481 lines
14 KiB
XML
481 lines
14 KiB
XML
<?xml version='1.0' encoding="utf-8"?>
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<!DOCTYPE book PUBLIC "-//OASIS//DTD DocBook XML V4.2//EN"
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"docbook/dtd/xml/4.2/docbookx.dtd">
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<book>
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<title>The Music Player Daemon - User's Manual</title>
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<chapter>
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<title>Introduction</title>
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<para>
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</para>
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</chapter>
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<chapter>
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<title>Installation</title>
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<para>
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We recommend that you use the software installation routines of
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your distribution to install MPD. Most operating systems have a
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MPD package, which is very easy to install.
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</para>
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<section>
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<title>Installing on Debian/Ubuntu</title>
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<para>
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Install the package <filename>mpd</filename>:
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</para>
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<programlisting>apt-get install mpd</programlisting>
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</section>
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<section>
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<title>Compiling from source</title>
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<para>
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Download the source tarball from <ulink
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url="http://mpd.wikia.com/wiki/Server">the MPD home
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page</ulink> and unpack it:
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</para>
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<programlisting>tar xjf mpd-0.14.2.tar.bz
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cd mpd-0.14.2</programlisting>
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<para>
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Make sure that all the required libraries and build tools are
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installed. The <filename>INSTALL</filename> file has a list.
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</para>
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<para>
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Now configure the source tree:
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</para>
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<programlisting>./configure</programlisting>
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<para>
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The <parameter>--help</parameter> argument shows a list of
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compile-time options. When everything is ready and
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configured, compile:
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</para>
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<programlisting>make</programlisting>
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<para>
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And install:
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</para>
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<programlisting>make install</programlisting>
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</section>
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</chapter>
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<chapter>
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<title>Configuration</title>
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<section>
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<title>Configuring the music directory</title>
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<para>
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When you play local files, you should organize them within a
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directory called the "music directory". This is configured in
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MPD with the <varname>music_directory</varname> setting.
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</para>
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<para>
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By default, MPD follows symbolic links in the music directory.
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This behavior can be switched off:
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<varname>follow_outside_symlinks</varname> controls whether
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MPD follows links pointing to files outside of the music
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directory, and <varname>follow_inside_symlinks</varname> lets
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you disable symlinks to files inside the music directory.
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</para>
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</section>
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<section>
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<title>Configuring audio outputs</title>
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<para>
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Audio outputs are devices which actually play the audio chunks
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produced by MPD. You can configure any number of audio output
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devices, but there must be at least one. If none is
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configured, MPD attempts to auto-detect. Usually, this works
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quite well with ALSA, OSS and on Mac OS X.
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</para>
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<para>
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To configure an audio output manually, add an
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<varname>audio_output</varname> block to
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<filename>mpd.conf</filename>:
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</para>
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<programlisting>audio_output {
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type "alsa"
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name "my ALSA device"
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device "hw:0"
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}
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</programlisting>
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</section>
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</chapter>
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<chapter>
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<title>Plugin reference</title>
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<section>
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<title>Output plugins</title>
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<section>
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<title><varname>alsa</varname></title>
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<para>
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The "Advanced Linux Sound Architecture" plugin uses
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<filename>libasound</filename>. It is recommended if you
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are using Linux.
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</para>
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<informaltable>
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<tgroup cols="2">
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<thead>
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<row>
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<entry>Setting</entry>
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<entry>Description</entry>
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</row>
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</thead>
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<tbody>
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<row>
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<entry>
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<varname>device</varname>
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<parameter>NAME</parameter>
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</entry>
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<entry>
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Sets the device which should be used. This can be
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any valid ALSA device name. The default value is
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"default", which makes
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<filename>libasound</filename> choose a device. It
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is recommended to use a "hw" or "plughw" device,
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because otherwise, <filename>libasound</filename>
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automatically enables "dmix", which has major
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disadvantages (fixed sample rate, poor resampler,
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...).
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</entry>
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</row>
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<row>
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<entry>
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<varname>use_mmap</varname>
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<parameter>yes|no</parameter>
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</entry>
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<entry>
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If set to <parameter>yes</parameter>, then
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<filename>libasound</filename> will try to use
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memory mapped I/O.
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</entry>
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</row>
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<row>
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<entry>
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<varname>buffer_time</varname>
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<parameter>US</parameter>
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</entry>
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<entry>
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Sets the device's buffer time in microseconds.
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Don't change unless you know what you're doing.
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</entry>
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</row>
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<row>
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<entry>
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<varname>period_time</varname>
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<parameter>US</parameter>
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</entry>
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<entry>
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Sets the device's period time in microseconds.
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Don't change unless you really know what you're
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doing.
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</entry>
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</row>
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<row>
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<entry>
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<varname>auto_resample</varname>
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<parameter>yes|no</parameter>
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</entry>
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<entry>
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If set to <parameter>no</parameter>, then
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<filename>libasound</filename> will not attempt to
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resample, handing the responsibility over to MPD.
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It is recommended to let MPD resample (with
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libsamplerate), because ALSA is quite poor at doing
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so.
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</entry>
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</row>
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<row>
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<entry>
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<varname>auto_channels</varname>
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<parameter>yes|no</parameter>
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</entry>
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<entry>
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If set to <parameter>no</parameter>, then
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<filename>libasound</filename> will not attempt to
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convert between different channel numbers.
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</entry>
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</row>
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<row>
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<entry>
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<varname>auto_format</varname>
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<parameter>yes|no</parameter>
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</entry>
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<entry>
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If set to <parameter>no</parameter>, then
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<filename>libasound</filename> will not attempt to
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convert between different sample formats (16 bit, 24
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bit, floating point, ...).
