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doc/user: add sections for bit-perfect playback and DSD

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Max Kellermann 2014-09-02 14:42:05 +02:00
parent 97b816200d
commit 68bdfa9d0e
1 changed files with 173 additions and 4 deletions
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doc/user.xml
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@ -680,7 +680,7 @@ systemctl start mpd.socket</programlisting>
<section id="config_audio_format">
<title>Audio Format Settings</title>
<section>
<section id="config_global_audio_format">
<title>Global Audio Format</title>
<para>
@ -1143,6 +1143,175 @@ systemctl start mpd.socket</programlisting>
</section>
</chapter>
<chapter id="advanced_usage">
<title>Advanced usage</title>
<section id="bit_perfect">
<title>Bit-perfect playback</title>
<para>
"Bit-perfect playback" is a phrase used by audiophiles to
describe a setup that plays back digital music as-is, without
applying any modifications such as resampling, format
conversion or software volume. Naturally, this implies a
lossless codec.
</para>
<para>
By default, <application>MPD</application> attempts to do
bit-perfect playback, unless you tell it not to. Precondition
is a sound chip that supports the audio format of your music
files. If the audio format is not supported,
<application>MPD</application> attempts to fall back to the
nearest supported audio format, trying to lose as little
quality as possible.
</para>
<para>
To verify if <application>MPD</application> converts the audio
format, enable verbose logging, and watch for these lines:
</para>
<programlisting>decoder: audio_format=44100:24:2, seekable=true
output: opened plugin=alsa name="An ALSA output" audio_format=44100:16:2
output: converting from 44100:24:2</programlisting>
<para>
This example shows that a 24 bit file is being played, but the
sond chip cannot play 24 bit. It falls back to 16 bit,
discarding 8 bit.
</para>
<para>
However, this does not yet prove bit-perfect playback;
<application>ALSA</application> may be fooling
<application>MPD</application> that the audio format is
supported. To verify the format really being sent to the
physical sound chip, try:
</para>
<programlisting>cat /proc/asound/card*/pcm*p/sub*/hw_params
access: RW_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 44100 (44100/1)
period_size: 4096
buffer_size: 16384</programlisting>
<para>
Obey the "format" row, which indicates that the current
playback format is 16 bit (signed 16 bit integer, little
endian).
</para>
<para>
Check list for bit-perfect playback:
</para>
<itemizedlist>
<listitem>
<para>
Use the <link linkend="alsa_output">ALSA</link> output
plugin.
</para>
</listitem>
<listitem>
<para>
Disable sound processing inside
<application>ALSA</application> by configuring a
"hardware" device (<parameter>hw:0,0</parameter> or
similar).
</para>
</listitem>
<listitem>
<para>
Don't use software volume (setting <link
linkend="config_audio_outputs"><varname>mixer_type</varname></link>).
</para>
</listitem>
<listitem>
<para>
Don't force <application>MPD</application> to use a
specific audio format (settings <link
linkend="config_audio_outputs"><varname>format</varname></link>,
<link
linkend="config_global_audio_format"><varname>audio_output_format</varname></link>).
</para>
</listitem>
<listitem>
<para>
Verify that you are really doing bit-perfect playback
using <application>MPD</application>'s verbose log and
<filename>/proc/asound/card*/pcm*p/sub*/hw_params</filename>.
Some DACs can also indicate the audio format.
</para>
</listitem>
</itemizedlist>
</section>
<section id="dsd">
<title>Direct Stream Digital (DSD)</title>
<para>
DSD (<ulink
url="https://en.wikipedia.org/wiki/Direct_Stream_Digital">Direct
Stream Digital</ulink>) is a digital format that stores audio
as a sequence of single-bit values at a very high sampling
rate.
</para>
<para>
<application>MPD</application> understands the file formats
<link linkend="dsdiff_decoder"><filename>dff</filename></link>
and <link
linkend="dsf_decoder"><filename>dsf</filename></link>. There
are three ways to play back DSD:
</para>
<itemizedlist>
<listitem>
<para>
Native DSD playback. Requires
<application>ALSA</application> 1.0.27.1 or later, a sound
driver/chip that supports DSD and of course a DAC that
supports DSD.
</para>
</listitem>
<listitem>
<para>
DoP (DSD over PCM) playback. This wraps DSD inside fake
24 bit PCM according to the <ulink
url="http://dsd-guide.com/dop-open-standard">DoP
standard</ulink>. Requires a DAC that supports DSD. No
support from ALSA and the sound chip required (except for
24 bit PCM support).
</para>
</listitem>
<listitem>
<para>
Convert DSD to PCM on-the-fly.
</para>
</listitem>
</itemizedlist>
<para>
Native DSD playback is used automatically if available. DoP
is only used if enabled explicitly using the <link
linkend="alsa_output"><varname>dop</varname></link> option,
because there is no way for <application>MPD</application> to
find out whether the DAC supports it. DSD to PCM conversion
is the fallback if DSD cannot be used directly.
</para>
</section>
</chapter>
<chapter id="plugin_reference">
<title>Plugin reference</title>
@ -1527,7 +1696,7 @@ systemctl start mpd.socket</programlisting>
<section id="decoder_plugins">
<title>Decoder plugins</title>
<section>
<section id="dsdiff_decoder">
<title><varname>dsdiff</varname></title>
<para>
@ -1558,7 +1727,7 @@ systemctl start mpd.socket</programlisting>
</informaltable>
</section>
<section>
<section id="dsf_decoder">
<title><varname>dsf</varname></title>
<para>
@ -2041,7 +2210,7 @@ systemctl start mpd.socket</programlisting>
If set to <parameter>yes</parameter>, then DSD over
PCM according to the <ulink
url="http://dsd-guide.com/dop-open-standard">DoP
standard others</ulink> is enabled. This wraps DSD
standard</ulink> is enabled. This wraps DSD
samples in fake 24 bit PCM, and is understood by
some DSD capable products, but may be harmful to
other hardware. Therefore, the default is