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This is an old revision of this page, as edited by 24.50.186.211 (talk) at 07:46, 5 September 2006 (→‎Legality and Acceptance: section deleted because the guideline violations have been remedied long ago). The present address (URL) is a permanent link to this revision, which may differ significantly from the current revision.

Bit-rate

The russian music download site allofmp3.com seems to be offering mp3's at a maximum bit-rate of 384kbps using the LAME codec. A qucik google search didn't provide any details on this, wondering if anyone knew anything worth adding.

I think the maximum for LAME mp3 is 320kbps. It is ~500kbps for Ogg's, though. --Russoc4 14:53, 29 June 2006 (UTC)[reply]

Update: If they are 384kbps, then they aren't mp3s. Most likely mp2s. From the article:

   * Layer 1: excellent at 384 kbit/s
   * Layer 2: excellent at 256...384 kbit/s, very good at 224...256 kbit/s, good at 192...224 kbit/s
   * Layer 3: excellent at 224...320 kbit/s, very good at 192...224 kbit/s, good at 128...192 kbit/s

LAME free-format allows for up to 640kbps, though practically nothing can decode it. 70.45.49.36 03:57, 11 July 2006 (UTC)[reply]

MPEG-I/II

In the first line it says "MPEG-1/2", and this needs to be explained on the page, I think. Here's what I know about it:

Phase 1 can handle input streams (or WAV files) with a sample frequency of 48000, 44100 or 32000 Hz and is therefore used most often, obviously.

Phase 2 will only support stream for 24000, 22050 and 16000 Hz. Basically, Phase 2 is intended for lower bit rates (e.g. for voice communication, or if you need small files with reduced quality, podcasts and live online audio-feeds and the likes).

The lowest bit-rate for Phase 2 is 8 kBits/sec while for Phase 1 the lowest bit rate is 32 kBits/s. 195.64.95.116 23:11, 11 Mar 2005 (UTC)

Response to 195.64.95.116: There are several problems in the above comments. Please refer to an authoritative source such as The official MPEG site or The MPEG Industry Forum site or perhaps the somewhat less official MPEG.ORG site. Be very careful to only get your information from such reliable sources, as there is a significant amount of confusion found on more random web sites. Do not call the MPEG-X numerical suffixes "Phases". Do not use roman numerals to denote them. MP3 originated from MPEG-1 Audio Layer 3, also properly referred to as MPEG-1 Part 3 Layer 3 (where "Part 3" refers to audio coding, "Part 1" is multiplexing, "Part 2" is video coding, etc.). The stuff above about sample frequency looks wrong too. I'm not personally aware of any connection between MP3 and MPEG-2, except for the former being a predecessor of the latter. Pangolin 06:16, 12 Mar 2005 (UTC)
Actually, it is exactly the other way around; You have to call the old MPEG-X Phases, since they are, and always have been. .MP2 and .MP3 used to both even be called .MPA and this was way before the Part 1 or Layer 3 issues came into play. Hey, I know, because I was there testing and using it back in the day. I don't really care what you think it should be, I know what the Phases meant, why they are there in the format, and when they started putting it in. The "Part 1" came into play because of incompatibility with Part 2, which is different from Phase 1 and Phase 2. MP3 originated from MPA, which in turn changed to MPEG Phase 1 Layer 3. To help make it clear, it was decided by its creators (hey, ask them) to use the Roman for the Layers. Furthermore, I don't know what fool put in the MPEG-2 part about 'the new' MP3, but that's a silly thing to do; That new MPEG-2 is not the same as MP2, the new MPEG-2 is no longer describing the Layer or Phase, it isn't even 'downwards' compatible with MP3.195.64.95.116 02:25, 28 Apr 2005 (UTC)
MPEG-2 Audio Layer 3 is also valid, and not limited to low bitrates (i have often seen MPEG-2 transport streams with 128kb/s Layer 3 audio). Furthermore, 8 kb/s is not ISO MPEG-2; it is part of a Fraunhofer sub-spec, known as MPEG-2.5—which is not endorsed by the ISO. MPEG-4 can also contain Layer 3,2,1 streams, as well as VQ and AAC. —Brian Patrie 05:27, 1 December 2005 (UTC)[reply]

Piracy

I think there should be a heading regarding the allegations of "piracy" and the RIAA lawsuits. perhaps mention of the mp3/warez "trading" scene? Does anyone agree? Alkivar 04:56, 19 Oct 2004 (UTC)

