path: root/doc/user.xml

<?xml version='1.0' encoding="utf-8"?>
<!DOCTYPE book PUBLIC "-//OASIS//DTD DocBook XML V4.2//EN"
               "docbook/dtd/xml/4.2/docbookx.dtd">
<book>
  <title>The Music Player Daemon - User's Manual</title>

  <chapter>
    <title>Introduction</title>

    <para>
    </para>
  </chapter>

  <chapter>
    <title>Installation</title>

    <para>
      We recommend that you use the software installation routines of
      your distribution to install MPD.  Most operating systems have a
      MPD package, which is very easy to install.
    </para>

    <section>
      <title>Installing on Debian/Ubuntu</title>

      <para>
        Install the package <filename>mpd</filename>:
      </para>

      <programlisting>apt-get install mpd</programlisting>
    </section>

    <section>
      <title>Compiling from source</title>

      <para>
        Download the source tarball from <ulink
        url="http://mpd.wikia.com/wiki/Server">the MPD home
        page</ulink> and unpack it:
      </para>

      <programlisting>tar xjf mpd-0.14.2.tar.bz
cd mpd-0.14.2</programlisting>

      <para>
        Make sure that all the required libraries and build tools are
        installed.  The <filename>INSTALL</filename> file has a list.
      </para>

      <para>
        Now configure the source tree:
      </para>

      <programlisting>./configure</programlisting>

      <para>
        The <parameter>--help</parameter> argument shows a list of
        compile-time options.  When everything is ready and
        configured, compile:
      </para>

      <programlisting>make</programlisting>

      <para>
        And install:
      </para>

      <programlisting>make install</programlisting>
    </section>
  </chapter>

  <chapter>
    <title>Configuration</title>

    <section>
      <title>Configuring the music directory</title>

      <para>
        When you play local files, you should organize them within a
        directory called the "music directory".  This is configured in
        MPD with the <varname>music_directory</varname> setting.
      </para>

      <para>
        By default, MPD follows symbolic links in the music directory.
        This behavior can be switched off:
        <varname>follow_outside_symlinks</varname> controls whether
        MPD follows links pointing to files outside of the music
        directory, and <varname>follow_inside_symlinks</varname> lets
        you disable symlinks to files inside the music directory.
      </para>
    </section>

    <section>
      <title>Configuring audio outputs</title>

      <para>
        Audio outputs are devices which actually play the audio chunks
        produced by MPD.  You can configure any number of audio output
        devices, but there must be at least one.  If none is
        configured, MPD attempts to auto-detect.  Usually, this works
        quite well with ALSA, OSS and on Mac OS X.
      </para>

      <para>
        To configure an audio output manually, add an
        <varname>audio_output</varname> block to
        <filename>mpd.conf</filename>:
      </para>

      <programlisting>audio_output {
    type "alsa"
    name "my ALSA device"
    device "hw:0"
}
</programlisting>
    </section>
  </chapter>

  <chapter>
    <title>Plugin reference</title>

    <section>
      <title>Output plugins</title>

      <section>
        <title><varname>alsa</varname></title>

        <para>
          The "Advanced Linux Sound Architecture" plugin uses
          <filename>libasound</filename>.  It is recommended if you
          are using Linux.
        </para>

        <informaltable>
          <tgroup cols="2">
            <thead>
              <row>
                <entry>Setting</entry>
                <entry>Description</entry>
              </row>
            </thead>
            <tbody>
              <row>
                <entry>
                  <varname>device</varname>
                  <parameter>NAME</parameter>
                </entry>
                <entry>
                  Sets the device which should be used.  This can be
                  any valid ALSA device name.  The default value is
                  "default", which makes
                  <filename>libasound</filename> choose a device.  It
                  is recommended to use a "hw" or "plughw" device,
                  because otherwise, <filename>libasound</filename>
                  automatically enables "dmix", which has major
                  disadvantages (fixed sample rate, poor resampler,
                  ...).
                </entry>
              </row>
              <row>
                <entry>
                  <varname>use_mmap</varname>
                  <parameter>yes|no</parameter>
                </entry>
                <entry>
                  If set to <parameter>yes</parameter>, then
                  <filename>libasound</filename> will try to use
                  memory mapped I/O.
                </entry>
              </row>
              <row>
                <entry>
                  <varname>buffer_time</varname>
                  <parameter>US</parameter>
                </entry>
                <entry>
                  Sets the device's buffer time in microseconds.
                  Don't change unless you know what you're doing.
                </entry>
              </row>
              <row>
                <entry>
                  <varname>period_time</varname>
                  <parameter>US</parameter>
                </entry>
                <entry>
                  Sets the device's period time in microseconds.
                  Don't change unless you really know what you're
                  doing.
                </entry>
              </row>
              <row>
                <entry>
                  <varname>auto_resample</varname>
                  <parameter>yes|no</parameter>
                </entry>
                <entry>
                  If set to <parameter>no</parameter>, then
                  <filename>libasound</filename> will not attempt to
                  resample, handing the responsibility over to MPD.
                  It is recommended to let MPD resample (with
                  libsamplerate), because ALSA is quite poor at doing
                  so.
                </entry>
              </row>
              <row>
                <entry>
                  <varname>auto_channels</varname>
                  <parameter>yes|no</parameter>
                </entry>
                <entry>
                  If set to <parameter>no</parameter>, then
                  <filename>libasound</filename> will not attempt to
                  convert between different channel numbers.
                </entry>
              </row>
              <row>
                <entry>
                  <varname>auto_format</varname>
                  <parameter>yes|no</parameter>
                </entry>
                <entry>
                  If set to <parameter>no</parameter>, then
                  <filename>libasound</filename> will not attempt to
                  convert between different sample formats (16 bit, 24
                  bit, floating point, ...).
                </entry>
              </row>
            </tbody>
          </tgroup>
        </informaltable>
      </section>

