downloaded the vorbis tools, decided to get a CD that displays different charactistics and that I know well, and settled for the Delerium Karma album.
I started RipperX, choosed to encode the first track using the highest setting (320k), High Quality mode, no CRC, no VBR and ran the encoding.
The result, well, to tell you the truth, I had to listen to the
Equipment used was my Denon AH-D750 headphones (decent quality, not studio reference quality though) driven by a NAD 3020i amp (leftover from when I got new stereo equipment) fed through my SB Live.
//Humming
I'm too stupid to preview.
Go to http://www.r3mix.net and read the articles and follow the links. Great info!!
In february 2000 c't magazin organised a blind listening test. 300 Audiophiles were involved, finalists tested 17 1-min clips from different artists (classic and pop):
original CD recording
128 Kbit/s Joint Stereo [MusicMatch (FhG) v4.4] encoded PC decoded Mac
256 Kbit/s Joint Stereo [MusicMatch (FhG) v4.4] encoded PC decoded Mac
Basicly it says, under most circumstances, 256kbit is indistinguishable from original CD (not the same, just that humans can rarely hear the difference).
But since size is a consideration, you can get close to the same quality using VBR with the latest LAME.
Hey says forget Xing, use either Lame or Fraunhofer for CBR, for VBR Lame is the only decent choice.
With my files --r3mix tends to average around 170kbit, some as high as 214kbit, some as low as 130kbit.
Go check it out, in particular the quality page:
http://users.belgacom.net/gc247244/quality.htm
Now if only someone sold a CD changer with a PC interface so I wouldn't have to rip my CD collection one CD at a time....
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.
If you search the vorbis/vorbis-dev mailing list archives, you'll find some more info about these.
Don't neglect to make the distinction between the format and the encoder. Both formats are well enough documented that writing a player is a trivial (in the mathematical sense) exercise, but writing encoders is an exercise left to the reader.
Right now, this only really applies to MP3s, but when vorbis becomes more widely accepted I'm sure we'll see commercial MP3 encoders ported over to the new format.
Stupid question, but how much do the vorbis encoder and the lame encoder have in common?
my sig's at the bottom of the page.
reread what I wrote, learn something about physics, and come back...
Hint, try to figure out how the 44.1 kHz sampling rate might work with a 80kHz sinus wave. A paper and a pen works. Then try to grasp why I wrote "sets an upper limit".
And don't call others silly on the basis of an electronic marketing brochure, sheeesh.
I have found it valuable, in the world of MP3 at least, to pick and choose your bit rates according to the content you're encoding. For example, a majestic piece of classical music on CD released within the last five years should be encoded with the highest bit rate you can manage, for the simple fact that you are going to be able to hear technological deficiencies more easily. For less "well-defined" music (i.e. techno mixes, heavy guitar rock and the like), 128 or 160 is going to suffice because you are going to have more difficulty picking up on the "bad parts."
As a personal example, I tested various bitrates from 56 to 320 on a [digitally remastered] Miles Davis CD and the higher the bit rate the better it sounded. However, the same experiment on Metallica Master of Puppets resulted in little to no improvement (audible to me anyway) over 160 kbps.
It is always going to come down to HOW you listen to which kinds of music. When I'm "banging my head," I'm less likely to hear a tiny millisecond pop. When I'm floating along with something more subtle (jazz and classical in particular), if I lose definition in the higher range I'm going to be distracted.
I find the same to be true when I'm watching television on the ole dish. I hardly notice MPEG artifacting when I'm engrossed in a "high-octane thriller" [ouch], but if I'm watching a long dramatic dialogue I will see every digital flaw.
There is no right answer when you're attempting to compress and digitize entertainment. Your mileage will always vary.
