The Linux MP3-HOWTO

Table of Contents

  1. Introduction.

  2. Copyright of this document.

  3. Where to get this document.

     3.1 Translations

  4. Acknowledgments.

  5. Disclaimer.

  6. Hardware Requirements & Performance Issues.

  7. Software Requirements.

     7.1 Rippers & WAV Recorders
     7.2 Encoders
     7.3 Players
     7.4 Streaming Servers
     7.5 Mixing
     7.6 ID3 Editors
     7.7 Stream Grabbers
     7.8 Misc

  8. Setting up your system.

     8.1 Setting up for Analogue Audio Capture
     8.2 Setting up for CD-ROM Audio Capture
     8.3 Additional Setting up

  9. Encoding from Audio.

  10. Encoding from CD-ROM.

     10.1 Command Line encoding
        10.1.1 RipEnc
        10.1.2 CD2MP3
     10.2 GUI Based Encoders
     10.3 Encoder Performance

  11. Streaming MP3's.

     11.1 Icecast
        11.1.1 Shout
        11.1.2 LiveIce
     11.2 Fluid
        11.2.1 Update for RC1
     11.3 Litestream
     11.4 Apache::MP3 Module
     11.5 Firewall Bypassing
     11.6 Bandwidth considerations
     11.7 Copyright Issues

  12. Listening to MP3's.

     12.1 Playing from File
     12.2 Playing from MP3 Streams
     12.3 Capturing MP3 Streams
        12.3.1 Streamripper
        12.3.2 wget
     12.4 Mixing
        12.4.1 eMixer
        12.4.2 GDAM

  13. ID3 Editing.
     13.1 id3ed
     13.2 mp3info
     13.3 ID3 Genre codes

  14. MP3 to Minidisc.

     14.1 Analog Recording
     14.2 Digital Recording
     14.3 Playlists

  15. Feedback.


  1.  Introduction.

  This document describes the hardware, software and procedures needed
  to encode, play, mix and stream MP3 sound files under Linux.

  It covers:

  ·  Encoding MP3's from a live or external source

  ·  Encoding MP3's from audio CD's

  ·  Streaming MP3's over a network

  ·  Listening to MP3's

  ·  Recording MP3 streams

  ·  Mixing MP3's

  ·  Editing ID3 information

  2.  Copyright of this document.

  This HOWTO is copyrighted 1999, 2000 Phil Kerr.

  This document is distributed under the terms of the GNU Free
  Documentation License.  You should have received a copy along with it.
  If not, it is available from

  3.  Where to get this document.

  The official version of this document can be obtained from the Linux
  Documentation Project  <>.

  The homepage for the most recent version of this HOWTO is:

  3.1.  Translations

  This HOWTO has been translated into the following languages:

  Please note that translations may be slightly out of date from this
  document as, naturally enough, the translations take time.


  <> By Lee,So-min <>


  <> By Arnaud
  Gomes-do-Vale <>


  <> By Andras Timar


  <> By Mariani
  Dario <>


  <> By Arielo


  <> By Reggy Ekkebus


  <> By Saito Kan <can->


  <> By Jernej
  Kovacic <>

  Many thanks to the above translators.  If you can translate this
  HOWTO, please drop the author an email. Also please state the URL
  where the translation will be housed.

  4.  Acknowledgments.

  In writing this HOWTO I have had to draw heavily on the Sound-HOWTO By
  Jeff Tranter, and the Sound-Playing-HOWTO By Yoo C. Chung.

  Thanks also to the other HOWTO authors whose works I have referenced:

  Linux System Administrators Guide By Lars Wirzenius.

  Linux Network Administrators Guide By Olaf Kirch.

  Multi Disk System Tuning HOWTO By Stein Gjoen.

  Also a big thank-you to all who have sent in feedback, comments and

  5.  Disclaimer.

  Use the information in this document at your own risk.

  I disavow any potential liability for the contents of this document.

  Use of the concepts, examples, and/or other content of this document
  is entirely at your own risk.

  All copyrights are owned by their owners, unless specifically noted

  Use of a term in this document should not be regarded as affecting the
  validity of any trademark or service mark.

  Naming of particular products or brands should not be seen as

  You are strongly recommended to take a backup of your system before
  major installation and backups at regular intervals.

  6.  Hardware Requirements & Performance Issues.

  Digital Audio processing is a resource intensive task that relies
  heavily on the processing and I/O capabilities of a system.  I would
  strongly recommend a Pentium class machine as a minimum.

  If you are going to be encoding from an analogue audio source via the
  line or microphone input, a PCI soundcard will give the best results.
  The I/O performance difference between an ISA and PCI based card is
  significant, over 132 MBytes/sec for PCI (quote taken from the PCI-
  HOWTO).  Naturally, the better the quality of the soundcard in terms
  of its signal-to-noise ratio, the better the encoded MP3.  I've been
  using the Soundblaster PCI128 and just switched over to a Soundblaster
  Live Value; both cards give good audio performance, but the Live has
  significantly better S/N ratios, good enough for semi-pro audio work.
  Remember the old data processing maxim:- garbage in - garbage out!

  Creative have a Beta driver for the Soundblaster Live! which can be
  downloaded from:


  When recording analogue audio to a hard disk, more commonly referred
  to as direct to disk or d2d recording, the performance of the disk,
  and its interface is critical.  If you are using an IDE based based
  system, mode 4 or UDMA is preferable as the transfer rate is
  sufficiently high enough to provide reliable data transfer without

  The ideal solution would be to use a SCSI based system as the drives
  and interface have far better throughput capabilities, a sustained
  5MB/sec for SCSI 1 through to 40-80MB/sec for ultra-ultra2/wide SCSI.
  IDE can peak at anything from 8.3 MB/s to 66 MB/s for Ultra DMA mode 4
  but these speeds are peak, average transfer rates will be slower.  If
  you can find, or afford, an AV SCSI drive, go for it.  AV drives have
  had the read/write head system optimised for continuous data
  transference; other SCSI and IDE drives normally cannot sustain
  continuous data transfer as the write head heats up!

  Naturally a drive that has cache will give more consistent results
  than one that doesn't, as the cache will act as a buffer if the heads
  do lift or it cannot handle the throughput.

  If your drive isn't up to spec, your recording will suffer from
  dropouts and glitches, where the drive failed to record the signal.
  If you are recording one-off events, such as live performances invest
  in a good SCSI based disk system.

  Another cause of d2d dropouts is a heavily loaded system.  Background
  tasks can cause the system to momentarily glitch.  Its recommended to
  run as few background services as you can, especially networked based
  services. For more information about setting network services, and
  startup scripts please refer to the SAG and NAG guides.

  Virtual memory paging will also cause glitches, so run with as much
  physical RAM as you can, I'd recommend at least 32 Mb, but you may
  well need more.

  For those wanting to extract the most out of their system, optimising
  the kernel probably wouldn't do any harm either.

  For streaming MP3's the better the network card the better the
  throughput, naturally a 100Mb interface will give better throughput
  than a 10Mb interface :)

  While the hardware specifications above will give you a decent system
  to encode audio data, don't discount using older, lower spec kit if
  that's all you have access to.

  It'll be a good challenge for a sys-admin to tweak a low-spec system
  to give good results, and the end result will probably be a happier
  Linux box.

  Another important issue is the audio cabling.  Cheap, poor quality
  cables and connectors will result in poor recording quality.  If your
  soundcard has the option to use phono, sometimes referred to as RCA
  connectors, use them.  Gold plated contacts will also help maintain
  audio quality, as will keeping audio cables away from data cables as
  there will be a chance of interference between them.

  But don't forget, spending a fortune on the best audio cabling will be
  lost if the rest of the system hasn't been optimised.

  For encoding MP3's from CD-ROM, the speed or type of drive will
  determine the time taken to read the raw information from it.  A
  single speed drive will probably be too slow for all but the most

  Your CD-ROM must be connected to your soundcard if you want to hear
  what you are recording, either using the internal connector or by
  connecting headphone's to the headphone output, although you will not
  be able to listen to MP3's through the CD-ROM headphone socket!

  For detailed instructions on setting up soundcards, now would be an
  excellent time to read the Sound-HOWTO.

  7.  Software Requirements.

  Converting audio to MP3's is normally a 2 stage process, first the
  audio is recorded into a WAV format, then the WAV is then converted
  into an MP3. Some utilities will do both processes in one go for you.

  The format you wish to encode audio from, CD or direct audio, will
  determine what software tools you need to produce the WAV file.

  If you are wanting to encode from audio input, you will need a program
  that will record from your soundcard's input and save the results in a
  WAV format. Below are some useful utilities (most of the comments are
  taken from the respective website of the app.)