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</entry>
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</row>
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</tbody>
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</tgroup>
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</informaltable>
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</section>
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<section>
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<title><varname>ao</varname></title>
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<para>
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The <varname>ao</varname> plugin uses the portable
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<filename>libao</filename> library.
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</para>
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</section>
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<section>
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<title><varname>fifo</varname></title>
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<para>
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The <varname>fifo</varname> plugin writes raw PCM data to a
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FIFO (First In, First Out) file. The data can be read by
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another program.
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</para>
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</section>
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<section>
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<title><varname>jack</varname></title>
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<para>
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The <varname>jack</varname> plugin connects to a JACK
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server.
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</para>
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</section>
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<section>
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<title><varname>mvp</varname></title>
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<para>
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The <varname>mvp</varname> plugin uses the proprietary
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Hauppauge Media MVP interface. We do not know any user of
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this plugin, and we do not know if it actually works.
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</para>
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</section>
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<section>
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<title><varname>httpd</varname></title>
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<para>
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The <varname>httpd</varname> plugin creates a HTTP server,
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similar to ShoutCast / IceCast. HTTP streaming clients like
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<filename>mplayer</filename> can connect to it.
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</para>
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<para>
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You must configure either <varname>quality</varname> or
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<varname>bitrate</varname>. It is highly recommended to
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configure a fixed <varname>format</varname>, because a
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stream cannot switch its audio format on-the-fly when the
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song changes.
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</para>
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<informaltable>
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<tgroup cols="2">
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<thead>
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<row>
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<entry>Setting</entry>
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<entry>Description</entry>
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</row>
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</thead>
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<tbody>
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<row>
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<entry>
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<varname>port</varname>
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<parameter>P</parameter>
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</entry>
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<entry>
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Binds the HTTP server to the specified port (on all
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interfaces).
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</entry>
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</row>
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<row>
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<entry>
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<varname>encoder</varname>
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<parameter>NAME</parameter>
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</entry>
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<entry>
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Chooses an encoder plugin,
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e.g. <parameter>vorbis</parameter>.
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</entry>
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</row>
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<row>
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<entry>
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<varname>quality</varname>
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<parameter>Q</parameter>
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</entry>
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<entry>
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Configures the encoder quality (for VBR) in the
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range -1 .. 10.
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</entry>
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</row>
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<row>
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<entry>
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<varname>bitrate</varname>
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<parameter>BR</parameter>
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</entry>
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<entry>
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Sets a constant encoder bit rate, in kilobit per
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second.
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</entry>
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</row>
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</tbody>
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</tgroup>
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</informaltable>
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</section>
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<section>
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<title><varname>null</varname></title>
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<para>
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The <varname>null</varname> plugin does nothing. It
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discards everything sent to it.
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</para>
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<informaltable>
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<tgroup cols="2">
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<thead>
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<row>
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<entry>Setting</entry>
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<entry>Description</entry>
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</row>
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</thead>
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<tbody>
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<row>
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<entry>
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<varname>sync</varname>
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<parameter>yes|no</parameter>
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</entry>
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<entry>
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If set to <parameter>no</parameter>, then the timer
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is disabled - the device will accept PCM chunks at
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arbitrary rate (useful for benchmarking). The
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default behaviour is to play in real time.
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</entry>
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</row>
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</tbody>
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</tgroup>
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</informaltable>
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</section>
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<section>
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<title><varname>oss</varname></title>
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<para>
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The "Open Sound System" plugin is supported on most Unix
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platforms.
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</para>
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<informaltable>
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<tgroup cols="2">
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<thead>
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<row>
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<entry>Setting</entry>
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<entry>Description</entry>
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</row>
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</thead>
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<tbody>
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<row>
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<entry>
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<varname>device</varname>
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<parameter>PATH</parameter>
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</entry>
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<entry>
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Sets the path of the PCM device. If not specified,
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then MPD will attempt to open
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<filename>/dev/sound/dsp</filename> and
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<filename>/dev/dsp</filename>.
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</entry>
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</row>
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</tbody>
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</tgroup>
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</informaltable>
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</section>
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<section>
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<title><varname>osx</varname></title>
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<para>
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The "Mac OS X" plugin uses Apple's CoreAudio API.
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</para>
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</section>
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<section>
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<title><varname>pipe</varname></title>
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<para>
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The <varname>pipe</varname> plugin starts a program and
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writes raw PCM data into its standard input.
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</para>
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</section>
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<section>
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<title><varname>pulse</varname></title>
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<para>
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The <varname>pulse</varname> plugin connects to a PulseAudio
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server.
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</para>
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</section>
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<section>
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<title><varname>shout</varname></title>
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<para>
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The <varname>shout</varname> plugin connects to a ShoutCast
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or IceCast server. It forwards tags to this server.
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</para>
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</section>
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<section>
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<title><varname>solaris</varname></title>
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<para>
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The "Solaris" plugin runs only on SUN Solaris, and plays via
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<filename>/dev/audio</filename>.
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</para>
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<informaltable>
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<tgroup cols="2">
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<thead>
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<row>
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<entry>Setting</entry>
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<entry>Description</entry>
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</row>
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</thead>
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<tbody>
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<row>
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<entry>
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<varname>device</varname>
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<parameter>PATH</parameter>
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</entry>
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<entry>
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Sets the path of the audio device, defaults to
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<filename>/dev/audio</filename>.
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</entry>
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</row>
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</tbody>
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</tgroup>
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</informaltable>
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</section>
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</section>
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</chapter>
|
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</book>
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