Aaron
P.S.- It should be noted that most consumer-grade speakers top out at 22 kHz in terms of their high-range frequency capability, so you're already losing out on detail in your music, particularly in the high-end formats like HDCD, DVD and SACD.
seriously folks, most people listen to cds on equipment less than perfect, in conditions less than perfect (cars, diskmans, one of those refurbished flashing light disco tower things from circuit city). in this case the difference between burned and original cds is probably a lot less than the difference between the different things they get played on. i actually prefer lower sample rates for some things, in cars for instance it makes the music punchier over the white noise of road, engine, etc.
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
Now, as many people have said, ogg is better than mp3. And, yes, this is true. Im a professional musician, and I do use digital sampling to do my work. But since, as most musicians, Im poor, hard drive space is at a premium. I find that oggs sound _much_ better than mp3s, and on average sound better than mp3pro mp3s at the same bitrate.
I usually use this formula to encode stuff in different formats to compare them.
ogg 160kbps ~= mp3 256kbps ~= mp3pro 192kbps ~= wma 192kbps (though, I have yet to find a 192kbps wma encoder.)
Though, you should wait until 1.0 final comes out before you do any major archiving of music, since the most recent release canidate of 1.0 might still have bugs, though I am seriously doubting it.
Also, if you do wish to encode mp3s and oggs, I suggest you use lame (http://lame.sourceforge.net/) since it is considered the best mp3 encoder, and the 3.8x and later version can infact use the ogg/vorbis libraries to encode oggs. Plus, its gpl.
Patrick "Diablo-D3" McFarland || http://AdTerrasPerAspera.com
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.
Joint stereo means the encoder can switch between stereo (encoding the left and right channels independently) and mid/side stereo (taking advantage of the similarities between two channels - they don't have to be exactly the same, just similar). If the encoder switches between stereo and m/s stereo at the proper times, joint stereo should sound the best. It has a bad reputation because certain encoders (I think Xing) don't know when to switch to stereo, and it's very noticible if you're using headphones. Since headphones have perfect stereo separation, it will sound almost like the signal is going from stereo to mono randomly. But LAME and FHG both do a good job of joint stereo.
I have not tried Ogg - I'm not sure if it has the same sort of variable bitrate options, joint stereo, high-quality special voodoo etc. I'm happy with Lame.
Not yet. Vorbis is always VBR, but you can't do any advanced tuning. It will have options for this in the future. Vorbis has channel-coupling, similar to joint stereo, but some people say it produces a high-frequency hissing sound. It's a fairly new option (introduced in RC2), so it still needs some tuning.
I've been using LAME 3.89 for awhile, with the following command line:
lame -r -o -k -v -V0 -x -b128 -mj -h -q1
After much figuring and careful listening, I've found this to work well enough that I never hear the difference between the encoded file and its original wav. Average bitrate works out to 190-220kbps, depending on source material, which seems quite acceptable in terms of disk use.
Notice the lack of an upper limit on VBR bitrate on the command line; if you're going through all the trouble of doing VBR, why bother with arbitrary limits? LAME's VBR makes very minimal use of the bit resevoir - the same feature which allows it to work as well as it does on CBR encodings. Don't set limits. It's mathematically shown that you'll hear the difference, and you won't care notice the slightly increased filesize of having a few 320kbps frames - especially on a 100GB drive.
Equipment is as follows:
Recent ALSA
CMI-based sound card from Zoltrix
Audio Alchemy DDE 1.1, fed with TOSLINK from the CMI card
Rotel RTC-940AX preamp
Ashly FTX-1001 power amp
Midrange Sony headphones (which I don't have in front of me)
Speakers of my own build (parts from Vifa and Madisound, simple crossovers, -good- cabinets)
The magic here on the hardware end is the CMI card, which doesn't resample its digital output to 48KHz, like most other consumer devices (SBLive, Yamaha XG come to mind) - thus, it is bit-perfect. Being samplerate-locked causes a bit of trouble on non-44.1 files, as ALSA doesn't have good software resampling, but it's a CD player replacement - not a gaming machine - and for that, it works perfectly.
Kid-proof tablet..
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 don't use Ogg or mp3 for my stuff.. I use shn (from etree.org) instead, which is a lossless audio compression method that cuts the ripped wav file roughly in half. I listen with Sennheiser 495 headphones plugged into a headphone amp. Ogg and mp3 simply aren't good enough.
