What Sounds Better, MP3 or Ogg? 660
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 (Score:4, Insightful)
Re:Ogg (Score:4, Funny)
Re:Ogg (Score:2)
If they can get a few more features added/bugs stomped, and cleanup the UI a bit, it'll be near-perfect...
Re:Ogg (Score:4, Insightful)
Fair enough, but consider that the bill is floating around Washington, DC, USA. Emphasis on USA. All your bill are not belong to us. Some of the finest electronic components right now aren't the ones available to the US market. The highest quality, the coolest features, etc. aren't what sells in the the largest, dumbed-down, mass-consumption markets. High end Sony ES home audio gear, the coolest DVD features, region-free players, etc. for example are regularly imported from the UK and Japan by people who want/need that type of gear. Perhaps your bill will prevent you from importing non-DRMS devices though. Where there's a will (and a market elsewhere), there's a way.
Re:Ogg (Score:2)
My thoughts exactly. I'm as generally as happy with OGG at 128 or 160 as I am with MP3 at 192, but then I wouldn't be able to use my music in a car-based MP3 player...
Re:Ogg (Score:5, Insightful)
Bah. You want to see ogg in commercial players? Use it then dammit and stop using mp3. Stop whining about lack of commercial support; it's a kind of Catch-22 see? If no one uses ogg because it isn't popular then of course it won't get commercial support. It's gonna take an initial sacrifice (so grow a spine and give up your precious mp3) so that ogg can become popular. Only then will we all reap the benifits (ubiquitous Ogg Vorbis).
Also, read this fascinating interview [binaryfreedom.com] from early this year with Jack Moffitt and Christopher Montgomery, the two head guys behind Xiph and ogg. They discuss many things including the Iomega HipZip, which does support Ogg Vorbis.
lobby intel for ogg support in their firmware (Score:4, Informative)
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. ;).
Re:lobby intel for ogg support in their firmware (Score:4, Interesting)
I emailed an mp3 radio station I once listened to and asked them to offer a vorbis stream. They responded and stated that they've looked into it (woohoo) because of mp3 licensing, but it would require a lot of time and work to switch over. They implied that they intended to switch over eventually however. A deluge of polite requests for vorbis streams would surely speed things up.
On a slightly different note, I recently purchased some hardware for which open Linux drivers were available. So I emailed the company and told them that the availability of free / open drivers was the deciding factor of my choice of their product over a competitor's (it was). We need to do everything we can to encourage and reward good behavior in hardware manufacturers. They do listen as evidenced by the parent.
Re:Ogg (Score:2, Informative)
Re:Ogg (Score:3, Informative)
And yes, the quality:bitrate ratio in ogg kicks mp3's ass.
Riovolt (Score:2)
Re:Ogg (Score:5, Insightful)
I just don't understand the objection to MP3... it's a decent format, well worth the $2/unit royalty for the decoder chips. Maybe MPEG doesn't compress as well as Ogg, but I would consider this an even trade for the less expensive decoding.
Re:Ogg (Score:4, Informative)
Read all about it at http://www.xiph.org/about.html [xiph.org].
Daniel
Re:Ogg (Score:3, Informative)
Open formats are critical to open information exchange, this is exactly why there is such a fight to keep patents out of the w3c standars.
They sound the same (Score:2, Informative)
I agree with this guy (Score:2)
What is the best Ogg encoder?
Re:And WMA is about 2X better (Score:3, Informative)
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.
Which listening tests? (Score:3, Interesting)
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.
vorbis needs a little while to market penitrate (Score:3, Interesting)
Re:vorbis needs a little while to market penitrate (Score:2)
I am reminded... (Score:4, Informative)
This sounds similar to a previous /. story. [slashdot.org] 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.
Re:I am reminded... (Score:5, Informative)
Here is a test [ff123.net] that, although not perfect, was at least semi-blind. The conclusions: at that bitrate, MPC ('MPegPlus' not 'MP3Pro') and AAC were the best, followed by LAME MP3, OGG & WMA8 all together, and finally the very worst was XING encoded MP3.
This test was run with Vorbis RC2. RC3 will be out in a week, with much improved noise masking. For a taste of RC3, you can check out the Vorbis CVS, which includes most of the RC3 fixes but encodes at a fixed rate of 128 kp/ps. This raises the low pass, improves the noise masking, and the stereo channel coupling code.
mp3 (Score:4, Informative)
I found the ogg files really tinny and light, so I'd stick with mp3.
