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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.

14 of 660 comments (clear)

  1. ogg vorbis all the way! by auttie · · Score: 5, Informative

    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
  2. Try VBR before you go to 300kbps by Dast · · 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 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.

  3. How to do listening tests by Dr.+Spork · · Score: 5, Interesting
    If you really care which format sounds best and want your listening tests to be taken seriously, do them right. (I worry many people don't really care about the sound, and want to just take this topic as an opportunity to plug the format which they favor for political reasons.)

    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

    1. Re:How to do listening tests by jonathan_ingram · · Score: 5, Informative
      If you don't have a spare friend, you can use the ABX testing method to see if you can distinguish between two files. Take a file, compress it, save to a WAV, and then give the files to the ABX program, which acts as your spare friend :)

      If you're running Windows, you can get ABX from http://www.pcabx.com/. On UNIX systems, the LAME source code 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.

  4. Define your fidelity level by brink · · Score: 5, Interesting
    Given that mp3 and ogg are each lossy (there will be erasure of sound elements) to a certain extent, this question is almost sort of disingenuous. Ultimately the answer reduces to "What sounds good to you?" As the question is stated, it kind of sounds like you want to be sort of an audiophile, but not go all out (which I can relate to, trust me.)

    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
  5. Re:Bit Rates by BorgDrone · · Score: 5, Funny

    I just saw a nice quote today: "audiophiles are people who listen to the audio equipment, not the music"

  6. Re:MPEGplus by jonathan_ingram · · Score: 5, Informative
    You must be using Windows. Monkey's Audio is a Windows only format... and you should not trust your data to a single-OS file format (yes, I would say the same for Linux-only file formats). Use FLAC instead.

    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 have shown that even at 128kp/s MPC is up there with AAC (MPEG-4 audio).

  7. Re:I am reminded... by jonathan_ingram · · Score: 5, Informative

    The tests that you link to were done incredibly badly, and should just be ignored.

    Here is a test 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.

  8. Re:Ogg by seanadams.com · · Score: 5, Insightful

    There are two chips which are very common for MPEG decoding in portable electronics - the MAS3507D and the STA013. Both of these chips are essentially "black boxes" - MPEG in, PCM out. Their DSPs have just enough horsepower to do MPEG decoding, and the firmware is all in ROM. Ogg decoding, as many have already pointed out, needs considerable amount of additonal CPU cycles and RAM as compared to MPEG. Ogg just wasn't designed for embedded systems. Right now the only remotely viable solution for OGG decoding in a portable device would be to go with something like an ARM system-on-chip. Would you pay $250 for a portable player that supported OGG when you can get an equivalent MP3 player for $150? I didn't think so.

    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.

  9. Clue! by tqbf · · Score: 5, Insightful
    How are surround sound speakers relevant to this discussion? Do you actually listen to music filtered through some cheesy "concert hall" effect?

    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.

  10. Arggh! Bad units... by Dominic_Mazzoni · · Score: 5, Insightful

    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.

  11. Re:Ogg by fossa · · Score: 5, Insightful
    "Ogg sounds better, but I can't go to walmart and buy a portable Ogg player."

    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...

    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 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.

  12. BTW, some terminology and thoughts from us at Xiph by xiphmont · · Score: 5, Informative
    My first thought when I saw this article was, "Oh boy... this should get ugly and yet remain light and fluffy" but all the posts I've seen (reading at +2) have been pretty good. I don't really have much of anything to add other than 'we have some really nice quality improvements in store for rc3', mainly new noise estimation metrics, lots of stereo fixes, and other random nicities (like 20kHz cutoff at 128...)

    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

  13. *blink* ye gods. by Chris+Johnson · · Score: 5, Interesting
    There _are_ no coincidences :D

    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:

    • new powermac to take on ADAT editing duties and run the quirky old transfer card I have
    • OS 8.6
    • Amadeus 2 v 3.2.3, which imports and exports Ogg- unsure quite what version- and Amadeus isn't free, but the deal is I _have_ bought it earlier and my registration number works on 3.2.3
    • iTunes (more on this later)

    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:

    "Conclusion: Ogg Vorbis, at least the version I tested, is not wildly superior to mp3. Used at bit rates under 192K it tries much harder to encode real high-frequency data, but on some sounds such as a tone sweep its sophistication backfires, producing artifacts that show up plainly in the sonograms."
    "However, used at higher bit rates it strikes a very clever balance, managing to pull together the best qualities of wildly different mp3 encoders into a single sonic presentation. Again, it behaves similarly to the very impressive BladeEnc in tonal purity, but instead of the miserable transient behavior of BladeEnc, it mimics the overstated transient behavior of Fraunhofer. This could easily be seen as best of both worlds."

    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...

    ...because I write them. And that means that although I am not a Linux C coder- since the code and the algorithms for quadratic and primitive root residue dithers and indeterminate-order noise shaping are in the GPL sphere, the Linux world can have those technologies freely- and the proprietary world can't. Which may mean that Linux players (mp3 or Ogg) can fairly easily boast strikingly better sound quality than proprietary ones...

    It's exciting to see the pieces of a truly superior free audio technology come together...