You just said "Bose" and "reproduce sounds *very* well" in the same paragraph. Sorry if I'm chortling right now...
Read the Bose Faq, and please be more careful with your future purchases.
>Deep bass tends to get crunchy VERY fast,
Sorry guy, that's your speakers talking. MP3 gets 'watery' when its compressed too much. Your drivers are likely made of plain paper. Read reviews of the product before you moderate. Some high, many, many, many rock-bottom low. And really, plain paper? My clock radio uses the same technology.
Really, I'm not trying to be a jerk, but before you spend mega-$$$ on anything again, look it up on the 'net. You just might be doing yourself a favour.
If you could be told what you can see or read, then it follows that you could be told what to say or think - BoC
Wasnt there a similar thread? Well, not one that directly adressed this, but talked about the improvements in RC2, and resulted in the usual quality comparison arguments?
IMPO, its too early to tell about OGG. Its not 100% complete, and im not going to take the time to convert my collection of a couple 1000 songs (all 192k+ mp3 / mp+) just to do a quality comparison on an format that isnt complete.
Wait until the full thing is out, then repeat the question.
in the words of the great Steve Albini.
Picky listening? For my money no digital format is all that good, not even CD let alone a lossy format like MP3 or Ogg. The warmth, the dynamics of vinyl records can't be beaten, especially if played through vacuum tube equipment.
May be one day when the sampling rate is high enough digital will approach analog quality. Audio DVDs have promise. Unfortunately the music industry oligarchs are not supporting it.
But if for convenience or out of necessity you compress, Ogg beats MP3.
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
I wonder why always people forget about main CD idea: audio CD doesn't have error correction. Developers of audio CD standard, back in 80's, though that nobody will notice loss of 100 harmonics out of 200 in 'regular' sound.
That's why there are so fscking expencive CD transports - they try to correct unavoidable loss of quality of CD's.
So, actually, whith cdparanoia, ogg vorbis and USB audio with external DAC you can get better quality than from CD. Especially, if you're listening from those cheap damn computer cdplayers.
Ok, the idea again: cd looses harmonics because of never-ideal reads, and ogg vorbis carefully removes unhearable parts of music.
eme
No. They have a SPECIFIC choice.
They have neither MORE, nor BETTER choices.
Chas - The one, the only.
THANK GOD!!!
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
Shit, if you are getting a 100 Gig drive, why not just screw the lossy compression and just save the wave files? 100 gigs should hold 150 cds in wave format.
.... and while you're at it, why not increase your browser cache to 10GB so you can keep every web page you visit back a whole year??
The Answer: Because you don't NEED to store information you'll never use (or will never hear). That's the whole point of compression.
Also, it should be pointed out that 150 CDs is a fairly small collection, especially when singles are taken into account on top of that.
----- rL
The post got me curious as I'd never tried ogg before so I downloaded the plugin for winamp and set about comparing the two formats. I pulled 30 seconds of a song (Electronic's Prodigal Son) to a wav and then encoded that wav into eight files, four files for each format. One for both mp3 and ogg in 64, 128, 192, and 256. I then added all nine files (eight encodes plus wav) to a play list and listened to them at random for a while. To be honest, I really couldn't tell the difference between anything in the 256 range. The ogg, the mp3, and the wav all sounded nearly identical. At 192, I could tell the difference (but not by much) between the encoded files and the wav, but not between each other. At the lower ranges, ogg sounded better, probably due to it's variable bit rate. My guess is however, that no one's considering encoding in 64 these days anyway.
As for what I'm running the sound through, I can't help but think it's all irrelvent. I mean, a good sounding mp3 will sound better than a crappy mp3 regardless of the system. You may be able to achieve higher quality sound via elaborate setups on high end consumer electronics, but the coded file hasn't changed. Seems to me like we were being asked which format was better, not what was the most important factor in determing sound quality. Any true audiophile probably wouldn't even play music on a computer unless he or she had to. I mean come on, do you really think a 100 dollar sound blaster card can do as much as a 2,000 dollar harmon kardon receiver?