VERY much agreed. User:Afolkman 1:41, 16 Nov 2004
It would be more appropriate to link to an external article, as this subject is not confined to the MP3 domain. I see, though, that we now have an MP2 and MP3 and the Internet section, so i've linked the word "piracy" to Pirate (disambiguation). I considered linking to copyright infringement, but decided that this would tend to reinforce the media-nurtured impression that the two are synonymous. I added the latter to See also. —Brian Patrie 06:07, 1 December 2005 (UTC)[reply]

Summary and Psychoacoustics

There are two things that I don't like about this page:

  • I think the one-line summary on the very top of the page should give more information -- one has to trawl way down the page before psychoacoustics are mentioned
  • The History section contains useful information, but I think there's too much babble about claimed or presumably more correct bitrates. That part should move to the Quality section.

I've changed the link to psychoacoustics from Pycho-acoustic coding to Psychoacoustics as there is a redirect. --Cpk 21:20, 4 Sep 2004 (UTC)


Numbers, Parts and Layers

MP3 refers to MPEG-1 Layer 3. MP2 (audio files) refer to MPEG-2 Layer 3. AFAIK, MPEG-2 Layer 3 is basically the same as MPEG-1 Layer 3, with some slightly different packetization. Is it worth even putting it in the list of similar formats? AAC is also known as MPEG-2 AAC, but this probably isn't worth worrying about. -D

MP3 actually refers to all MPEG layer 3 audio. MP3 at sampling frequency at least 32 KHz is called MPEG-1 layer 3 and uses MPEG-1 packets; MP3 at sampling frequency up to 24 KHz is called MPEG-2 layer 3 and uses MPEG-2 packets. "MP2" is primarily MPEG-1 layer 2 audio used in classical MPEG applications such as CD-i and Video CD, but you'll often find MP3 files labeled as MP2 to get them through file type filters on web hosting services. Winamp processes all files named *.mp2 and *.mp3 as generic MPEG audio, sending them to its "Nitrane" MPEG audio decoder. See http://www.mpeg.org/MPEG/MPEG-audio-player.html --PP

Argh. Looks like you're right. I'm still confused, though. According to the MPEG specs, MPEG-1 Layer 2 describes video, Layer 3 describes audio. Are there sub-layers to "Layer 3", and is that what we're talking about? I find this whole thing very messy, and it'd be nice to clean it up on the MPEG pages. -D

Response to -D: Don't confuse "Parts" and "Layers". Part 2 is video. Part 3 is audio. In the case of Audio Layer 3, the term "Layer" refers to a lower level of the hierarchy than the term "Part". Layer 3 is something inside of Part 3. Pangolin 06:51, 12 Mar 2005 (UTC)

Codecs and Algorithms

Hmm, I'm a bit confused. 'codec' is said to be the _same_ as an audio compression algorithm. I would think a codec is a specific _implementation_ of an audio compression algorithm. Am i just plain wrong? or? --arcade

Yes, in my opinion, codec (coder + decoder, analagous to modem being short for modulator + demodulator) is something completely different than an algorithm. While an algorithm could be said to be a set of instructions to yield a desired result, a codec is an implementation of both an algorithm and also the reverse of the same algorithm, to aid media creation and conversion tools. In other words, I think they're two completely different things. I'll fix this problem in the description. Even the article page for Codec that this one links to, says that a codec is a device or program processing the data in some way. I.e. not an algorithm, which is only a set of instructions on how to process the data. --Jugalator
It's even more complex a picture than that. In this case, the standard specifies only the decoder, and says very little about the encoder. So the standard does not specify a codec, only part of one. The algorithm used for encoding is not specified. But, properly, the term algorithm is a general term that can also apply to the specified decoding process by itself. Pangolin 06:51, 12 Mar 2005 (UTC)

NPOV complaints

This article reads like it was written by an audiophile. Choice excerpts:

  • To many other listeners, 128 kbit/s is unacceptably low quality, which is unfortunate since many commonly-available encoders set this as their default bitrate.
  • It is important to know that despite of all the flaws, recent multiformat listening tests (http://www.rjamorim.com/test/multiformat128/results.html) once again show that LAME MP3 easily rivals its technological successor AAC. (Vorbis aoTuV is tied with Musepack at first place, Lame MP3 is tied with iTunes AAC at second place, WMA Standard is in third place and Atrac3 gets last place). (bold original)