  7.1.  Rippers & WAV Recorders

  To grab from analog audio line-in.  Wavrec

  Wavrec is distributed as part of wavplay, which can be downloaded


  To convert CD audio data to WAV format, sometimes known as CD ripping:




  Cdparanoia is a Compact Disc Digital Audio (CDDA) extraction tool,
  commonly known on the net as a 'ripper'. The application is built on
  top of the Paranoia library, which is doing the real work (the
  Paranoia source is included in the cdparanoia source distribution).
  Like the original cdda2wav, cdparanoia package reads audio from the
  CDROM directly as data, with no analog step between, and writes the
  data to a file or pipe in WAV, AIFC or raw 16 bit linear PCM. Compared
  to cdda2wav, it's much slower but really gets the best results you can
  get even from CDs that are difficult to rip for scratches or other



  RipEnc is a bourne shell script frontend to Cdparanoia, cdda2wav,
  tosha and Bladeenc, 8hz-mp3, l3enc. It utilizes CDDB lookups to
  automate the naming of songs as they are ripped. A manual naming
  option is also available. The entire CD can be ripped or you can pick
  the songs to rip. ID3 tags are also supported.



  cd2mp3 is a freeware software, designed to handle 2 other programs
  cdda2wav(that rips the tracks from cd) and lame (that encodes the



  RipperX is a GTK program to rip CD audio and encode mp3s. It has
  plugins for cdparanoia, BladeEnc, Lame Mp3 encoder, XingMp3enc, 8hz-
  mp3 and the ISO v2 encoder. It also has support for CDDB and ID3 tags.



  Grip is a GTK-based CD-player and CD-ripper/MP3-encoder. It has the
  ripping capabilities of cdparanoia built in, but can also use external
  rippers (such as cdda2wav). It also provides an automated frontend for
  MP3 encoders, letting you take a disc and transform it easily straight
  into MP3s. The CDDB protocol is supported for retrieving track
  information from disc database servers.  Grip works with DigitalDJ to
  provide a unified "computerized" version of your music collection.


  7.2.  Encoders

  To convert the WAV file to MP3 format you will need an encoder:

  Blade's MP3 Encoder

  BladeEnc is a freeware MP3 encoder. It is based on the same ISO
  compression routines as mpegEnc, so you can expect roughly the same,
  or better, quality . The main difference is the appearance and speed.
  BladeEnc doesn't have a nice, user-friendly interface like mpegEnc,
  but it is more than three times faster, and it works with several
  popular front-end graphical user interfaces.



  In the great history of GNU naming, LAME stands for LAME Ain't an Mp3
  Encoder.  LAME is not an mp3 encoder.  It is a GPL'd patch against the
  dist10 ISO demonstration source.  LAME is totally incapable of
  producing an mp3 stream. It is incapable of even being compiled by
  itself. You need the ISO source for this software to work.  The ISO
  demonstration source is also freely available, but any commercial use
  (including distributing free encoders) may require a license agreement
  from FhG (Fraunhofer Gesellschaft, Germany).



  This is a very fast MP3 encoder for x86-CPU, which is based on LAME
  ver 3.29beta and optimized by PEN@MarineCat, Keiichi SAKAI, URURI, kei
  and shigeo.  (You will also need to download NASM to compile the
  source, which can be found  <>)


  7.3.  Players

  To play the MP3's you will naturally need a player:

  Xmms (Formerly known as X11Amp)

  This player has most of the features as Winamp  from Windows 95/98/NT
  but it will of course feature some specials only available for the
  Linux version.



  Xaudio is a very fast and very robust multi platform solution for
  Digital Audio playback, especially targeted at MPEG Audio (MP1, MP2
  and MP3) decoding.



  AlsaPlayer is a new type of PCM player. It is heavily multi-threaded
  and tries to exercise the ALSA library and driver quite a bit. It has
  some very interesting features unique to Linux/Unix players. The goal
  is to create a fully pluggable framework for playback of all sorts of
  media with the focus on PCM audio data.  Full speed (pitch) control,
  positive *and* negative! First Linux- and only GPL player that does
  this!! MP3's and CD's do varispeed :)



  What is mpg123? It is a fast, free and portable MPEG audio player for
  Unix.  It supports MPEG 1.0/2.0 layers 1, 2 and 3 (those famous "mp3"
  files), and it has been tested on a wide variety of platforms,
  including Linux, FreeBSD, NetBSD, SunOS, Solaris, IRIX, HP-UX and
  others. For full CD quality playback (44 kHz, 16 bit, stereo) a
  Pentium (or fast 486), SPARCstation10, DEC Alpha or similar CPU is
  required. Mono and/or reduced quality playback (22 kHz or 11 kHz) is
  even possible on slower 486 CPUs.



  FreeAmp is an extensible, cross-platform audio player. It features an
  optimized version of the GPLed Xing MPEG decoder which makes it one of
  the fastest and best sounding players available. FreeAmp provides a
  number of the most common features users have come to expect in a
  clean, easy to use interface.


  7.4.  Streaming Servers

  Streaming servers allow you to 'broadcast' MP3's across a network,
  whether this is your intranet or the internet itself.


  Welcome! Icecast is a Mpeg Layer III Audio broadcasting system brought
  to you by the team.  Icecast comes bundled with
  iceplay, and icedir. Iceplay is a playlist streamer that will allow
  you to send pre-encoded files to your Icecast server.



  Fluid Streaming Server is a program for streaming media over networks
  and in its current form using the mp3 format.

  <> (old site)

  <> (new site)


  Litestream is an Open Source, infinitely scalable high-capacity MP3
  streaming system for Unix.



  Module for streaming MP3's using the Apache WebServer.

  This module takes a hierarchy of directories containing MP3 files and
  presents it as a browsable song library for streaming over the web.
  [Available from CPAN]

  7.5.  Mixing


  LiveIce is the source client for Icecast which encodes an mpeg stream
  for broadcast as it is created. Unlike clients such as Shout and IceDJ
  this permits the broadcast of live audio, rather than prerecorded

  LiveIce is bundled with Icecast, newer versions together with
  documentation may be found at the website below:



  eMixer is an easy-to-use front-end to mpg123 that allows you to play
  and mix two mp3 streams together. The ability to mix two mp3s makes
  eMixer act like a cross-fader, effectively giving the user DJ-like
  capabilities from the computer console. eMixer is also very able in a
  "real time" party environment. eMixer is based on the original mp3
  mixing code upon which liveice's mixing component is built.



  GDAM is real-time digital dj mixing software package. Any number of
  mp3 files can be played and mixed simultaneously. Effects can be
  added, changed, and rearranged dynamically. GDAM features a client-
  server architecture; all sound is produced by a server, which receives
  instructions from any number of clients. Other features include
  plugins for audio effects and interface components, caching and
  looping, sequencing, assisted beatmatching, a waveform viewer/beat
  calculator, contiguous queuing (no pauses between songs when using a
  playlist), an online help system, an mpg123 clone which runs on a gdam
  server, flexible command line interface for direct control of server,
  recording of entire mix or any point in stream to buffer/disk/mp3
  encoder, support for multiple sound devices, and support for midi
  hardware control.




  7.6.  ID3 Editors


  id3ed is an ID3 tag editor for mp3 files. You can set tags
  interactively or from the command line, or a combination of both.
  id3ed can set genre by name or number. You can also remove or view



  MP3info is a small utility for reading and writing MPEG Layer 3 (MP3)
  ID3 tags. Console (command-line and interactive ncurses) and GTK
  versions are included.


  7.7.  Stream Grabbers


  Streamripper records shoutcast streams. If the stream contains track
  info (meta data), streamripper creates a separate file for each track.



  GNU Wget is a freely available network utility to retrieve files from
  the World Wide Web using HTTP and FTP, the two most widely used
  Internet protocols. It works non-interactively, thus enabling work in
  the background, after having logged off.

  Wget is far more than a stream grabber!


  7.8.  Misc

  Volume Normalization


  If you have encoded live audio, or have encoded from older cd's you
  may find variations in the overall sound level.

  To change the encoded volume levels of the MP3's you will need to
  normalise them using wavnorm.



  Sox is a very handy sound conversion utility which I'd recommend
  having, and you will need it if you wish to use wavnorm.


  You may also need a mixer program; Xmixer works well and is included
  with most distributions.

  8.  Setting up your system.

  This section will describe the basics of setting up your Linux system
  to record audio from either an analogue or CD-ROM source.