      <section>
        <title><varname>ao</varname></title>

        <para>
          The <varname>ao</varname> plugin uses the portable
          <filename>libao</filename> library.
        </para>
      </section>

      <section>
        <title><varname>fifo</varname></title>

        <para>
          The <varname>fifo</varname> plugin writes raw PCM data to a
          FIFO (First In, First Out) file.  The data can be read by
          another program.
        </para>
      </section>

      <section>
        <title><varname>jack</varname></title>

        <para>
          The <varname>jack</varname> plugin connects to a JACK
          server.
        </para>
      </section>

      <section>
        <title><varname>mvp</varname></title>

        <para>
          The <varname>mvp</varname> plugin uses the proprietary
          Hauppauge Media MVP interface.  We do not know any user of
          this plugin, and we do not know if it actually works.
        </para>
      </section>

      <section>
        <title><varname>httpd</varname></title>

        <para>
          The <varname>httpd</varname> plugin creates a HTTP server,
          similar to ShoutCast / IceCast.  HTTP streaming clients like
          <filename>mplayer</filename> can connect to it.
        </para>

        <para>
          You must configure either <varname>quality</varname> or
          <varname>bitrate</varname>.  It is highly recommended to
          configure a fixed <varname>format</varname>, because a
          stream cannot switch its audio format on-the-fly when the
          song changes.
        </para>

        <informaltable>
          <tgroup cols="2">
            <thead>
              <row>
                <entry>Setting</entry>
                <entry>Description</entry>
              </row>
            </thead>
            <tbody>
              <row>
                <entry>
                  <varname>port</varname>
                  <parameter>P</parameter>
                </entry>
                <entry>
                  Binds the HTTP server to the specified port (on all
                  interfaces).
                </entry>
              </row>
              <row>
                <entry>
                  <varname>encoder</varname>
                  <parameter>NAME</parameter>
                </entry>
                <entry>
                  Chooses an encoder plugin,
                  e.g. <parameter>vorbis</parameter>.
                </entry>
              </row>
              <row>
                <entry>
                  <varname>quality</varname>
                  <parameter>Q</parameter>
                </entry>
                <entry>
                  Configures the encoder quality (for VBR) in the
                  range -1 .. 10.
                </entry>
              </row>
              <row>
                <entry>
                  <varname>bitrate</varname>
                  <parameter>BR</parameter>
                </entry>
                <entry>
                  Sets a constant encoder bit rate, in kilobit per
                  second.
                </entry>
              </row>
            </tbody>
          </tgroup>
        </informaltable>
      </section>

      <section>
        <title><varname>null</varname></title>

        <para>
          The <varname>null</varname> plugin does nothing.  It
          discards everything sent to it.
        </para>

        <informaltable>
          <tgroup cols="2">
            <thead>
              <row>
                <entry>Setting</entry>
                <entry>Description</entry>
              </row>
            </thead>
            <tbody>
              <row>
                <entry>
                  <varname>sync</varname>
                  <parameter>yes|no</parameter>
                </entry>
                <entry>
                  If set to <parameter>no</parameter>, then the timer
                  is disabled - the device will accept PCM chunks at
                  arbitrary rate (useful for benchmarking).  The
                  default behaviour is to play in real time.
                </entry>
              </row>
            </tbody>
          </tgroup>
        </informaltable>
      </section>

      <section>
        <title><varname>oss</varname></title>

        <para>
          The "Open Sound System" plugin is supported on most Unix
          platforms.
        </para>

        <informaltable>
          <tgroup cols="2">
            <thead>
              <row>
                <entry>Setting</entry>
                <entry>Description</entry>
              </row>
            </thead>
            <tbody>
              <row>
                <entry>
                  <varname>device</varname>
                  <parameter>PATH</parameter>
                </entry>
                <entry>
                  Sets the path of the PCM device.  If not specified,
                  then MPD will attempt to open
                  <filename>/dev/sound/dsp</filename> and
                  <filename>/dev/dsp</filename>.
                </entry>
              </row>
            </tbody>
          </tgroup>
        </informaltable>
      </section>

      <section>
        <title><varname>osx</varname></title>

        <para>
          The "Mac OS X" plugin uses Apple's CoreAudio API.
        </para>
      </section>

      <section>
        <title><varname>pipe</varname></title>

        <para>
          The <varname>pipe</varname> plugin starts a program and
          writes raw PCM data into its standard input.
        </para>
      </section>

      <section>
        <title><varname>pulse</varname></title>

        <para>
          The <varname>pulse</varname> plugin connects to a PulseAudio
          server.
        </para>
      </section>

      <section>
        <title><varname>shout</varname></title>

        <para>
          The <varname>shout</varname> plugin connects to a ShoutCast
          or IceCast server.  It forwards tags to this server.
        </para>
      </section>

      <section>
        <title><varname>solaris</varname></title>

        <para>
          The "Solaris" plugin runs only on SUN Solaris, and plays via
          <filename>/dev/audio</filename>.
        </para>

        <informaltable>
          <tgroup cols="2">
            <thead>
              <row>
                <entry>Setting</entry>
                <entry>Description</entry>
              </row>
            </thead>
            <tbody>
              <row>
                <entry>
                  <varname>device</varname>
                  <parameter>PATH</parameter>
                </entry>
                <entry>
                  Sets the path of the audio device, defaults to
                  <filename>/dev/audio</filename>.
                </entry>
              </row>
            </tbody>
          </tgroup>
        </informaltable>
      </section>
    </section>
  </chapter>
</book>