Most of the stuff under the encoder comparison is also POVish. The Alternatives is similar. Random speculation has worked its way in. Finally, the Online Music Resources is marginal. - Fennec (はさばくのきつね) 03:34, 10 Jul 2004 (UTC)

What part of that "despite of all the flaws" did you find 'audiophile'-ish? You can't disregard the fact that inherent to this format (MP3) there are quality-issues involved. Everybody wants (and needs) to know that, in order to understand what MP3 is about. So, to then state under some Quality-section that MP3 is some kind of ugly sounding bad quality trashy format, seems very unfair to me. Check these if you are not convinced; http://www.heise.de/ct/00/06/092/ (I believe there is a translated version of it somewhere) http://jthz.com/mp3/#MYTH So, that is why I posted the bold part; use the right encoder, with the right config, and you'll have flawless quality MP3 encoding. It's been proven. 195.64.95.116 01:30, 5 Sep 2004 (UTC)

I agree, this does read like it was written by an audiophile. Feel free to edit if you don't like it.

The entire MP3 quality section reeks of NPOV and unverified information, especially statements like this "However, listening tests show that with a bit of practice many listeners can reliably distinguish 128 kbit/s MP3s from CD originals [...] reaching the point where they consider the MP3 audio to be of unacceptably low quality." Needs a rewrite for sure (and not by some audiophile with no sources). 70.45.49.36 04:06, 11 July 2006 (UTC)[reply]

I find the main text perfectly reasonable, and these complaints hard to follow and inconsistent. MP3 is an audio format. I have no clue if the author considers him/herself to be an "audiophile", but certainly I see no shame in being an audiophile, or any problem in the idea that an article about an audio format should be written by an audiophile. Who better ? someone who is not interested in sound and isnt able to distinguish high quality and low quality reproduction ? I personally DO consider myself to be an audiophile, yet am almost ashamed to say that I sometimes struggle to distinguish 128 bit MP3 from a CD original. So its good. But its not perfect and under ideal conditions I can easily tell, and for thius reason I consider 128 bit unacceptable, and always rip at max quality 320 bit VBR. Which I think fits perfectly with the authors text. Whats the problem guys ? / Dave smith

The problem is no sources. Even an audiophile would be OK as a source, if published and referencable, but an editor's own audio opinions are not encyclopedic. Dicklyon 18:57, 25 July 2006 (UTC)[reply]
As above. It doesn't matter how plausible the text is or how well it jives with your subjective experience. The article makes several unsourced claims about relative "quality" (which is not defined in specific terms) of MP3s of various bitrates, and it makes several unsourced claims that people who make claims about audio quality have been proven to be unreliable. It's not unreasonable to demand sources be cited or the section be rewritten or deleted.
Not related to the article, but responding to your comments, you should be aware that using VBR results in the lowest possible bitrate being chosen for each frame to maintain (not drop below) a certain level of quality (and in LAME at least, that quality level is verified by trial and error). If you want max quality per frame, you'd be using 320 kbps CBR, so that the encoder is under no pressure to drop the bitrate on frames where it thinks it can get away with it. However, 320 kbps CBR is unreliable in some encoders (even LAME), introducing a kind of ringing artifact in the high end, depending on what other settings you use (highpass filter, psy model) and the content of the audio. —mjb 19:11, 25 July 2006 (UTC)[reply]
I'd just like to say that i think an audiophile is the perfect candidate for writing on this sort of topic if it had good sources. Now i know ive come across articles on the internet (not blogs) that have tested mp3 satisfaction on subjects, it just a matter of finding them. ive read the section and it all seems to match up with what i rememeber so i think leaving the [citation needed] will do for now. also, im goin to activly look for said articles whenever i can now. also, i propose that the NPOV tag get removed and replaced with a {{Unreferenced}}. -(chubbstar)talk | contrib | 15:42, 26 July 2006 (UTC)[reply]

Quote:

  • MP3 files encoded with a lower bit rate will generally play back at a lower quality. With too low a bit rate, "compression artifacts" (i.e., sounds that were not present in the original recording) may appear in the reproduction. A good demonstration of compression artifacts is provided by the sound of applause: it is hard to compress because of its randomness and sharp attacks. Therefore compression artifacts are audible as ringing or pre-echo.