  I'm basing this section around my Intel based Linux system which is
  running Redhat, but should be reasonably distribution neutral. If you
  have any success in using this HOWTO on other hardware, please get in

  Naturally a reasonable prerequisite is a working soundcard.  At this
  point in the HOWTO, I invite you to read the excellent Linux Sound
  HOWTO, by Jeff Tranter.  After which a good read of the Linux Sound
  Playing HOWTO, by Yoo C. Chung.  Both of the above mentioned HOWTO's
  cover the details of getting a sound system working under Linux far
  better than I could.

  8.1.  Setting up for Analogue Audio Capture

  Firstly, set up your audio.  There are a multitude of ways to route
  audio before it gets to your Linux box, some common ones are:

  Line out to Soundcard Line in.  Most audio devices have a Line output
  sockets.  Line level is a standard that specifies what voltage the
  audio device will send out. If I remember correctly it is 500mV for
  domestic and Semi Pro devices, and 750mV for Pro audio devices.  I
  would guess that the standard set for most soundcards will be 500mV,
  but some of the newer Pro audio may be to the higher standard  It
  shouldn't make too much difference unless you are recording at very
  high levels.

  The Line level output is normally used to connect HI-FI equipment to
  an amplifier, so things such as Tape Decks, Radio Tuners, CD players,
  DAT machines and Mini-Disc players should connect without problem.
  Turntables can be more of a problem, see below for more information.

  You could capture audio from VCR's as well.  Most VCR's will either
  have Line out for sound, or you can Get a Line out from a SCART socket
  if your VCR has one.

  Amplifier Tape out to Soundcard Line in,  Soundcard Line out to
  Amplifier Tape in.  This configuration is essentially replacing a
  traditional tape recorder connected to your HI-FI amplifier with your
  Linux system.  The Soundcard Line out to Tape in allows monitoring of
  the recording levels.

  Mike to Soundcard Mike in.  The voltages generated by microphones is
  very much smaller than those used in Line level devices.  If you were
  to plug a Microphone into the Soundcard Line in, chances are you would
  never record anything.

  WARNING, doing the reverse, plugging a Line level device into the
  Soundcards Microphone input, can damage your soundcard!!

  Turntable to Mike in.

  Many thanks to Mark Tranchant for the following.

  The raw output from a record deck cartridge is very low level.
  However, you cannot plug it directly into a microphone input and
  expect good results. The output requires equalization, as records are
  mastered with less bass and more treble to optimize the physics of the
  moving needle. This equalization is carefully defined and referred to
  as RIAA equalization. You *need* to run the output through a phono
  preamp first, and then into a line input.

  Music keyboards & synths should be connected to the Soundcards Line
  in, with guitars connecting to Line in via a DI (Direct Injection,
  used to convert the signal to Line level) box.

  Before you plug in anything into your soundcard, make sure the volume
  levels are turned down to minimum, or if using microphones they are
  either turned off or away from speakers.

  8.2.  Setting up for CD-ROM Audio Capture

  Setting up your Linux system to extract audio data from CD-ROM is
  reasonably straight forward.

  If you can hear a track playing from your CD-ROM through your speakers
  or amplifier, connected to your soundcard, then there's a reasonable
  chance you should be able to record from it.

  8.3.  Additional Setting up

  Log in as per normal to your system, then using a mixer program set
  the recording levels that are loud enough to give you a decent
  recording level, but aren't too loud and distorting.  I normally just
  judge this by ear, after a while you'll get to know what levels are
  best for your kit.

  I recommend either turning off all unnecessary services or switching
  to the single user runlevel, especially when encoding from an audio
  source. This is to ensure that the bare minimum of services are
  running and thus minimising system glitches when recording.

  I've set up a separate SCSI drive, exclusively to record the audio to,
  which I'll refer to as /mp3.  I've done this mainly for the
  performance gains in using a SCSI drive.  Also, recording onto a
  dedicated drive, where you are almost certain the head isn't going to
  suddenly skip to another part of the drive as you are writing audio
  data to it, is a good thing :)

  For details on setting up a Linux system with multiple disk drives, a
  good read of the Multi-Disk-HOWTO, by Stein Gjoen may be useful.

  9.  Encoding from Audio.

  Firstly, make sure you have enough space on your drive.  At CD
  quality, 44.1 Khz 16 Bit stereo, 1 minute takes nearly 10 Mb (5 MB per

  I normally record at DAT quality, which is 48 Khz 16 Bit stereo.

  Using wavrec I use the following syntax:

  [dj@megajukebox]$ /usr/local/bin/wavrec -t 60 -s 48000 -S /mp3/temp.wav

  The first part is an explicit path to wavrec.  The '-t 60' specifies
  the length of time to record for, in seconds.

  The third option, -s 48000 refers to the sample rate in samples/sec.
  (48000 is the rate for DAT, 44100 is CD)

  The last option is the path to the output file.

  To see the full set of options, run waverec -help, or see it's man

  This will produce your WAV file  Next you will need to encode it into
  MP3 format.

  Use bladdenc with the following command line.

  [dj@megajukebox]$ /usr/local/bin/bladeenc <em/[source file] [destination file]/ -br 256000

  The -br option  sets the bit rate, in this case I've set the rate to
  the maximum rate of 256k bits/s.  The path to bladeenc may also be
  different on your system to the one I've used in my example.

  To see the full set of options, run bladeenc -help, actually this is
  an invalid option, but will display the list of options.

  The same encoding using Lame (as well as Gogo as it is based on Lame)
  would need the command line

  [dj@megajukebox]$ /usr/local/bin/lame <em/[source file] [destination file]/ -b 256

  10.  Encoding from CD-ROM.

  In a similar way to encoding from audio, encoding from CD is a 2 stage
  process.  Firstly the audio data is extracted from the cd and
  converted into a wav file.  Then the wav file is converted into MP3.

  There are basically 2 types of encoders, console based and X based.
  Both do the same job, but the X based are easier to use (and look

  Again, before you start to encode, check you will have enough drive
  space on your system.

  10.1.  Command Line encoding

  I've written a very simple Perl script that will rip and encode tracks
  from a CD.


  if ($ARGV[0] ne "") {

  $count = 1;

  do {

  $cdcap = system("cdparanoia", $count, "/mp3/cdda.wav");
  $track = "$ARGV[1]/track".$count.".mp3";
  $enc = system("bladeenc  /mp3/cdda.wav $track -br 256000");

  until $count > $ARGV[0];

  else {
  print "Usage cdriper [no of tracks] [destination directory]\n\n";

  Please note: The above script is very basic and has nothing fancy,
  like error checking or CDDB.  Improve at your leisure :)

  The main lines of interest are:

  $cdcap = system("cdparanoia", $count, "/mp3/cdda.wav");

  This line calls the CD ripper, cdparanoia.  Cdparanoia converts raw CD
  audio data to WAV format.

  I'm using Cdparanoia, but if you wish to use CDDA2WAV, the command
  line would be:

  $cdcap = system("cdda2wav", $count, "/mp3/cdda.wav");

  The salient options are $count, which is the number of tracks to rip,
  and then the path for the outputted WAV file.  In my example this will
  go to a tmp directory on my MP3 SCSI drive.

  The WAV file is then converted into a MP3 file using Bladeenc.

  I've written this Perl script in order to rip a CD without having to
  rip and encode each track, and without having to use the batch mode of
  Cdparanoia.  This cuts down on free disk space needed as Cdparanoia's
  batch mode will rip the whole disk, and take up anything upto 600 Meg.

  If you wanted to use Lame or Gogo, replace the encoder line with:

  $enc = system("lame  /mp3/cdda.wav $track -b 256");


  $enc = system("gogo  /mp3/cdda.wav $track -b 256");

  Here is a dump of the available option for each of the encoders.


  BladeEnc 0.91    (c) Tord Jansson          Homepage:
  BladeEnc is free software, distributed under the Lesser General Public License.
  See the file COPYING, BladeEnc's homepage or for more details.

  Usage: bladeenc [global switches] input1 [output1 [switches]] input2 ...

  General switches:
    -[kbit], -br [kbit]  Set MP3 bitrate. Default is 128 (64 for mono output).
    -crc                 Include checksum data in MP3 file.
    -delete, -del        Delete sample after successful encoding.
    -private, -p         Set the private-flag in the output file.
    -copyright, -c       Set the copyright-flag in the output file.
    -copy                Clears the original-flag in the output file.
    -mono, -dm           Produce mono MP3 files by combining stereo channels.
    -leftmono, -lm       Produce mono MP3 files from left stereo channel only.
    -rightmono, -rm      Produce mono MP3 files from right stereo channel only.
    -swap                Swap left and right stereo channels.
    -rawfreq=[freq]      Specify frequency for RAW samples. Default is 44100.
    -rawbits=[bits]      Specify bits per channel for RAW samples. Default is 16.
    -rawmono             Specifies that RAW samples are in mono, not stereo.
    -rawstereo           Specifies that RAW samples are in stereo (default).
    -rawsigned           Specifies that RAW samples are signed (default).
    -rawunsigned         Specifies that RAW samples are unsigned.
    -rawbyteorder=[order]Specifies byteorder for RAW samples, LITTLE or BIG.
    -rawchannels=[1/2]   Specifies number of channels for RAW samples. Does
                         the same as -rawmono and -rawstereo respectively.

