Ok, I'm getting a bit bored of this now. It wasnt intended to spark up a huge debate, and I dont claim to be an expert on it. Continually picking small holes in my posts would seem to be a waste of time. The purpose of this thread was an attempt to improve the quality of peoples mp3's which, I'm sure is something a lot of people here are interested in doing.
Waldo, If you think I'm wrong, please provide a reason, and an alternative.
As I said, Im no expert, and If you want to argue with what I'm saying...first read the 'quality' and 'critique' pages at
www.r3mix.net as everything I'm saying came (almost) straight from there.....these arent my opinions.....
Comments:
Picking bones over the title of the thread is up to you, but I did comment earlier that it was to get people in to read.
Just because someone invented something, doesnt mean they are always gonna be the best. Sony invented the CD player, but are about average in terms of quality.
In terms of quality, I agree, an mp3 will never be CD quality, but the point is, if you can get an mp3 to a reasonable size, such that 'NOONE' can tell the difference (and I keep saying there have been tests to prove this) then I'm, sorry, but loss in 'technical quality' does not necessarily mean loss in sound quality. If the mp3 sounds the same as the wav, then I'm happy.
Again, to quote r3mix.net:
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03. Nice page, but no matter what bitrate, there's always quality loss.
(Beefchow) FYI all mp3's no matter what the bitrate have quality loss... it's not lossless compression. 'Close to cd' is all you can aim for.
(zZoVutu) Very good site, despite it says that you can have perfect mp3 cd ripping (that it isn't true as we all know) the guy knows what he says.
"lossy compression" Does not mean you defacto lose sound quality.
This is what the C't test proved. That at a certain bitrate the mp3 and the original is judged to be of the same quality. Therefore: a perfect transparent music encoding makes the source and the encoded material sound exactly the same.
The problem is that most people play mp3's through their soundcard on their computer through a poor quality cable and then they say: "my original cd sounds better". Quite normal I would say. The original tracks and the decoded tracks on cdr is the only way to make a good comparison.
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As for juggling settings for different songs thats rubbish. The whole point of VBR, is it analyses the song and puts the compresssion where its needed. every song I have ever encoded with the settings I described 'sounds' exactly the same as the original....if I changed the settings, all I would do would change the size/quality ratio.
VBR measures the level of a signal and if it's under the ATH (absolute treshold of hearing) the signal will be left out of the encoding.
The LAME site is good, but tells you very little real information about mp3 quality....they even point you at r3mix.net. To quote
http://www.sulaco.org/mp3/ :
-r3mix.net Up-to-date ripper and encoder information, and the truth about MP3 quality.
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I'm getting totally bored with this, so here is a copy from the 'quality' page in r3mix.net, which, if you can be bothered to read, explains all this shit....
Facts:
128 kbit/s is not cd quality
256 kbit/s is cd quality (x) (in case of Lame or some Fraunhofer, not Xing)
In february 2000 c't magazin organised a blind listening test. 300 Audiophiles were involved, finalists tested 17 1-min clips from different artists (classic and pop):
original CD recording
128 Kbit/s Joint Stereo [MusicMatch (FhG) v4.4] encoded PC decoded Mac
256 Kbit/s Joint Stereo [MusicMatch (FhG) v4.4] encoded PC decoded Mac
all on cdrs and played in a Recording Studio on:
B&W Nautilus 803, Marantz CD14 with amp PM14 (Straightwire Pro cabling and extra's) [DM30000- so bit more than $15000]
Sennheiser Orpheus Electrostatic Reference-headphones with tweaked accompanying amp (digital and analog out) [>$10000]
Conclusions:
90% of the 128 Kbit material was picked out
MP3@256 was rated to have the same music quality as cd!
If you find MP3@256 to be of inferior quality compared to the original cd, you're very likely to be doing something wrong with the test (correct decoder, no objective double blind testing, DSP filters distorting the process, ...) Maybe this is something for you. You can always read the article in the german c't 6/2000 on p92.