Isn't the mp3 compression scheme 'adaptive' in some sense that it only throws out perceptively irrelevant data-subspaces? It seems like ringing/pre-echo artifacts would be associated with other forms of stuff, not medium-quality mp3... is there a source to verify this? Ninjagecko 04:24, 7 August 2006 (UTC)[reply]

It does try to make sure the errors due to coding are perceptually small, or masked. But since the filters are essentially time-symmetric, it puts as much error before a transient, where it's likely to be audible, as after, where it's more likely to be masked. That is the scheme is not perfect, and some of its imperfections are these "precursor" sounds. I don't have a reference about who finds these to be audible at what bit rate. Dicklyon 04:41, 7 August 2006 (UTC)[reply]

MP3: Now with NPOV

The old article was full of awful propaganda, so I revamped it and moved some things into the LAME article. D. G. 10:44, 4 Sep 2004 (UTC)

I kinda disagree with you on that. As with Winamp, also LAME is very much to blame for the grand success of the format. Let's not forget that this was one of the first and highest quality FREE MP3 encoders out there (next to Blade), whilst others needed payment or licensing before being used. So, when speaking of MP3, one needs to speak of LAME. The one would never be this popular without the other. I would hardly call that propaganda, or POV, it's simple fact. 195.64.95.116 01:21, 5 Sep 2004 (UTC)

I'm not sure what you're proposing is a "proven scientific fact." There's no such thing as a proven scientific fact. Anyway, anything is "up for discussion." You simply can't make an edit and declare that it is unquestionable. I believe you are saying that it's a "scientific fact" that nobody can detect the difference between 256kbps MP3 and an uncompressed source. That is untrue, although almost nobody can tell the difference, some golden ears listeners can on certain samples. If you're going to insert these statements you should back them up. You cannot just say "numerous listening tests."
I deleted the paragraph from the design limitations section, because this is already discussed in the MP3 quality section. Please clarify it there if you want. Rhobite 23:01, Oct 3, 2004 (UTC)

It is proven scientific fact that 1 added to 1 makes 2. If you can't get that, you're not worth my time, and I'm not going to discuss proven facts concerning our hearing or MP3 quality, I have better things to do. If you want to state silly 'claims' on MP3 quality which have no base whatsoever other than gossipy (trying to sound like an expert) speak, I'm going against that. I would refer to rec.audio.pro and being a member of the Audio Engineering Society. The point where experts, the high-end listeners and the likes, will not be able to distinguish the MP3 file from the original currently lies around 180 kbps VBR, and 224 kbps CBR mp3 files. This is where 50% of them will say that the mp3 is the original and/or vice versa, i.e. where they can't tell the difference. (This is researched on using LAME.)195.64.95.116 18:53, 4 Oct 2004 (UTC)

Can you link to a study, other than that German study from 4 years ago? Can you link to any specific posts on rec.audio.pro?
You can all test it yourselves, this is quite easy to do. If you can't understand that, you don't belong in this discussion anyway. It's like discussing existence of gravity, or the magnetic polar fields. They are there, and you can't babble on about it the way you want to. Furthermore, the articles in rec.audio.pro or elsewhere would not be read or understood by you anyway. 195.64.95.116 16:04, 31 Dec 2004 (UTC)
The burden is on you. If you refuse to discuss your edits or link to any studies, you are not "worth my time" either. Please do not curse in your edit summaries, and do not personally attack me.
I do as I see fit thank you very much. You must have been deserving of me personally attacking you. 195.64.95.116 16:04, 31 Dec 2004 (UTC)
MP3 and CDDA are not directly comparable. I will not defer to your self-proclaimed "authority."
Of course they are; they are both the end-medium formats people listen to.195.64.95.116 16:04, 31 Dec 2004 (UTC)
Also, if you insist on reverting, please don't reintroduce your own grammar and spelling errors. Rhobite 19:13, Oct 4, 2004 (UTC)
If anything I've corrected yours.195.64.95.116 15:54, 31 Dec 2004 (UTC)

It may help here to consider a fact as something that is not known to be disputed anywhere today by otherwise reasonable people. Since there is obviously some dispute here about this "fact", our WP:NPOV policy arises to meet the occasion. Perhaps it is time to "characterize the dispute" if necessary, or merely to to back off on the fact with something like, "many people even claim they cannot detect a difference between P and Q." I will watch this page for a while. Be sure you are well familiar with the contents of the WP:NPOV article. And as always, remember wikilove and have a nice day! Tom - Talk 16:35, 6 Oct 2004 (UTC)

any criticisms toward the technology would be personal points of view made by the users of mp3's.