An engineering challange (Score:3, Interesting)
bitrate the least of the trouble at that level? (Score:4, Insightful)
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...
Re:bitrate the least of the trouble at that level? (Score:4, Informative)
Yes and no, but mostly no. I have a pair of Bose 601 Series II speakers attached to my stereo. They reproduce sounds *very* well. The stereo in question is a Sony STR-DE635 reciever with dolby digital, et cetera. It has 80 watts x 5 channels of discrete amplification. While there are stereo systems with a higher signal to noise ratio, this is a pretty damn good setup. I play mp3s from either my dreamcast (analog output) or my new Apex AD-3201 DVD player, which has a truly crappy interface for playing mp3s, but sounds okay, and plays VBRE without any trouble. It's got a coaxial digital connection to my stereo, and it spits mp3s out at it at 44.1KHz, 16 bit PCM (after decoding).
With all this said, I can definitely hear the inconsistencies in lower-bitrate (like 128Kbps) mp3s. The only encoding rates I'll use any more are mid-high VBRE (which will go up to 290Kbps or so) and 320Kbps for archival purposes of very touchy music, like classical pieces. If I have something which is purely spoken word, and it doesn't involve screaming, I'll sometimes drop down to 64Kbps mono just to make the files smaller, but generally I encode them as VBRE along with everything else.
The thing you really tend to lose a lot of in 128Kbps mp3s is bass. Deep bass tends to get crunchy VERY fast, even at slightly higher bitrates like 192Kbps. You can actually hear that even on computer speakers (I use a microsoft digital sound system in analog mode only) or on your car stereo (I burn mp3s back to CD fairly frequently) but especially on a high-end stereo, which will more faithfully reproduce the sounds its given. So actually, on a higher-end stereo, you will hear every bad frequency caused as a compression artifact.
I have no idea what low-bitrate audio sounds like on true 'prosumer' level home theater sounds like, but I bet it's really atrocious. It's bad enough on my only somewhat upscale layout.
Re:bitrate the least of the trouble at that level? (Score:2, Funny)
/sarcasm
Re:bitrate the least of the trouble at that level? (Score:2)
I ended up ebay'ing a sony ES receiver and it was significantly better (though it was around $450 or so, which kind of blew my budget). Love it though - but the two-way remote is an ergonomic atrocity.
Re:bitrate the least of the trouble at that level? (Score:2)
Re:bitrate the least of the trouble at that level? (Score:4, Interesting)
The S/N ratio of most soundcards is farily poor, especially compared to decent receivers.
Speaker wire and cabling makes a difference. No one agrees on how much, though
Speaker placement & seating position greatly impacts sound quality (e.g. stereo imaging).
Speakers play an obviously important role.
Finally, most people just arent' aware of, or don't care about, the actual sonic quality of stereo equipment. We just want something that sounds good to us.
Consider a previous poster, who loves his Bose speakers and Sony DE receiver. Home audio enthusiasts generally agree that the Sony DE line is definitely inferior to the higher DB and ES series. Likewise, Bose is generally known to be of lower sonic quality than other equivalently priced speakers. But he loves the setup, and that's the thing that matters.
Similarly, I've got a solid mid-range receiver (Onkyo 696) matched to low-end speakers (KLH bookshelf). Sounds great to me, but it's not really top-quality sound.
The encoder quality is just one of many possible limiters in sound reproduction. And if you're playing it through low-fi equpiment (e.g. computer soundcard and computer speakers), then just pick one that sounds decent and run with it.
MP3 at 128 is fine if you use good encoder (Score:2, Informative)
check out http://ff123.net/cbr128.html [ff123.net]
VBR (Score:4, Informative)
ogg, 192 kbits (Score:2, Redundant)
-sam
ogg vorbis all the way! (Score:5, Informative)
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
check www.r3mix.net (Score:4, Informative)
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.
Choose Ogg anyway... (Score:2)
Gerv
MP3 is more flexible (Score:2)
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...
Re:MP3 is more flexible (Score:2)
Re:flaco (Score:2)
Re:flaco (Score:2)
Monkey's Audio is Windows only, and it is a very stupid idea to archive data into a Windows-only format.