In short, senor Taco, my expirience seems to be that if you're looking for good quality take the bastards up to 256 and use what you feel like, there's no difference I can distinguish. If you're looking to save a little space but maintain a decent level of quality, I'd go with ogg.
Abandon All Hope, Ye Who Enter Here
CDs are 1411kbps. 128kbps just _can't_ be CD quality.
Yeah, I don't get how people can claim 128kbps is CD quality either. If MP3s weren't a lot smaller than the original source then there would be little point in their use as they made storage and transmission more convenient. It is widely known that MP3 is lossy anyways. Put those together and MP3 is going to be lesser than a CD at any of its available bitrates.
> although not perfect
You got that right - the test was only done at 128 kbit/s! Yikes.
I think I've heard of lossless compressed-audio codecs, but I can't recall any names off the top of my head. For video editing, WAV has been sufficient as the few hundred megs needed for audio is nothing compared to the tens of gigabytes needed for video. As for MP3, I've been using 160-kbps VBR lately with LAME for CD rips. I can't tell the difference, but then I don't claim to have "golden ears" either. Tape rips get 128-kbps VBR as there's already been a fair amount of loss introduced when the tape was produced.
20 January 2017: the End of an Error.
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.
Random access.
"You're right," Fisheye says. "I should have set it on 'whip' or 'chop.'"
I like Ogg Vorbis. I guess I never tried doing VBR MP3s, but I really hated how MP3s would consistently screw up `sss' noises (both from voices and from things like cymbal/hi-hat). I've been able to hear problems with Ogg files too, but they have deficiencies in much less common sounds.
Also, MP3s use coding methods that aren't optimal.. even if an MP3 file is encoded at an infinite bitrate, there can be compression artifacts.
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.
"You techies disgust me!"
You modern whippersnappers disgust me! Whenever I want to listen to music, I have a band come to my house.
Secrecy corrupts democracy: What should be the Response to Violence?
Bush's education improvements were
Then I'm doomed.
I never wanted to consider myself an audiophile, and yet I'm still concerned with how well my equipment reproduces the sound (At least whenever I am going to purchase something). I also listen to the 'equipment' in the form of the various compression codecs to determine which one sounds closer to the original before I go compressing the music I enjoy...
Can I at least say I'm not an audiophile because I'm not always doing this?!?
Please!?!
-- Sometimes you have to turn the lights off in order to see.
I've spent too much time looking at various video compression schemes over the years, and find that I now have a tendency to look at the artifacts rather than the images...
Wait til they get legislation passed preventing the manufacturers from creating units that don't include "copyright protection technology."
It's not enough to bash in heads, you've got to bash in minds. - Captain Hammer
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
If you encode at a high enough bitrate, you can't effectively tell the difference from CD, even if you're an arrogant audiophile who thinks he can. 128kbps is virtually indistinguishable from CD for some (provided it's a good rip and many aren't). For others it may be higher but the point of lossy compression is creating the same effective perception of sound as the original sample.
N4st0r, trixx0r h0bb1tz0rz! Th3y st0l3 0ur pr3c10uzz!
When I decided to rip my music collection, I decided on flac, an awesome lossless codec than averages around 50% compression.
Lossless compression obviously sounds (literally) perfect now, but makes more of a difference for the long-term health of your digital music collection. If you had ripped everything to mp3 last year, and then this year decided to convert your mp3s to Ogg, then your music would have gone through two generations of lossy encoding (or you would have gone through the effort of ripping everything all over again). Not good, especially if a more desirable codec is going to come along next year... Besides, your hard drive size is going to grow exponentially, and in a few years you're going to feel pretty silly having a compressed, shittier-than-original-fidelity music collection that takes up a miniscule percentage of your hard drive.