  Global only switches:
    -quit, -q            Quit without waiting for keypress when finished.
    -outdir=[dir]        Save MP3 files in specified directory.
    -quiet               Disable screen output.
    -nocfg               Don't take settings from the config-file.
    -prio=[prio]         Sets the task priority for BladeEnc. Valid settings are
                         HIGHEST, HIGHER, NORMAL, LOWER, LOWEST(default) and IDLE
    -refresh=[rate]      Refresh rate for progress indicator. 1=fastest, 2=def.
    -progress=[0-8]      Which progress indicator to use. 0=Off, 1=Default.

  Input/output files can be replaced with STDIN and STDOUT respectively.


  LAME version 3.50 (
  GPSYCHO: GPL psycho-acoustic model version 0.74.

  USAGE   :  lame [options] <infile> [outfile]

  <infile> and/or <outfile> can be "-", which means stdin/stdout.

      -m mode         (s)tereo, (j)oint, (f)orce or (m)ono  (default j)
                      force = force ms_stereo on all frames. Faster and
                      uses special Mid & Side masking thresholds
      -b <bitrate>    set the bitrate, default 128kbps
                      (for VBR, this sets the allowed minimum bitrate)
      -s sfreq        sampling frequency of input file(kHz) - default 44.1
    --resample sfreq  sampling frequency of output file(kHz)- default=input sfreq
    --mp3input        input file is a MP3 file
    --voice           experimental voice mode

      -v              use variable bitrate (VBR)
      -V n            quality setting for VBR.  default n=4
                      0=high quality,bigger files. 9=smaller files
      -t              disable Xing VBR informational tag
      --nohist        disable VBR histogram display

      -h              use (maybe) quality improvements
      -f              fast mode (low quality)
      -k              disable sfb=21 cutoff
      -d              allow channels to have different blocktypes
    --athonly         only use the ATH for masking

      -r              input is raw pcm
      -x              force byte-swapping of input
      -a              downmix from stereo to mono file for mono encoding
      -e emp          de-emphasis n/5/c  (obsolete)
      -p              error protection.  adds 16bit checksum to every frame
                      (the checksum is computed correctly)
      -c              mark as copyright
      -o              mark as non-original
      -S              don't print progress report, VBR histograms

    Specifying any of the following options will add an ID3 tag
       --tt <title>     title of song (max 30 chars)
       --ta <artist>    artist who did the song (max 30 chars)
       --tl <album>     album where it came from (max 30 chars)
       --ty <year>      year in which the song/album was made (max 4 chars)
       --tc <comment>   additional info (max 30 chars)

  MPEG1 samplerates(kHz): 32 44.1 48
  bitrates(kbs): 32 48 56 64 80 96 112 128 160 192 224 256 320

  MPEG2 samplerates(kHz): 16 22.05 24
  bitrates(kbs): 8 16 24 32 40 48 56 64 80 96 112 128 144 160


  GOGO-no-coda ver. 2.24 (Feb 12 2000)
  Copyright (C) 1999 PEN@MarineCat and shigeo
            Special thanks to Keiichi SAKAI, URURI, Noisyu and Kei
  This is based on LAME3.29beta and distributed under the LGPL
  gogo inputPCM [outputPCM] [options]

   inputPCM is input  wav file
  if input.wav is `stdin' then stdin-mode
  outputPCM is output mp3 file (omissible)

  -b  kbps     bitrate [kpbs]
  -br bps      bitrate [ bps]
  -silent      dont' print progress report
  -off         {3dn,mmx,kni(sse),e3dn}
  -v {0,..,9}  VBR [0:high quality 9:high compression]
               You should combine this with -b option
  for only RAW-PCM input
  -offset bytes skip header size
    -8bit       8bit-PCM [dflt 16bit-PCM]
    -mono       mono-PCM [dflt stereo-PCM]
    -bswap      low, high byte swapping for 16bitPCM
    -s kHz      freq of PCM [dflt 44.1kHz]
  -nopsy       disable psycho-acoustics
  -m  {s,m,j}  output format s:stereo, m:mono, j:j-stereo
  -d  kHz      change sampling-rate of output MP3
  -emh {n,c,5} de-emphasis
  -lpf {on,off} 16kHz filter [dflt use if <= 128kbps; not use if >= 160kbps]
  -test        benchmark mode
  -delete      delete input file, after encoding

  10.1.1.  RipEnc

  RipEnc performs the same task as the code above, but is written in
  shell and is easier to use :)

  Here's what it looks like.

  RipEnc version 0.7, Copyright (C) 1999  Michael J. Parmeley
  <>, RipEnc comes with ABSOLUTELY NO WARRANTY

  There is currently NO encoding process running in the background
  Your encode.log file is 982607 bytes long.

  <Enter 'd' for details, 'v' to view the encode log, or 'del' to delete the encode log>

  1) Change working directory....................[/megajukebox/tmp]
  2) Choose encoder..............................[lame]
  3) Choose ripper...............................[cdparanoia]
  4) Choose id3 tool.............................[none]
  5) Toggle between Manual and CDDB naming.......[manual]
  6) Setup XMCD_LIBDIR variable for CDA..........[/var/X11R6/lib/xmcd]
  7) Set preferred naming convention.............[artist-name_of_song.mp3]
  8) Rip whole CD?...............................[no]
  9) Set small hard drive option?................[no]
  10) Please select your Cd-Rom device...........[/dev/cdrom]
  11) Set the Bitrate for the encoded MP3's......[256]
  12) List the files in your working directory
  13) Start
  14) About
  15) Exit

  10.1.2.  CD2MP3

  Cd2mp3 is a single pass cdda to MP3 encoder.  It will convert a audio
  track to MP3 without producing an intermediate wav file.

  Here's how it looks in operation:

  [dj@megajukebox]$ cd2mp3 options ALL
  Using language: 1 - English.
  Translator: Gustavo Sverzut Barbieri(k-s) <>

  %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% Cd2Mp3 1.0 %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
  Author: Gustavo Sverzut Barbieri (k-s) <>
  *** device: -D/dev/cdrom        type: -Icooked_ioctl
  *** audio device: /dev/dsp
  *** preset: tape
  *** copyright: Yes
  *** author:
  *** album:

  recording: '1' as 'track-1.mp3' (wait)
          Ok! (recorded)
  recording: '2' as 'track-2.mp3' (wait)

  Naturally you can feed it options from the command-line, here is a
  dj@megajukebox]$ cd2mp3 --help

  %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% Cd2Mp3 1.0 %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
  Author: Gustavo Sverzut Barbieri (k-s) <>
    cd2mp3 options <track_number>="<track name>" ... <track_number>="<track name>"or:  cd2mp3 options ALL  (to record all tracks)
  or: cd2mp3 options LIST=<reclist_path>  (to read pairs: <track_number>="<track name>" from a file)
  ou: cd2mp3 options PLAY=<track_number> VOL=[0..100] (only plays the track)

    DEV=<cdrom_device> (default: /dev/cdrom)
    DEV_TYPE=[generic_scsi|cooked_ioctl] (default: cooked_ioctl)
    AUDIO_DEV=<audio_device>  (default:/dev/dsp)
    LANGUAGE=<language_number> (LANGUAGE=help to see supported languages)
    PRESET=[phone|voice|fm|tape|hifi|cd|studio] (PRESET=help will give more info)
    ALBUM="<album's title>"
    AUTHOR="<author's name>"

  10.2.  GUI Based Encoders

  GUI based encoders offer all the functionality of console based
  encoding, but wrap it all up in a nice easy to use interface. Grip and
  RipperX are similar in operation, both offer you the ability to select
  one, several or all tracks on a CD and convert them.  They also offer
  CDDB support which allows you  to retrieve the album and track
  information from a server and saves you having to enter the
  information by hand.

  10.3.  Encoder Performance

  In the encoding sections I've mentioned 3 different encoders,
  bladeenc, lame and gogo.  The main difference is their performance in
  encoding (although there are differences in the available options
  which were listed earlier).

