Discussion in 'Mixing & Song Critique' started by Marcus Black, Aug 7, 2003.

  1. Marcus Black

    Marcus Black Guest

    Just wondering which encoders you guys use for making mp3:s. And what rate do you think sounds good enough? And what´s an acceptable filesize for a 3-4 minute tune? I need to send mp3:s to labels all the time and when they sound good enough the file is too big and vice versa. Are there differences in encoders? Different formats? I´d love to find something that gives me ok sounding mp3:s that the labels won´t have to download forever. Currently I´m using "Mpegger". I´m a mac user. Thnx.
  2. Sen

    Sen Guest

    44.1 khz / 160 kbit/s at least...that will keep your songs of 4-5 minutes around 5Mb...i think...
    Any encoder will do I guess...why not use your recording software's encoder....
    good luck :cool:
  3. Alécio Costa - Brazil

    Alécio Costa - Brazil Well-Known Member

    Mar 19, 2002
    PT 5.1.1 has its own encoder at the bounce window options. Ya need to pay $10 Iguess so.
    For ordinary stuff at the PC, I use Nero or even Music match Juke Box. 192kbps/44.1k will keep stuff around 5 to 6 MB.
  4. falkon2

    falkon2 Well-Known Member

    Mar 17, 2003
    Aha! MP3 and lossy codecs - this is ONE area I can actually claim as my forte and not be lying. :D

    I religiously use the LAME codec with VBR (variable bit-rate) at alt-preset-extreme for all my archiving purposes, and alt-preset-standard for all my distribution purposes.

    Though I'm scrapping MP3 entirely for archiving - I prefer to use a non-lossy compression technique for that. Usually Monkey's Audio .ape, which cuts filesize by 50 - 60%. No matter how "transparent" a lossy codec is, there's treble content which is lost forever. Even if your ear can't hear the loss as it is, your ear would be able to if you ever boosted the treble (say, in a remixing session)

    If space AND quality is at a premium, though, MPC would be the way to go - the true successor to the MP3 format. It has the most accurate reproduction so far of all the lossy formats, and typically runs at about 250++kbps average.

    But back to the topic at hand: Why I use LAME?

    Well, simply because, no two encoders are the same - while audio samples as most people know them in digital are just that, "samples" of voltage at a specified rate (44.1kHz), MP3 is a frequency-based format - It stores frequency information vs time, rather than voltage levels vs time.

    Encoders use various algorithms to translate the thousands of digital values into frequency content, which is why MP3 is so much smaller than .wav/.aiff, and also why different encoders produce different results. Some algorithms are simply superior, and some are plain stupid. Some even introduce artifacts of their own, like treble flanging, or drastic cuts about 16kHz for no apparent reason.

    LAME, by far, has been decided as *THE* MP3 encoder to use by more than a handful of audiophiles ( for more)

    VBR - a method where rather than filling up a constant amount of frequency information per second (You're bound to always put too much or too little information in at any given moment to maintain the same quality throughout the song if you have a strict limit), additional algorithms within the encoder analyzes the audio, decides how much information is needed to make the encode "transparent". (being a relative term) The bitrate of the resulting MP3 changes throughout the song. This is *THE* best compromise between quality and filesize. Imagine having to print out a book and use exactly one page for each chapter. (no more, no less) If the chapter is too short, there's wasted space on the page. If the chapter is too long, you lose critical information.

    This is why I don't use encoders bundled with software whose primary purpose is something other than encoding. Let those software packages do what they're good at. Encoding, it's algorithms and inherent anomalies is already a science of it's own. Better to use something that was developed by someone who knows what the heck he's actually doing. :D

    More about LAME (Installation, simple use, presets, etc). There's a Q&A section halfway down the first post that answers quite a bit more on VBR vs constant bit rate, amongst other things.

    The frontend software package is for PC, though.
  5. heinz

    heinz Guest

    I second the recommendation for LAME, I have used it with great success for the last year or two. I have it hooked up as an external encoder in AudioGrabber.
  6. Marcus Black

    Marcus Black Guest

    falcon2: Thanks for a great reply :D

    Is LAME available for Mac? Anybody know anything about the Dolby-encoder that has been used to encode the songs on the i-tunes site?
  7. Alécio Costa - Brazil

    Alécio Costa - Brazil Well-Known Member

    Mar 19, 2002
    very nice info folks. but have you compared LAME with SHAME? ( oh bad joke about PT´s bundled encoder)
  8. Opus2000

    Opus2000 Well-Known Member

    Apr 7, 2001
    Depending on what software you have you may be able to bounce down to that format..

    Wavelab gives you two options..Lame or the one you have to pay for! I use the Lame as well and find it just fine for MP3's..I always do mine at 192/44.1 as this way you don't cause any aliasing in the higher frequency range!

    Cubase SX and Nuendo both offer this in the export audio feature as well! I think all software should have this as an option!

    Opus :D
  9. MisterBlue

    MisterBlue Member

    Mar 15, 2003
    Another vote for LAME frome here. I have it integrated with EAC (ExactAudioCopy). The combination rocks and - best of all - is totally FREE.

    Disclaimer: Must be a Windows user to play. Shirt and shoes required ;) .

  10. heinz

    heinz Guest

    Unless I am mistaken, ProTools employs the Frauenhofer codec (see this link). I am very familiar with that codec and IMO the LAME codec produces not only superior sonic quality, but gives you MANY more audio options upon encoding.
  11. falkon2

    falkon2 Well-Known Member

    Mar 17, 2003
    VBR! Don't go for 192! Go for VBR!

    I trust a complex algorithm more than I trust a single number to decide how much information needs to be jotted down to get the music to sound like it should sound.

    You'll notice that for most commercial CD's, VBR at average and above quality will always settle for something like >200kbps average. (This of course has quite a bit to do with everything becoming overcompressed and clipped - Each squared off top takes a cartload of frequency information to reproduce, but anyway, meh)

    One thing I really can't stand about the mp3 format is the fact that there's a split-second gap of silence at the start of the file (because of the inherent nature of how information is stored, this gap can never be eliminated from the encoding process). This can screw up timing pretty badly, and you can never get seamless transitions between tracks like those on live CDs.

    Newer formats get rid of this problem. (like MPC which I'm currently pimping)

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