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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
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...
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
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Theoretically you should arrive at the same results (or at least very similar results), but I don't see the need for the CD burning process. I guess any double-blind test will do the job, though.
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.]
No. You should pick music that's most representative of what you're likely to be compressing. This is supposed to be a test of compression as a practical technique- what sounds best to you- not some kind of theoretical comparison. If you happen to listen to noisey old music and a particular compression approach neatly removes the noise, you're a fool not to use it.
There's no point in questioning authority if you aren't going to listen to the answers.
What difference does it make if your receiver does Dolby Digital? Your MP3s aren't an AC3 source. Receivers with "all the bells and whistles" are often of LOWER quality than those dedicated to doing one task well. Dolby Digital is for movies with earth-shattering-kabooms.
Are there really people here that think a "16 bit" sound card can't reproduce full CD audio? How do you think they play WAV files?
It's amazing the number of completely irrelevant factors people are bringing up here. Is there a word for the phenomenon that occurs when someone shells out money for something and then feels the need to factor its presence into anything remotely related to it?
It's also amazing that nobody is bringing up some REAL issues:
The quality of your connectors is more important than that of your sound card. Bring the audio to your receiver over SPDIF or TOSLINK, not over analog RCA cables! Sound cards --- ALL of them --- have really awful RCA connectors.
Even SPDIF and TOSLINK aren't lossless --- but the conveyance of waveform audio in your computer to your audio peripherals is. Since the inside of your computer has lots of interferance (hard drives, power supplies), it logically makes sense to deliver your audio as far away from your computer as possible before converting it to send to your receiver.
So USB audio makes a *lot* of sense for setups that simply want to do faithful MP3 playback --- a cheap Roland UA-30 will do SPDIF, TOSLINK, powers itself off the bus, and can sit yards away from your computer.
I don't understand the original question or some of the responses regarding bit rates. I encoded my entire CD collection at 192kbs MP3. I'm not an audiophile by ANY means (and I don't want to be: I'd rather not TRAIN myself not to like my sound system!!!) --- but I *regret* doing this; guitar and (real) drum driven music sounds awful in a good car stereo (Pioneer+JL+DynAudio) at 192, and tolerable at 256.
Even 2 years ago disk space was cheap enough to make 256 the reasonable choice. But when you can get a 75G stackable firewire drive/enclosure for less than $200, what possible incentive could you have for encoding at less than 256?
I can't tell the difference between 256 and anything above. VBR improves sound quality when you set a floor of 256 and a ceiling of infinity; otherwise, it's just a silly hack to save disk space at the expense of your MP3 files. It may not noticeably damage audio quality, but it sure as hell makes your MP3 files more complex, harder to analyze and play with/sort/etc. MP3 is just a poor file format for what VBR asks it to do.
Another big gotcha with MP3 is joint-stereo, the "reasonable default" in many encoders. Joint stereo is another psychoacoustic hack that saves an inconsequential amount of disk space at the expense of noticeable degradation in sound quality. It "spoofs" stereo for frequency ranges that its model believes is hard to localize in human ears. Make sure you nail your encoder at real stereo.
The most painful gotcha of all, fortunately, is one that most people have managed to avoid, and that is that codec quality is a HUGE factor. My original batch of 600 CDs was done with bladeenc (mass groan!); bladeenc is/was completely broken. People aren't kidding when they say that Fraunhofer sounds better than random other encoders. Fortunately LAME is a great choice.
As for Ogg: it's great that we have an open source codec. This will come in very handy for streaming audio delivery and for the cores of sound engines in games or other random programs. Because of this it's also great that Ogg is (apparently) more efficient than MP3. One hopes it will continue to become more and more efficient so it can give Microsoft's compromised but extremely efficient format a run for its money.
But since disk space isn't an issue, if you don't trust MP3 (putting you squarely in the minority), I'd say use Shorten or some other lossless format before making the irrevocable decision to put all your music into young Ogg Vorbis. It takes a *long time* to re-encode all of your CDs (*sob*).
Remember this: your time is far more valuable than disk drive space. Don't encode your music to the weak sound system you may have now: encode it to the ideal, even if you can't exploit it now, so that you'll be able to listen to your music without wasting time re-encoding it later on.
I have no idea where you got the idea that 128/44 is standard CD quality. I'm not even sure what 128/44 means.
Let's figure out what the bitrate of CD-quality audio is:
1. 44100 Hz (i.e. 44 kHz)
2. Two channels
3. 16 bits per sample
44100*2*16 = 1411200 bits per second, or 1411 kbps. That's the bitrate of CD audio.
Note that these are bits, not bytes. A CD takes up 1411/8 = 176 kB per second.
So the fact that an MP3 sounds pretty good at 192 kbps (which is 24 kB per second - the capital B for Bytes instead of bits) is actually quite impressive. It's compressing by about a factor of 7.
Luckily, most rippers don't even give you a choice. They just rip the raw bytes and stick a WAV header on each track. Good rippers verify that they're reading the CD correctly, of course, but they don't do any compression or re-encoding.
I 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.
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.'"
Short answer: Yes. That is, once you find a sampling rate for which (1) you can't hear any difference, or (2) the difference you hear doesn't bother you, then go with that one, and spend your money on upgrading your audio chain instead of buying bigger hard drives.
Longer answer: It's not a static, answer-it-once-and-forget-it process. The better the gear, the better it will be at showing up the flaws in whatever you're playing. And the more "educated" you become as a listener, the more sensitive you will be to the flaws that are there. Either of these factors can make today's "acceptable" recording be "unacceptable" tomorrow. So repeat your comparison periodically. You may change your mind.
If you're using MP3s primarily for storage/handling convenience (and if you'll have future access to the source material), use the lowest sample rate that will do the job for you today. But be prepared to find that once you've upgraded your equipment, you may find yourself disatisfied with your recordings.
But beware: If you're planning to keep your music for a while, especially if you won't be able to re-rip it, I'd suggest that at minimum you use a sampling rate that generates no perceptable difference (to you) now. I've got a lot of stuff on 1/4-inch quarter-track tape that I dubbed in the 60s and 70s at 3.75 ips. Most doesn't get played any more, but the gems that do make me want to kick myself for not spending the extra buck or so that it would have cost to double the recording speed (with better frequency response and signal-to-noise ratio) at the time.
Always try to use your best technology for the irreplacable stuff, even if you can't hear the difference. Today that means high sampling rates and lossless compression. Thirty years from now, you'll be glad you did.
I've done personal tests and compared the quality of 320 mp3 ( max bitrate ) to 160 WMA ( max bitrate ). To me, every wma file sounds a lot better than the mp3s and it takes considerable less space. About half. Maybe Microsoft got something right.