The treshold of mp3 transparency lies somewhere between 128kbit/s and 256kbit/s, depending on the kind of music and your hearing and equipment.
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Why not to use any Xing encoders (Xing, Audio Catalyst, MpegDJ, ...):
stereo separation problems in joint stereo mode (listen with headphones)
all above 16kHz is mutilated, at any given bitrate (up to 320, and VBR High)
In mp3 you can have long and short blocks. Xing encoders don't use "short blocks", which, by definition, leaves Xing encoding quick peaks and sharp signals not nearly as accurate as with short blocks, which are used in Blade, FhG and LAME encoders.
the code is buggy, and music will get mangled from time to time (try first few seconds of Rammstein on Matrix Soundtrack)
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Knowing the facts of mp3, you could, if space is not really an issue use:
cbr 256kbit/s by Lame or some Fraunhofer encoders
Remarks: Guaranteed perfect(x) transparent encoding, but guaranteed overkill on most parts of the music. Also: space is always an issue if you use mp3, because otherwise why no use a lossless (eg. zip) compressor for your music (eg. WavPack, Monkey's Audio or my favorite LPAC) or just store the wavs?
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The classic trade-off between space and quality for mp3-archival quality is:
cbr 192kbit/s by any Fraunhofer encoder (audioactive, radium codec, mp3enc, ...)
Remarks: Decent sound quality, but not perfect so no archival quality. Clearly audible encoding artifacts on some music when using hq headphones. Everytime you see a vbr encoder take a bitrate >192 to encode a frame, you know that 192 is not sufficient for that part of the music.
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LAME brings us the first (and still only) optimally tweaked (unlike Fraunhofer) VBR mp3 encoder that does not mess up (unlike Xing):
lame --r3mix infile.wav outfile.mp3 (LAME 3.89b)
"--r3mix -b112" is synonym for "--nspsytune --vbr-mtrh -V1 -mj -h -b96 --lowpass 19.5 --athtype 3 --ns-sfb21 2 -Z --scale 0.98 -X0", explained here
Remarks: Perfect(x) sound quality at optimal bitrate. Sounds perfect to me and many others with -V2 -b32, but for the sake of people with very expensive audio equipment and "golden ears" an extra few bytes (about 10% size increase) are allowed with this setting. Downside to Lame VBR is the slightly longer encoding time when comparing to CBR. The resulting average bitrate will be 170-175 kbit/s (tested on +500 random mp3's). On a rare occasion you will get a small file (120 kbit) and some very rare tracks require mp3 to use up to 260-270 kbit on average to be reproduced with good quality. However, the global average for all your music is 170-175 kbit/s. If you're into really loud and busy music, like some metal collections, your album averages could be around 200kbit/s. If you're into classical music, your album average may be 160kbit/s. To learn more why this size difference, read here.
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Why VBR?(answer by ThomasLG)
VBR seems like a no-brainer to me. Near the beginning and ending of a song (assuming it starts and ends softly), where the volume is lower, and the music is less "demanding" in terms of its encodability, it makes sense to drop the bit rate, simply because there's not much there to encode, and the wasted space is overkill. In the middle of the song, where it may be more complicated, the idea of giving the encoder the option of "bumping up" the rate on a frame-by-frame basis is great! You may end up with a file that's the same overall size as a 170kbps CBR, but that uses frames as low as 32 on the really dead parts, and as high as 320 on the really tough parts. The bitrate is dynamically adapting to keep the quality constant. To know that the whole file isn't bloated where it isn't necessary, is a real bonus.
As you see there are fragments of a few seconds when the mp3 requires 128kbit/s to sustain quality, and there are parts when the mp3 uses 224 and 256kbit/s to sustain that same quality. Now, as a CBR user, what would you do? Or you choose to encode 128kbit/s and have some horrible sounding seconds in your music clip, or you're left to encode this clip in 256 or even 320 kbit/s, wasting a lot of bits in the process. VBR is being used in all new compression techniques: AAC, MPC, ... and also in MP3, as the standard defines!
So you don't even have to know anything to use it... 
