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Commit 24bc277b authored by Max Kellermann's avatar Max Kellermann
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doc: added "audio outputs" section to the user's manual

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doc/user.xml
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......@@ -72,14 +72,287 @@ cd mpd-0.14.2</programlisting>
<chapter>
<title>Configuration</title>
<para>
</para>
<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>
<para>
</para>
<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>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>
</chapter>
</book>
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