doc: added "audio outputs" section to the user's manual
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Max Kellermann
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doc/user.xml
281
doc/user.xml
@ -72,14 +72,287 @@ cd mpd-0.14.2</programlisting>
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<chapter>
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<title>Configuration</title>
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<para>
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</para>
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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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<para>
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</para>
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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>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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</chapter>
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</book>
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