256k mp3, ogg soon (Score:2)
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.
Try VBR before you go to 300kbps (Score:5, Interesting)
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 [sulaco.org] 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 [xmms.org] does a fine job. I did need to upgrade mpg123 [mpg123.de] to pre0.59s, however.
Anyway, consider vbr before you go straight to 300kbps.
here's a sweet lame setting (Score:2, Informative)
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.
Re:Try VBR before you go to 300kbps (Score:2, Funny)
I have to agree that the variable bit rate sounds better than the constant. However, through my equipment, mp3s have a distinct lower quality than cds. There is a noticable improvement when they are burned to audio though, so it is not a lack in the mp3 format, just in my mp3 playing hardware. I have a very difficult time telling the difference between a song from a pressed cd and one that I have ripped, encoded and then burned back to cd audio format.
I have a Rotel [rotel.com] amp and pre. My source is a Harman Kardon [harmankardon.com] FL8550 [harmankardon.com] cd changer. My speakers are JBL [jbl.com] S38 [jbl.com] "bookshelf" speakers (they're bookshelf only if you have a BIG bookshelf.)
My soundblaster live value card can't compete with the FL8550's dual 20 bit Burr-Brown digital to analog converters. My next equipment purchase is going to be an Onkyo [onkyousa.com] SE-U55 [onkyomm.com] USB sound processor. Hopefully, that should let me use mp3s for more than casual listening.
One last thing: if you think computer addictions can be expensive, just try getting hooked on audiophile quality hardware! The interconnects I'm going to buy are over a $100 each for the bottom end of the line. But you can hear the difference.
lame --dm-preset standard (for ver. 3.90 or newer) (Score:3, Informative)
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 [hydrogenaudio.org], channel #Project_Mayhem at irc.openprojects.org
and
r3mix.net [r3mix.net], channel #r3mix at irc.openprojects.org
Um... on side note, have you seen The Wavelet Tutorial [rowan.edu] yet? Wavelets are planned for Ogg Vorbis 2.x, stay tuned... :)
Serious Sam sequel will use Ogg (Score:3, Informative)
"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 [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."
How to do listening tests (Score:5, Interesting)
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
Re:How to do listening tests (Score:5, Informative)
If you're running Windows, you can get ABX from http://www.pcabx.com/ [pcabx.com]. On UNIX systems, the LAME source code [mp3dev.org] comes with an ABX program (in the misc/ directory, I think).
Here is an example of a test that took place using a slightly different testing methodology, more akin to MUSHRA (which is used to evaluate lots of encoders at the same time): http://www.ff123.net/128tests.html [ff123.net].
Mp3.. Ogg? (Score:2, Insightful)
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
Re:Mp3.. Ogg? (Score:3, Informative)
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.
What about Wave? (Score:3, Informative)
lossless compression sounds better (Score:2, Informative)
There are over a dozen [firstpr.com.au] lossless audio compression packages available. They all sound the same. I'll just note that FLAC [sourceforge.net] is open source (GPL & LGPL), patent free, and has WinAmp and XMMS plugins available.
Doug Moen.
Ogg! (Score:2)
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
Comment removed (Score:3, Insightful)
MPEGplus (Score:4, Informative)
Of course, with a drive that size, you could go all-out and use Monkey's Audio [monkeysaudio.com], 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 [google.com] and search around for some comparisons, and make an educated choice.
Re:MPEGplus (Score:5, Informative)
MPC has better licensing than Monkey's Audio: the *decoder* is open source (GPL even), so you will always be able to decode your music. *encoding* is only possible on Windows however (although there is an older binary version available for i386 Linux systems), and the encoder will be made shareware in the near future. This is a real pity, because tests [ff123.net] have shown that even at 128kp/s MPC is up there with AAC (MPEG-4 audio).
Define your fidelity level (Score:5, Interesting)
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.
I would suggest MP3 (Score:2)
If ultimate best sound is the issue, then... (Score:2)
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 [sourceforge.net], 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.
What are you using them in? (Score:2)
Why would you bother with Ogg? (Score:2, Interesting)
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.
blind test (Score:2, Interesting)
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
Use Flac (Score:2)
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.
if *you* can't tell the difference... (Score:2)
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.