If you want to stream your music and don't have a spare T1 to saturate, then you can always convert to mp3/Ogg/Codec-du-jour on the fly (if you're not serving tons of streams), but the important thing is that a lossless encoding of the original bits sits on your hard drive.
Actually most online radio stations (the big ones anyway) compress on the fly from their audio feed.
so the encoding is less of an issue (as long as they can encode in realtime with a not insanely grunty machine).
'There is a Light that never goes out.'
I disagree, you should do these tests on the best audio system you can find, because the files you will be producing are archival. You can always upgrade your playback equipment, but once encoded you can never upgrade this source material.
20 January 2017: the End of an Error.
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.
64k *is* low bitrate (you can't get that low with MP3 without either resampling the input or having the output sound truly terrible). The lowest that beta4 would go was 112k.
To go lower with Vorbis you'll have to resample the input (it's not currently tuned for sample rates other than 44.1/48KHz, but it will work). In Linux this is easily done with something like (from memory)
sox input.wav -r 22050 -t wav - | oggenc -b 64 -o output.ogg -
The '-b 64' specifies the bitrate you would get if you were giving it CD quality (44KHz) input. In reality, it just specifies a set of noise masking, channel coupling, and low passing switches... you'll probably get around 40k with it.
-- Help Digitise the Public Domain at DP.
How can I use it if it has no source? If I can't integrate it into my existing programming, it won't do me any good. I run all my music through my own sound server.
now we need to go OSS in diesel cars
The flac site has a feature-oriented comparison of various lossless codecs. The Monkey's Audio site has a performance-oriented comparison (they compare an older version of flac, unfortunately).
For those who are listening to your mp3s over your hifi system, consider investing in an external DAC and put it between your soundcard's digital out and your amplifier.
The reason is that the soundcard is a very bad position for digital-to-analogue conversion to take place, due to the presence of all sorts of interference in the computer casing. Furthermore, the electronics on the soundcard are usually not good enough to properly drive your line-out RCA cables (or worse, stereo jack out).
Audiophiles will claim that putting an external DAC between your CD player and amplifier makes a world of difference. From my own experience, putting it between your soundcard and amplifier gives a real improvement even to non-audiophiles.
Cheap but good entry-level DACs can be gotten from $200-$300. (Check out Cambridge Audio, AMC etc.) Its shelf life is at least 8-10+ years, and so IMHO is a very worthy investment, especially if you listen to a lot of mp3s over your hifi system.
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.
Maybe it's a sign my ear/audio gear isn't first rate, but I have no hope of distinguishing a 256 Kbps LAME mp3 from the uncompressed WAV. With my big stereo speakers, I often can't even hear the artefacts on a 192 Kbps mp3 when I use my big speakers, even when they're up loud.
I find this totally acceptable, and will not replace LAME as my primary encoder unless I get indistinguishability at a lower bitrate.
Notice that I'm not saying any 256 Kbps mp3 will sound just like the original--only those encoded on a recent LAME beta. If you can reliably distinguish a 256Lame from the original in a double-blind test, I'll be impressed. Please describe the song and your gear.
spork
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!
Well
Explenation: A cognitive dissonance is a state where there is a gap between your "beliefs" (cognition) and behaviour. Cognitive dissonance is a very unhealthy state to stay in, so people usually "resolve" cognitive dissonance by either altering their beliefs or behaviour. Quite often, the former is easier to alter.
So, in the case of buying an extremely expensive HiFi equipment, if you do not believe the sound quality worth the investment, you will be in cognitive dissonance, and hence to resolve it you convince yourself there is nothing remotely like the 10k USD tube-based amplifier you just bought
In other words, Cognitive dissonance is not the process of changing your beliefs, but the reason for doing it.
I had the advantage of living near an excellent audio store here in Seattle, of which I availed myself before buying my audio gear. They have an excellent page on exactly how you should structure your listening tests: you should listen to the tune.
(The rest of this comment is a small rant; feel free to ignore it.)