  A little example.  I ripped a track from a CD and then encoded it with
  the different encoders. All encoders were run with the same system
  conditions and all produced stereo out mp3's.

  [dj@megajukebox]$ ls -l cdda.wav
  -rw-rw-r--   1 dj       dj       59823164 Feb 10 00:56 cdda.wav

  [dj@megajukebox]$ bladeenc cdda.wav -br 256

  BladeEnc 0.91    (c) Tord Jansson          Homepage:
  BladeEnc is free software, distributed under the Lesser General Public License.
  See the file COPYING, BladeEnc's homepage or for more details.

  Files to encode: 1

  Encoding:  ../test.wav
  Input:     44.1 kHz, 16 bit, stereo.
  Output:    128 kBit, stereo.

  Completed. Encoding time: 00:05:58 (0.78X)

  All operations completed. Total encoding time: 00:05:58


  [dj@megajukebox]$ lame cdda.wav -b 256
  LAME version 3.50 (
  GPSYCHO: GPL psycho-acoustic model version 0.74.
  Encoding ../test.wav to ../test.wav.mp3
  Encoding as 44.1 kHz 128 kbps j-stereo MPEG1 LayerIII file
      Frame          |  CPU/estimated  |  time/estimated | play/CPU |   ETA
   10756/ 10756(100%)| 0:02:28/ 0:02:28| 0:02:29/ 0:02:29|    1.9074| 0:00:00


  [dj@megajukebox]$ gogo cdda.wav -m s -b 256
  GOGO-no-coda ver. 2.24 (Feb 12 2000)
  Copyright (C) 1999 PEN@MarineCat and shigeo
            Special thanks to Keiichi SAKAI, URURI, Noisyu and Kei
  MPEG 1, layer 3 stereo
  inp sampling-freq=44.1kHz out sampling-freq=44.1kHz bitrate=256kbps
  inp sampling-freq=44.1kHz out sampling-freq=44.1kHz bitrate=128kbps
  input  file `../test.wav'
  output file `../test.mp3'
  {  10751/  10755} 100.0% (  2.94x)  re:[00:00:00.03] to:[00:01:35.42]
  End of encoding
  time=  95.430sec

  It would appear that Gogo has a much optimised algorithm for encoding
  than Bladeenc and Lame.

  11.  Streaming MP3's.

  A streaming server allows you to transmit MP3 files over a TCP based
  network.  This can be the Internet itself or your local network /

  The connection principal is very similar to that of a web server,
  files are streamed when a client (the MP3 player) connects to the

  11.1.  Icecast

  After downloading and untaring, a good look around the doc/ directory
  would be a good thing, the HTML manual is very helpful and

  If you have downloaded the source code, follow the instructions
  regarding compiling pertinent to your system.

  Icecast will not work correctly unless you correctly set the
  servername in the config file, icecast.conf, which is located in the
  etc directory. It must match the name that resolves to your IP address

  If you see the following line when Icecast starts-up you have

  -> [05/Jan/2000:17:21:04] WARNING: Resolving the server name [] does not work!

  Edit icecast.conf which is located in the etc directory and locate the
  line containing the entry for "server_name" and enter your servers
  name.  If you are unsure you can find out by using the hostname
  command, or by cat'ing /etc/hosts.

  Once you've made the neccesary changes you'll need to either copy the
  conf file to the bin directory, or start icecast with the -c option
  and specify the location, like so:

  [dj@megajukebox bin]$ ./icecast -c ../etc/icecast.conf

  If everything has been configured correctly, you should see something
  similar to the following:

  [dj@megajukebox bin]$ ./icecast -c ../etc/icecast.conf -d /home/dj/mp3/icecast/
  Icecast Version 1.3.0 Starting...
  Icecast comes with NO WARRANTY, to the extent permitted by law.
  You may redistribute copies of Icecast under the terms of the
  GNU General Public License.
  For more information about these matters, see the file named COPYING.

  [05/Jan/2000:18:36:30] Icecast Version 1.3.0 Starting..
  [05/Jan/2000:18:36:30] Using stdin as icecast operator console
  [05/Jan/2000:18:36:30] Tailing file to icecast operator console
  [05/Jan/2000:18:36:30] Server started...
  [05/Jan/2000:18:36:30] Listening on port 8000...
  [05/Jan/2000:18:36:30] Using [megajukebox] as servername...
  [05/Jan/2000:18:36:30] Max values: 1000 clients, 1000 clients per source, 10 sources, 5 admins
  -> [05/Jan/2000:18:36:30] [Bandwidth: 0.000000MB/s] [Sources: 0] [Clients: 0] [Admins: 1] [Uptime: 0 seconds]

  The -d option sets the directory for log files and templates.

  Below is the list of command-line options:

          -c [filename]

  Parse as a configuration file. Please note that any command line
  parameters you supply after this override whatever is in file. Also note that
  icecast.conf in the current directory is already parsed when you specify
  this file, so anything in icecast.conf not overridden by the new configuration
  file will be used by the server.

          -P [port]

  This is the port used for all client, source, and admin connections. It's set
  to 8000 by default.

          -m [max clients]

  Allow this number of client connections. When this number is reached, all
  client connections will be refused with 'HTTP/1.0 504 Server Full'

          -p [encoder password]

  This sets the password that the encoder must use to be allowed to stream
  to the server. Note that if you have compiled the server with crypt()
  support, this argument must be an encrypted string.


  This will send the icecast server into the background (i.e daemon process).
  To use the admin commands now, you have to connect to the server as an
  admin, using some sort of telnet client.

          -d [directory]

  Make all log files created by icecast, and all templates that icecast looks
  for be relative to this directory.

  So, thats the server started, but you now need to connect an MP3
  source to the server.

  You can choose from two applications which deliver MP3 data to the
  server, Shout and LiveIce.

  11.1.1.  Shout

  Shout provides Icecast with a static playlist of MP3's to stream and
  is included with Icecast.

  You create the playlist if the MP3 files you want to stream with the

  find [MP3 directory] -name '*.mp3' -print > playlist

  At it's most basic level, to start the shout service, issue the

  [dj@megajukebox bin]# ./shout megajukebox -P hackme -p playlist

  The -P option specifies the password needed to add a mount-point to
  Icecast, this is the aptly set as hackme..... I strongly suggest you
  change it otherwise someone may :)  The -p option specifies the
  location of the playlist file.  Below is a list of all of the command
  line options:

  [dj@megajukebox bin]# ./shout
  Usage: shout <host> [options] [[-b <bitrate] file.mp3]...
          -B <directory>  - Use directory for all shout's files.
          -C <file>       - Use file as configuration file
          -D <dj_file>    - Run this before every song (system())
          -P <password>   - Use specified password
          -S              - Display all settings and exit
          -V              - Use verbose output
          -X <desc>       - Use specified description.
          -a              - Turn on automatic bitrate (transfer) correction
          -b <bitrate>    - Start using specified bitrate
          -d              - Activate the dj.
          -e <port>       - Connect to port on server.
          -f              - Skip files that don't match the specified bitrate
          -g <genre>      - Use specified genre
          -h              - Show this text
          -i              - Use old icy headers
          -k              - Don't truncate the internal playlist (continue)
          -l              - Go on forever (loop)
          -m <mount>      - Use specified mount point
          -n <name>       - Use specified name
          -o              - Turn of the bitrate autodetection.
          -p <playlist>   - Use specified file as a playlist
          -r              - Shuffle playlist (random play)
          -s              - (Secret) Don't send meta data to the directory server
          -u <url>        - Use specified url
          -v              - Show version
          -x              - Don't update the cue file (saves cpu)
          -z              - Go into the background (Daemon mode)
          -t              - Enable title streaming

  11.1.2.  LiveIce

  LiveIce can work in 2 modes, it can pass a playlist to Icecast or can
  pass live audio from the soundcard.

  After untaring and reading the README concerning building the package,
  make sure you have mpg123 installed and available as LiveIce requires

  There are two ways of configuring LiveIce editing the config file with
  vi/emacs/or whatever or by using the TK based configuration tool,
  which is a pretty way of editing it :)

  The best place for describing the internals of liveice.cfg can be
  found at LiveIce's homepage where Scott covers all of the options.
  This is a copy of my config file with LiveIce set to mixer mode
  (stream from a list of MP3's)

  NOTE:  I've added comments to the file, so if you cut and paste make
  sure the comments haven't wrapped around to a new line otherwise
  LiveIce will not work :)

  # liveice configuration file
  # Automatically generated

  SERVER megajukebox              # Your server name * MUST BE THE NAME THE SERVER RESOLVES TO *
  PORT 8000                       # The port Icecast is running on

  NAME Megajukebox                # Information regarding the name of your server which is sent to MP3 players, and
                                  # to directory servers.
                                  # Examples 'Sarah FM' or 'ThisTown: Loud and Heavy Jazz - Internet Radio 24/7'

  GENRE Live                      # Information regarding the genre.  Examples 'Talk' or 'Dance'

  DESCRIPTION                     # Information regarding the station.  Example 'The best for reggae in the North'

  URL http://megajukebox:8000     # The URL and port of the server.