This makes no sense. Quality of audio-reproduction can be measured. If this wasn't the case, something like MP3 wouldn't exist, nor would it sound as good as it does these days.195.64.95.116 15:54, 31 Dec 2004 (UTC)

as they are based on the opinions of persons they should not be included in the article. instead try adding links to reveiws made by some sort of professional organization and let reader form their own opinions instead of trying to guide their opinions with your own through the artical--Larsie 21:40, 19 Oct 2004 (UTC)

Compression scheme vs. encoding scheme

CDDA does not compress audio. It is uncompressed 44.1 kHz, 16-bit stereo audio. CDDA is merely a format for encoding this audio along with error correction. MP3, on the other hand, is a compression format which can compress many sample rates and sizes, including 44.1/16/stereo, as well as a wide range of other combinations. Such as 22 kHz, 48 kHz, and even 96 kHz. Because of this, CDDA and MP3 are not directly comparable. You simply CANNOT say that one is better than the other, it's apples and oranges. Rhobite 21:42, Oct 27, 2004 (UTC)

Agreed, you cannot simply say that one is "better" than the other. But this is not because CDDA does "encoding" while MP3 does something completely different called "compression", but because the domain of MP3 is much greater than that of CDDA. Both are formats for storing PCM digital audio, and from the experience of the average computer user, both are overwhelmingly used for nothing except 44.1/16/stereo. For such an appropriately restricted application, they are perfectly comparable. One might say "using equipment XYZ and audio samples PQR, 50% of sample of 100 untrained listeners considered MP3 (using encoder MNO with settings JKL and bitrate ABC) to be not noticably worse than its CDDA source", and this would be a perfectly valid, reproducible test. [[User:Smyth|– Smyth]] 11:53, 30 Oct 2004 (UTC)
Fact remains that one can encode to MP3 from a much higher (high-end) quality audio source than the "box" where CDDA needs to fit in. For CDDA one would need to bring quality down, where samplerate, dynamics and bit-depth are concerned. This is not related to "tests" or opinions, this is sheer reality. CDDA has limitations as well, and they can actually be regarded as more important (reproduction-quality wise) than those of MP3. All this as long as it concerns playback quality and nothing more. 195.64.95.116 16:24, 31 Dec 2004 (UTC)

Sampling rate

The text refers to "available sample frequencies." Would it be possible to define what "sample frequences" or "sample rate" means?

Sampling frequency? [[User:Smyth|– Smyth]] 01:46, 8 Nov 2004 (UTC)

Value judgments

Wikipedia articles don't make value judgments or recommendations, such as recommending that non-professionals never have a reason to use lossless compression. In any case, this statement is not true: "Those who will only listen, do not need to use lossless compression, since they won't hear the difference with MP3." You can't make blanket statements like that, some people can indeed tell MP3 - even with a good compressor - from audio that has never been compressed.

They can up to a measured degree. This can be proven and it often has been. Beyond certain high enough bitrates NO HUMAN will be capable of telling the difference.195.64.95.116 16:16, 31 Dec 2004 (UTC)

Etree and archive.org distribute lossless copies of nearly every show - there is obviously a large group of listeners who feels MP3 is inadequate for their uses. Rhobite 23:06, Nov 15, 2004 (UTC)

The fact that MP3 is now often considered a degraded format, surpassed by other formats, has nothing to do with that. It's simply because
1) there are bad mp3 encoders around (lots of them in fact)
2) MP3 tends to need some type of special treatment beforehand to reach optimum quality, other f

Yes, it's true that people under stress, with bad music equipment or in a noisy or unsuitable environment can have trouble differentiating between mp3 and lossless, or don't care, but that doesn't mean that they still can't hear it. Lossy formats should not be recommended with a clear conscience. BKmetic 00:00, 23 July 2006 (UTC)[reply]