I ended up buying my integrated amplifier from NAD (which has the added benefit of a humorous name), CD player from Marantz, and speakers from Axiom. (You can't imagine how good these inexpensive speakers are.) I aimed for decent sound in my office without spending too much money. I'm not an audiophile, but I also know a particularly obvious imperfection will bother me. The system is pretty well-balanced for the price.
After replacing my crapomatic integrated bookshelf stereo in the office with decent components, the issues of MP3/OGG/etc. became irrelevent. They're all crap, and your only options are separating less stinky crap from really stinky crap, if you're listening to generated waveforms coming from your internal DAC. My SGIs, which spank any PC with an internal sound card for fidelity and noise level, don't even match up.
I really want to try one of the USB-connected external DACs, because I like the ability to manage my entire CD catalog from one place, without having to switch CDs. However, I couldn't continue kidding myself that the sound from the computers was the least bit faithful.
Havers, matey. OGG is capable of up to 256 channels, each encoded seperately.
The problem is this is that the goal is not the best accuracy to original waveform, but how people perceive the output. While similarity to the original waveform can be analogous to a point, some encoding decisions may deviate more from the original waveform than the other algorithm at points in order to take advantage of some human perceptual trick to make it sound closer to the original.
In this case, a waveform that seems mathematically more similar to the original could theoretically sound much crappier. When testing perceptual encoding techniques, the only way to go is to have very nicely structured blind listening tests with a great variety of people, equipment, and music.
XML is like violence. If it doesn't solve the problem, use more.
The AC I'm replying to is the only onw who pointed this out.
Please don't ask "which track sounds the best". Provide the track numbers of the original WAV files, and ask "which track sounds closest to the original".
If you mix the original WAV file in with your double-blind comparison, you will be picking which one you like, not which one is most accurate. What if your personal preference is for more bass than the original was mixed with? If one of the encoders is muddy, you might pick that over the most accurate.
To do it right, you'd have to double-blind all the various encodings and listen to them back to back with the original WAV file and see which one matches it best. And I wouldn't take the extra step of going back to WAV, as you'd be introducing another encoding layer that may introduce different artifacts.
Nope, no sig
Best of all, when you work out the cost per CD, SHNs only use about 40 cents of storage for an average length CD, so it might be a bit more than you spend with mp3, but it means that a $27,000 CD collection would fit on about $600 of hard drive, and that ratio is only getting better.
cute... but I disagree =)
... content. It's not perfect, but it does what I need for now. I have a friend with a $50k setup... drool... We watched "The Rock" at his house a few times, that was damned sweet.
I spent a great deal of time in highschool as the head of our Sound & Light crew... Anything in the school that needed sound, or lighting, was my department.
I spent MANY an hour in front of 24-channel massive audio mixer boards (things like this and this for big productions), and we had many concerts that were hosted through these beasts as well. In short, I became very very very sensitive to any sort of distortion in music. It was quite humerous, I would be sitting at home listening to some music and I would picture, in my head, the mixer board and say "ok, gotta adjust this knob here, this one here, and this one"... It was BAD.
So thus was born the audiophile.
At a $1400 stereo later I am
In short, no, I do listen to the music. But I listen far far too closely.
As for my MP3/OGG choice? Well, I don't care about disk space. I am fond of lame at ~192kbit/s VBR encoding (range 64-320) which does an excellent job, but even up to 320kbit/s I can usually notice some distortions on my stereo system. All formats are lossy. I played around with ogg a while back but due to lack of players I decided to stick with MP3.
If you want high quality, play it off of the CD that you own, otherwise you'll have to settle for less.
Also, it depends a lot on the type of music you're listening to. A lot of dance/trance/techno can do perfectly fine compressed without a lot of loss, however listen to anything with acoustic instruments and even the CD standard itself isn't good enough to convey it without noticable loss.
Moral: Try not to use anything better than what you currently own. Never use anything better than what you can afford, because then you will be tempted to figure out how you can afford it.
If God gave us curiosity
A more accurate one: "audiophiles are people who listen to the price of the audio equipment". ;-)
Grab.