  PUBLIC 0                        # Set this to 1 if you want Icecast to announce your station and list it's details
                                  # on a directory server, otherwise leave 0

  XAUDIOCAST_LOGIN                # can be either ICY_LOGIN or X_AUDIOCAST_LOGIN.  X_AUDIOCAST is better.

  MOUNTPOINT /techno              # Sets the mountpoint name of the stream for Icecast.  Only used if X_AUDIOCAST is used
                                  # otherwise defaults to icy_0

  PASSWORD hackme                 # Icecast's admin password

  SAMPLE_RATE 44100               # The sample rate of the stream
  STEREO                          # Can be MONO or STEREO

  NO_SOUNDCARD                    # See below

  HALF_DUPLEX                     # Sets the soundcard duplex mode.  Can be HALF_DUPLEX or FULL_DUPLEX
  USE_GOGO                        # Sets the encoder to use.  Check the README for the list
  BITRATE 128000                  # Sets the bit rate of the stream (see below)
  VBR_QUALITY 1                   # Sets the variable bit rate quality.

  MIXER                           # See below

  PLAYLIST /megajukebox/playlist  # Location of the playlist (see details on the find command later in this chapter)

  TRACK_LOGFILE track.log         # Filename and location to dump list of MP3's streamed

  Once you have your config file you start LiveIce like so:

  [dj@megajukebox liveice]$ ./liveice
  opening connection to megajukebox 8000
  Attempting to Contact Server
  connection successful: forking process
  opening pipe!...
  writing password
  Setting up Interface
  Soundcard Reopened For Encoding
  Input Format: 16Bit 44100Hz Stereo
  Output Format: 256000 Bps Mpeg Audio
  IceCast Server: megajukebox:8000
  Mountpoint: /techno
  Name: megajukebox - this and that radio - broadcasting 24/7
  Genre: Techno
  Description: a load of digital noise -> but i know you like it :)

   Press '+' to Finish
  adding /megajukebox/demotunes/track_1.mp3
  adding /megajukebox/demotunes/track_2.mp3
  adding /megajukebox/demotunes/track_3.mp3
  adding /megajukebox/demotunes/track_4.mp3
  Adding New Channel 1
  Adding New Channel 2
  Channel 1 selecting
  Channel 2 selecting
  Playing track_1.mp3
  searching for Id3v2
  searching for Id3v1
  copying the data
  fixing the nulls
  adding the url
  closing input file
  Using log track.log

  The last line is a peak meter.

  These are the keyboard controls for mixer mode:

  Action                          Channel 1 Key   Channel 2 Key
  ~~~~~~                          ~~~~~~~~~~~~~   ~~~~~~~~~~~~~
  Select next track on channel    1               a
  Select prev track on channel    q               z
  Start/Stop channel              2               s
  Reset channel                   w               x
  Increase volume on channel      3               d
  Decrease volume on channel      e               c
  Increase speed on channel       4               f
  Decrease speed on channel       r               v
  Sticky mode On/Random/Off       5               g
  Preview channel                 t               b
  Random Track                    u               m

  The above liveice.cfg is for mixermode.  To use LiveIce in audio mode
  change the line relating to MIXER to NOMIXER and set NO_SOUNDCARD to
  SOUNDCARD and restart LiveIce.

  Forgetting to set the correct options will lead to some interesting
  warning ;)

     946:Error: Line In mode *and* no soundcard??????? Eeejit!

  Once you have it all correctly set up and have plugged in an external
  source, you should be able to stream =:)

  [dj@megajukebox liveice]$ ./liveice
  Initialising Soundcard
  16Bit 22050Hz Stereo Full Duplex
  opening connection to megajukebox 8000
  Attempting to Contact Server
  connection successful: forking process
  opening pipe!...
  writing password
  Setting up Interface
  Soundcard Reopened For Encoding
  Input Format: 16Bit 22050Hz Stereo
  Output Format: 32000 Bps Mpeg Audio
  IceCast Server: megajukebox:8000
  Mountpoint: /daves_band_live_at_the_club
  Name: megajukebox - Dave and the Dynamite - Live at the Roxy
  Genre: Live/Rock
  Url: http://megajukebox
  Description: megajukebox::Louder than a frog in a trashcan..... and almost as musical

   Press '+' to Finish
  Lvl: L:   8704 R:  11776

  The last line is a signal level meter, if the input signal is too high
  you will get a *clip* warning.  If you do, turn down the gain of the
  input source.

  The keen eyed amongst you may of noticed that in liveice.cfg the first
  comment lines point out that the file is automatically generated.  If
  you are using the TK based GUI and you've made
  manual changes, you will lose them when you save.  Either use the GUI
  or learn vi/emacs :)

  11.2.  Fluid

  After untaring the bundle cd to the directory, then read the README :)

  Fluid has three basic modes of operation, transmit, relay and forward.
  I'll only focus on transmit.

  The config files associated for transmit are located in
  config/MP3TX.cfg.  To test the server run with the following, at this
  point the default config settings should be ok:
  java Fluid TX

  Naturally enough you'll need Java of some form installed first.  You
  can use either the Blackdown port of JDK available from
  <> or if you are using Redhat, Kaffe.

  Fluid comes with a few sample MP3 files, so if everything is working
  you should see something similar to this (I've started the server
  using Kaffe in this example, you may have to start it using java):

  [dj@megajukebox Fluid-Beta2J]$ kaffe Fluid tx
  ------- Fluid Streaming Server Beta 2 -------
  This program is ShareWare(tm) and it will not
  be crippled in any way because of it. However
  if you do like the program and will use it
  commercial purposes, we ask of you to contact
  us at the address below for pricing info:

   Eldean AB                  E-mail:
   Sjoangsvagen 7   
   S-192 72 Sollentuna

     Fluid is Copyright Subside (C) 1998
         written by Lars Samuelsson

  * Transmission mode *
  Reading config from: config/MP3TX.cfg
  Reading playlist: playlist.m3u
  Server started on port: 2711
  Accepting administrator login on port: 2710
  P| Dr. Nick - Hello Everybody

  If you get this far, it looks like things are working, but I'm sure
  you'll want to stream more than the demo files!

  You'll need to compile a playlist of the MP3's you want to stream.
  This will be a static list users will not be able to alter this list
  or make requests.  This playlist is named playlist.m3u and is located
  by default in the root directory.

  To compile a playlist of all MP3's in a particular directory (or disk)
  use the following command:

  find [MP3 directory] -name "*.mp3" -print > playlist.m3u

  By default the server uses port 2711, which is where your listeners
  will connect to, if you need to change this this can be done in the
  config file.

  The server can be remotely administered by telneting to it's admin
  port, by default port 2710 like so:

  [dj@megajukebox Fluid-Beta2J]$ telnet localhost 2710
  Connected to localhost.localdomain.
  Escape character is '^]'.
  You are connected to the -Fluid- Streaming Server
  Type "help" for a command reference
  The following commands are available:
   help conn curr exit
  Information about the currently broadcasted song:
  Title:   Beer Talk
  Artist:  Homer Simpson
  Album:   The Simpsons
  Year:    1996
  Comment: Borrowed this as an example
  Genre:   Comedy

  The reference to "jaguar" is the admin password, this is the default.
  There is no prompt for the password so please don't sit there waiting
  for one! I suggest that you change the password from the default oth­
  erwise you will invite a hack! This can be changed in the config file,
  which looks like this:

  [dj@megajukebox config]$ cat MP3TX.cfg

  # --- The lines are ---
  # 1. PORT number (the server will use)
  # 2. PORT number (for maintaining the server remotely)
  # 3. Maximum number of connections (the server will accept)
  # 4  Packetsize when reading/sending (in bytes)
  # 5. Bitrate of the mp3s in kbit/s (all mp3s must have same bitrate)
  # 6. Delay between songs (in milliseconds)
  # 7. Password for remote administration
  # 8. Playlist name (list in .m3u format)
  # 9. Name of the file to write song info to (from ID3-tag)

  The reference to the playlist being in m3u format means that it is in
  the same format as produced by the find command mentioned earlier.

  11.2.1.  Update for RC1

  Fluid has been updated, and released under a GPL License (Way to go

  Here are the main differences:

  ·  A new format fluid.config

  ·  A new default port 4711

  ·  A new method of starting and stopping.  Fluid now used the scripts
     fluid.start and fluid.stop

  ·  Source code is now part of the dist. Yippee!

  11.3.  Litestream

  Litestream is similar in it's operation to Icecast.  The system is
  composed of the streamer and the source.

  To start the streamer you issue a command in the following format:

  [dj@megajukebox]$ litestream <source port> <stream host> <stream port> <max listeners> <log ident> [<yp host> <yp port>]

  The options are as follows:

  source port Is the port you connect the 'audio' source to.

  stream host This is the IP or FQDN of your server (find this out with
  the hostname command for domain name or cat /etc/hosts for your IP

  stream port This is the port number clients connect to.

  max listeners This is the number of clients who can connect to your

  log ident Log messages are written to /var/log/messages.  <log ident>
  is a ident string used to identify the Litestream server, eg.

  If you use the ident string 'Litestream' the server starting message
  in /var/log/messages would look similar to:

  Sep 18 19:32:20 linux Litestream[1901]: stream.c:555: main: 'server started'

  <yp host & yp port> These 2 options specify the host and port of an yp
  server, used to announce your server to mp3 directory servers.

  Here is an example:

  [dj@megajukebox]$ litestream 5000 megajukebox 5555 1000 LitestreamServer

  Ok, so that's the server system.  Once started it sit's there waiting
  for an input to be connected, which is done with the source command.

  The format is like so:

  source <ip or hostname> <port> <name> <genre> <url> <irc> <icq> <aim> <public? (0, 1)> <reported bitrate (16, 18, 56, 128, etc.)> <playlist.txt> <log ident>

  Below is the list of options for source.

  ip or hostname is the hostname or IP address of the computer running
  Litestream Server.

  port is the port on which Litestream Server is waiting for a streaming

  name is the name of your stream.

  genre is the genre of your stream.

  url is the URL for your streams web page.

  irc is the IRC channel for your stream.

  icq is your ICQ ID.

  aim is your AIM ID.

  public is either 0 or 1.  0 for private, 1 for public.

  reported bitrate is the stream bitrate which will be reported to the
  YP server.  This does not necessarily reflect the actual bitrate of
  your stream.

  playlist.txt is the name of the file containing your playlist.

  log ident is the syslog identifier which will be prepended to each log
  message generated by Litestream Streaming Source.

  Here is an example which would match the Litestream command earlier:

  [dj@megajukebox]$ source megakukebox 5000 'Megajukebox  The best jukebox ever' Various \#megajukebox 0 N/A 1 128 /jukebox/playlist LitestreamSource

  11.4.  Apache::MP3 Module

  The Apache:MP3 module allows the Apache WebServer to stream MP3's.  It
  does not offer the same level of features as Fluid, Litestream &

  The module is dependant on MP3::Info and mod_perl, both of which are
  available on CPAN.

  After downloading the files needed a read of the README will help
  Firstly make the module, then make the config changes to Apache's
  config files (your changes may be made in either httpd.conf or

  You will also have to create directories for icons and a directory
  hierarchy for MP3's.

  Section 6 of the README details the editing (or creation) of a Perl
  config file.

  This is to preload the MP3::Info module which without this may cause
  Apache to crash.

  In httpd.conf create an entry like so:

   Perlrequire /etc/httpd/conf/startup.perl

  Alter the /etc/httpd/conf reference to point to where your Apache
  config files are located on your system.

  Now create the file startup.perl at the location specified above.  At
  the very minimum you should have the following (again adjust the path
  of perl to fit your system).

  [dj@megajukebox conf]$ cat /etc/httpd/conf/startup.perl


  use MP3::Info();

  Then restart Apache and point a web browser at one of your new MP3
  directories :)

  There is more documentation available on Apache::MP3, from the
  command-line run the following:

  [dj@megajukebox conf]$ perldoc Apache::MP3

  11.5.  Firewall Bypassing

  A large portion of companies and institutions now use firewalls to
  block users from connecting to certain ports on remote servers.

  A way to bypass this is to run your MP3 server using port 80. You will
  not be able to run a webserver on the same URL or IP address.

  When running Icecast in 'firewall bypass' mode you will need to start
  shout with the option to set the port to port 80, like so:

  [dj@megajukebox bin]$ ./shout megajukebox -e 80 -P hackme -p ../playlist

  11.6.  Bandwidth considerations

  Streaming audio can consume vast quantities of bandwidth if the MP3
  servers' bit-rate is set too high.

  Consider this scenario.  A T1 link has a capacity of approx. 1.55
  Mb/Sec.  If you stream your MP3's at 128K/Bps stereo, each connecting
  player will use 256K/Bps, so only 6 users could connect to your MP3
  server at any time without problems. And at 256K/Bps, you will not get
  too many modem users connecting!

  So you must make a decision at what to set your stream rates not only
  on what your server's internet connection is rated at, but what your
  users will be connecting at.  24K/Bps Stereo will give a reasonable
  quality signal that 56K modem users will be able to connect to, and
  for the same T1 line would allow approx. 32 simultaneous connections.

  If your server is running on an Intranet, bandwidth issues will still
  have to be considered especially if your network is running 10M/Sec.

  But please let either your ISP or sys admin know you are going to
  stream otherwise you may be in for a shock. Some ISP's will charge you
  for bandwidth over a certain limit and sys admins like to know why
  their network is now running slow :)

  11.7.  Copyright Issues

  I think it's reasonable to assume that record companies will not like
  you streaming material without their permission or payment of some
  kind! So what can you stream?

  This is an area where you will need to be aware of the legal
  ramifications, because it will be you who will be liable.

  Below are two links, one for the Electronic Frontier Foundation who
  are advocates of freeing restrictions surrounding the technology.  The
  other link is to the Recording Industry Association of America, which
  seeks to protect the rights of artists from piracy.