This may be completely off kilter but to me it would seem simpler to actually look at the 5 difference in the signal given the identical encoder and differing bit rates. Any time someone uses "excellent" "acceptable", "poor" and "good" are all opinions (as oposed to factual). I personally have tried to hear the difference between the 128 bit and the CD on a good systems and I can hear the difference (after carefully selecting the music and carefully listening). In my car I have a great stereo, that would not be the case there since the road noise etc interferes just enough that I am unable to discern the difference. I will not make a value judgement as to what is acceptable, good or excellent since my opinons would only hold true to me at this instance. If one could measue it objectively it would be much more helpful. Look at it as follows Break down the audio into 5 frequency bands. Weight each band as it impacts our hearing (the center 3 bands are more important for human hearing than those below say 400hz and above 6kHz <--- Example) Using a digital oscilloscope, measure the difference in peaks in each band, then try to determine the loss of the resonant frequencies as opposed to the main frequencies since they tend to give music that "depth" the audiophiles like (yes, that is a qualitative opinion, not measurable but I am trying). Doing so would create a simple, objective comparison of formats and you could also use the data to compare encoders and thus determine the quality comparatively (objectively) in lieu of throwing subjective valuations around. [Moto] --71.112.37.172 19:28, 1 August 2006 (UTC)[reply]

Your simple view of what it takes to make objective quality measurements on audio signals is uninformed by the years of engineering effort in this direction. It is a hard problem, partly because hearing is so complicated, and quality measurements need to correlate to what you hear. Dicklyon 19:53, 1 August 2006 (UTC)[reply]

Minor Tidbits to Cleanup

The link in the "see also" section to Marcy Playground's music album "MP3" belongs in disambiguation. 68.100.224.150 17:56, 29 June 2006 (UTC)[reply]

"It provides a representation of PCM"

This isn't true. MP3 represents frequency domains, it can be decoded to PCM or DSD for example or in theory directly to analogue.

PCM

(From User talk:Smyth:)

You recently reverted my edit on MP3. MP3 does not contain a representation of PCM. It contains a representation of audio in a completely different way to how PCM represents audio. You can decode MP3 to DSD. It is NOT compressed PCM like FLAC is. Since you reverted my edit, I suggest you do some more research and then revert it again once you have learnt. Thanks. -- 82.152.177.71 13:11, 16 July 2006 (UTC)[reply]

I find it hard to get a definite answer for this because PCM is so universal, but references to each MP3 frame encoding exactly 1152 samples, and even being marked with a PCM sampling rate, make it sound very much as if PCM has a special status. Compare JPEG files: they contain a representation of an image in a completely different way to a straightforward rectangular grid, but they nevertheless are a representation of a rectangular grid of pixels, and not something else. – Smyth\talk 15:22, 16 July 2006 (UTC)[reply]
If you don't know *what* PCM is, why on earth are you reverting edits related to the technical details of PCM? There are three essential ways to encode audio data (only two of which are widely used), and PCM simply uses an entirely different method than MP3.
Your analogy is not a very good one in this case. Horses and cars are the same because they can both be a means of transporting a person from one location to another, but that doesn't change the fact that they are very different on several fundamental metrics.
The specifications for PCM audio are available in several places on the internet, and they are generally technical (as is the case with most worthwhile specifications). Google is your friend, and if you don't understand the material then refrain from changing the information in the article.

Remove NPOV and replace with Unreferenced template?

i propose that the NPOV tag get removed and replaced with a {{Unreferenced}}. I would do it myself but ill be there might be disagreement which i think should be discussed. The only bit i think to be potentially not NPOV is the "Layer 1: excellent at 384 kbit/s, Layer 2: excellent at.. etcetc." part. Anyhoo. -(chubbstar)talk | contrib | 15:57, 26 July 2006 (UTC)[reply]

Comments on encoders

I absolutely agree with this part :

>>Good encoders produce acceptable quality at 128 to 160 Kbit/s and near-
>>transparency at 160 to 192 kbit/s, while low quality encoders may never reach
>>transparency, not even at 320 kbit/s. It is therefore misleading to speak of
>>128 kbit/s or 192 kbit/s quality, except in the context of a particular
>>encoder or of the best available encoders. A 128 kbit/s MP3 produced by a good
>>encoder might sound better than a 192 kbit/s MP3 file produced by a bad
>>encoder. Moreover, even with the same encoder and resulting file size, a
>>constant bitrate MP3 may sound much worse than variable bitrate MP3

— Preceding unsigned comment added by Eclipsed aurora (talkcontribs) 30 July 2006


MPEG I, II, II.5

In my CDex program I have the option to encode with MPEG I, II oder II.5. Which one should I select? 84.60.102.161 08:38, 28 August 2006 (UTC)[reply]

Choose MPEG I. II and II.5 are for low bitrates e.g. speech.

http://news.bbc.co.uk/2/hi/technology/5312696.stm --Daraheni 03:41, 5 September 2006 (UTC)[reply]