My Thoughts (Score:2)
First thing's first. I listen through a Yamaha SW1000XG [xgfactory.com] sound card, a mid-range Phonic [phonic.com] 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.
Do your own double blind listening tests (Score:2, Informative)
If you really want to do reliable tests on wav files, then visit PCABX [pcabx.com] to get the PCABX program [cdabx.com] 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 [r3mix.net]. 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 [belgacom.net] carried out at r3mix that showed the blind preferences of 42 users over a month of time.
Why use a lousy format in the first place? (Score:2)
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.
Vorbis on portable MP3 players: A challenge (Score:2, Interesting)
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...
shorten is lossless (Score:2)
you can even get a streaming shorten plugin [etree.org] 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.
Archiving v. just listening (Score:2, Interesting)
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
Lame VBR --r3mix (Score:2)
I must've failed (Score:2, Interesting)
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
Ogg is much better, at least in high bit rates! (Score:4, Interesting)
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
Clue! (Score:5, Insightful)
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.
Re:Clue! (Score:3, Informative)
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.
While digital interfaces bring a theoretical possibility for a quality change over analog links, this is _not_ due to the properties of the cables or jacks.
Short of a connector totally covered in corrosion, no jack or reasonable cable will ever influence signals in the audio band.
Even SPDIF and TOSLINK aren't lossless
Yes they are, these are straight digital interfaces. Short of malfunction no data will be lost through them.
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.
VBR is part of the mp3 stadard, so it's not a hack by any stretch of the imagination.
VBR is IMO the Right Way(TM)to do audio coding as it essentially let you select a target quality instead of a target bitrate.
Current implementations of VBR are good enough to not degrade the sound noticeably so there is no real reason not to encode with VBR.
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.
If joint stereo is a hack, then what do you call all the other techniques that make up mp3/ogg/whatever encoding.
JS simply utilizes the fact that significant signal is common for both channels and encodes this only once. Storing this information twice makes little sense.
JS is a efficient way to reduce space, which can be used to increase overall sound quality by using less aggressive compression on areas which actually matter.
96K Ogg 320K MP3 (Score:3, Interesting)
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, some terminology and thoughts from us at Xiph (Score:5, Informative)
BTW, for more in depth discussion that has been ongoing, have a look at the forums at r3mix.net [r3mix.net] and the Ogg-specific forums at Hydrogen Audio [hydrogenaudio.org]. 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 [xiph.org] 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 [xiph.org]
.ogg wins in a blind test (Score:4, Interesting)
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.
*blink* ye gods. (Score:5, Interesting)
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 [airwindows.com] 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...
Sounds better? (Score:3, Funny)
"Ogg" just makes me feel like I'm choking on a donut.
:-)
Re:Bit Rates (Score:2, Funny)
LOL good one!
If you were an audiophile, you wouldn't be listening to compressed music at all..
Re:Bit Rates (Score:5, Funny)
Re:I can't understand why most ppl use CBR for MP3 (Score:2, Insightful)
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.]
Re:I can't understand why most ppl use CBR for MP3 (Score:2, Informative)
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.
Re:nerves (Score:2, Informative)
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.
Re:nerves (Score:4, Informative)
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.
Re:nerves (Score:3, Insightful)
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.
Re:nerves (Score:3, Informative)
SHN, perhaps?
Re:nerves (Score:2)
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?
Simple. (Score:2)
There is a definite, easy to hear diffrence if you compare them though.
Re:nerves (Score:2)
It completely baffles me that... (Score:4, Informative)
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
Re:What? (Score:2, Informative)
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.
Re:What? (Score:2)
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.
Arggh! Bad units... (Score:5, Insightful)
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.
Re:try shn format (Score:2)
Re:For me, ogg sounds streets ahead (Score:2)
My music already IS that way, no encoder is ever going to make it sound natural
Re:VBR at "100%" quality (Score:2)
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.
Re:My favorite way to rip.... (Score:2, Informative)
EAC and LAME are the best way to rip and encode mp3s (respectively)
EAC: http://www.ping.be/satcp/eac00.htm [www.ping.be] Use secure mode when ripping, its slower but you wont get pops, clicks and bleeps.
LAME: http://www.hot.ee/smpman/mp3/ [www.hot.ee]
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 [r3mix.net]
Re:Is this a valid test for of audio compression? (Score:4, Informative)
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.