  I strongly suggest visiting both of the sites, and any others relevant
  to where you are physically based.



  12.  Listening to MP3's.

  So, hopefully, you should now have some MP3 files ready to listen to,
  and have the choice of playing from file or stream.

  12.1.  Playing from File

  Playing from file is reasonably straight-forward with all players. The
  only big difference is some are command-line based and some are X

  Playing an MP3 file from file requires you to pass the mp3 file as a
  parameter, like so:

  [dj@megajukebox]$ mpg123 /mp3_files/SampleFile.mp3


  [dj@megajukebox]$ xaudio /mp3_files/SampleFile.mp3

  If you want to play a series of files, pass them in as a list:

  [dj@megajukebox]$ alsaplayer /mp3_files/SampleFile1.mp3 /mp3_files/SampleFile2.mp3

  To play all the tracks in a directory, just wildcard the file
  selected, like so:

  [dj@megajukebox]$ xmms /mp3_files/*.mp3

  12.2.  Playing from MP3 Streams

  Playing from a MP3 stream is quite easy, just replace the file with
  the streams url and port number:

  [dj@megajukebox]$ mpg123 http://localhost:8000


  [dj@megajukebox]$ freeamp http://megajukebox:2711

  12.3.  Capturing MP3 Streams

  A stream from a MP3 server can be captured and save locally using the

  12.3.1.  Streamripper

  To capture an MP3 stream using streamripper, type the following:

  [dj@megajukebox]$ streamripper -h megajukebox -p 8000
  Press CTRL-C to stop
  name: Megajukebox
  reponse: 200
  genre: Megajukebox
  bitrate: 128
  This stream contains no meta data, ripping as one large ass track
  1st track ripped
  socket error: : Success

  Here's a list of options.

  [dj@megajukebox]$ streamripper
  Usage: streamripper -h <host> -p <port> [-d dir]
          -h <host>      - Required: The host name
          -p <port>      - Required: The host port, typically 8000
          -d <dir>       - Optional: The destination directory
          -l <logfile>   - Optional: Output everything to a log file <logfile>
          -v             - Optional: Be extra verbose
          -c             - Optional: Don't display status counter
          -q             - Optional: Quiet mode, don't make mp3 files
          -s             - Optional: Output stream to stdout for mpg123
          -n             - Optional: Wait until next track before writing
          -o             - Optional: Create a directory with the stream name for output

  12.3.2.  wget

  To save a stream using wget:

  [dj@megajukebox]$ wget http://megajukebox:8000 -O download01.mp3
  --13:41:41--  http://megajukebox:8000/
             => `download01.mp3'
  Connecting to megajukebox:8000... connected!
  HTTP request sent, awaiting response... 200 OK
  Length: unspecified [audio/mpeg]

      0K -> .......... .......... .......... .......... ..........
     50K -> .......... .......... .......... .......... ..........
    100K -> .......... .......... .......... .......... ..........
    150K -> .......... .......... .......... .......... ..........
    200K -> .......... .......... .......... .......... ..........
    250K -> .......... .......... .......... .......... ..........
    300K -> .......... .......... .......... .......... ..........
    350K -> .......... .......... .......... .......... ..........
    400K -> .......... .......... .......... .......... ..........
    450K -> .......... .......... .......... .......... ..........
    500K -> .......... .......... .......... .......... ..........

  12.4.  Mixing

  12.4.1.  eMixer

  eMixer gives you the ability to mix MP3's in a similar manner to a
  DJ's mixing desk.

  Newer versions support 2 sound cards so you can output your mix on one
  card and monitor or cue the next track on another.

  As usual once untaring, read the readme on how to build the package.

  You will need to create a playlist of MP3 files, do this with the find
  command mentioned in the Streaming section.

  You will need mpg123 installed before you can run eMixer.

  Here are the control keys (taken from the readme)

  "up, down"              scroll thru playlist
  "page up, page down"    scroll thru playlist screen full at a time
  "enter"                 start/stop track
  "tab"                   change channel
  "}","]"                 toggle between volume and speed controls/windows
  "space"                 restart active track
  "left, right"           fader controls
  "insert"                decrease volume/speed in channel one
  "home"                  increase volume/speed in channel one
  "delete"                decrease volume/speed in channel two
  "end"                   increase volume/speed in channel two
  "< , / , >"             left, center & right positions of fader
  " + , = "       (NEW)   switch between faders
  " q "                   start/stop channel channel one
  " w "                   start/stop channel channel two
  " p "                   toggle between playmodes - single, loop, continuous, random
  " a "                   stop all channels
  " f "                   file menu
  " u "                   util menu
  " h "                   help menu
  " ~, ` "                cancel menu drop down
  " s "                   turn SIM Play on
                          (SIM Play starts the same track in both channels
                          simultaneously )

  12.4.2.  GDAM

  GDAM is a GUI based MP3 mixer with a large bank of built in, real-time

  13.  ID3 Editing.

  ID3 tags are information fields contained within the MP3 data that
  contain information about the MP3 such as who the artist is, what
  album the track came from, and so on.

  Editors allow you to view and edit this information.

  13.1.  id3ed

  [dj@megajukebox]$ id3ed
  id3ed v1.10 - mpeg layer 3 file information editor
  Usage: id3ed [-s songname] [-n artist] [-a album] [-y year] [-c comment]
               [-k tracknum] [-g genre] [-q] [-SNAYCKG] [-l/-L] [-r]
               [-i] <mp3files> [-v]

    -q                    no line interface; only set tags specified on command
                          line. Use twice to suppress all output except errors.
    -SNAYCKG              prompt to edit the specified tags only
                          Other tags can still be set with the -[snaycg] options.
    -l/-L                 display list of genres
    -r                    remove id3 tag from files
    -i                    show current id3 tag only, don't edit
    -v                    output program version/license

  [dj@megajukebox MyBand]$ id3ed track01.mp3

  File track01.mp3: (tag v1.1)
  songname[max:30]: Our Kick Ass Demo
  artist[max:30]: Us
  album[max:30]: White Album
  year[max:4]: 1999
  comment[max:28]: Will be a classic some day!
  tracknum[max:3]: 1
  genre[0-255/name]: 5

  If we now want to see the Id info we run id3ed with the -i flag, like

  [dj@megajukebox MyBand]$ id3ed -i track01.mp3
  track01.mp3: (tag v1.1)
  songname: Our Kick Ass Demo
  artist: Us
  album: White Album
  year: 1999
  comment: Will be a classic some day!
  tracknum: 1
  genre: Funk(5)

  13.2.  mp3info

  MP3info works in a similar manner to id3ed, here is a snippet from
  it's man pages:

         mp3info [ -option < value > ] filename[s]...

         This is a small tool to get and set MP3 information.

         -T     Do  NOT  get  TAG  Info.  Use on slowseeking media,
                since TAG is at the very end of the file.

         -s <0/1>
                Print info. Pretty  much  obsolete,  as  it  always
                should be on.  Defaults 1.

         -f <format>
                Format of printed string. See README for details.

         -F <Formatnumber>
                Predefined format of printed string. Try it out.

         -w     Write  TAG Info. Will write a TAG at the end of the
         -W     Wipe TAG Info. Will wipe clean TAG Info.  Overrides
                all  other  TAG  Options, and turns -s off. To both
                wipe tag and display info, use -W -s 1

         -n <name>
                Set song title. Implies -w. Will go away soon,  use
                -t instead.

         -t <name>
                Set song title. Alias for -n. Implies -w.

         -a <artist>
                Set name of artist. Implies -w.

         -l <album>
                Set album name. Implies -w.

         -y <year>
                Sets published year. Implies -w.

         -c <comment>
                Sets comment. Implies -w.

         -g <genre>
                Set genre number. Implies -w.

         -G <genrename>
                Set genre by name. Implies -w.

  13.3.  ID3 Genre codes

  ID3 uses a code to refer to the genre,  here is the code listing [info
  taken from <>].

  The following genres are defined in ID3v1

        1.Classic Rock
       10.New Age
       22.Death Metal
       37.Sound Clip
       46.Instrumental Pop
       47.Instrumental Rock
       56.Southern Rock
       60.Top 40
       61.Christian Rap
       64.Native American
       66.New Wave
       73.Acid Punk
       74.Acid Jazz
       78.Rock & Roll
       79.Hard Rock

     The following genres are Winamp extensions

       82.National Folk
       84.Fast Fusion
       91.Gothic Rock
       92.Progressive Rock
       93.Psychedelic Rock
       94.Symphonic Rock
       95.Slow Rock
       96.Big Band
       98.Easy Listening
      104.Chamber Music
      107.Booty Bass
      109.Porn Groove
      111.Slow Jam
      117.Power Ballad
      118.Rhythmic Soul
      121.Punk Rock
      122.Drum Solo
      123.A capella
      125.Dance Hall

  14.  MP3 to Minidisc.

  By Afra <>

  To record MP3s from Linux to your minidisc, there are two methods:
  analog and digital. Each method is dependant on the type of soundcard
  your Linux box has. The main difference between analog and digital
  recordings is the sound quality you achieve during recording.  With an
  analog recording you may hear some static noise/fuzz, although some
  analog soundcards try to minimize this. Don't think that just because
  you have an analog sound card that you are doomed, because you are
  not. This just means you will have to spend a little time trying to
  minimize the noise by fiddling with mixer settings.

  No special software is required, just a regular MP3 player for your
  Linux box. Basically both methods will require a connection from your
  soundcard's output to your Minidic's input.

  14.1.  Analog Recording

  If your soundcard does not have a "Digital Out" connection, you may
  will have to go via the analogue recording route.

  To record an MP3 from your Linux box onto your Minidisc, connect an
  analog chord from your sound card's Line Out to your Minidisc's Line
  In input.  Set your Minidisc in the recording mode as you usually
  would and play the MP3. You should see the recording levels on your
  Minidisc rise. Upon playback, you may notice a small or large amount
  of static noise, which is due to your soundcard.

  To minimize the static noise during recording, keep your Minidisc's
  recording/volume levels within the -3dB and 0dB range.  To help clear
  up the noise you may hear connect up your Minidisc as described above
  and turn on the Recording Mode but do not play an MP3. You should
  still hear the noise and this may help in fiddling with your mixer

  14.2.  Digital Recording

  You will experience better sound quality with digital recording.
  Connect your optical cable (if you have to buy one, the sizes of the
  MD and card port matter, usually 3.5mm but check your soundcard and
  Minidisc user manual) to your sound card's Digital Out.

  The cable should now be glowing on the other end - good. Now, connect
  the glowing end to your minidisc Digital Input. Set your minidisc in
  record mode and play the mp3 from the computer.

  14.3.  Playlists

  There is one problem when playing MP3 tracklistings: how will your
  Minidisc know when to set an automark when a track jumps onto another
  (ie. change track numbers automatically)? The solution is this MP3
  file which you can download at

  You may use this file, which is an MP3 file containing 2 seconds of
  silence, to insert before each MP3 on your playlist, so the MD will
  know when to automark.

  15.  Feedback.

  New hardware and software is being released all the time.   If you are
  using newer versions of the hardware and / or software listed within
  this document, or can add to anything within this area, please send
  your information for inclusion to and I'll include it
  in the next release.

  While I'd like to reply to every question, please note that on
  occasions I will not be able to reply quickly due to work.

  Happy MP3'ing!!

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