mirror of https://github.com/mpv-player/mpv
1151 lines
37 KiB
XML
1151 lines
37 KiB
XML
<?xml version="1.0" encoding="iso-8859-1"?>
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<!-- $Revision$ -->
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<sect2 id="audio-dev">
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<title>Audio output devices</title>
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<sect3 id="sync">
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<title>Audio/Video synchronisation</title>
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<para>
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<application>MPlayer</application>'s audio interface is called
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<emphasis>libao2</emphasis>. It currently contains these drivers:
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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><entry>Driver</entry><entry>Comment</entry></row>
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</thead>
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<tbody>
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<row><entry>oss</entry><entry>
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OSS (ioctl) driver (supports hardware AC3 passthrough)
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</entry></row>
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<row><entry>sdl</entry><entry>
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SDL driver (supports sound daemons like <emphasis role="bold">ESD</emphasis>
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and <emphasis role="bold">ARTS</emphasis>)
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</entry></row>
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<row><entry>nas</entry><entry>
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NAS (Network Audio System) driver
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</entry></row>
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<row><entry>alsa5</entry><entry>
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native ALSA 0.5 driver
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</entry></row>
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<row><entry>alsa9</entry><entry>
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native ALSA 0.9 driver (supports hardware AC3 passthrough)
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</entry></row>
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<row><entry>sun</entry><entry>
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SUN audio driver (<filename>/dev/audio</filename>) for BSD and Solaris8 users
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</entry></row>
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<row><entry>macosx</entry><entry>
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native MacOSX driver
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</entry></row>
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<row><entry>win32</entry><entry>
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native Win32 driver
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</entry></row>
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<row><entry>arts</entry><entry>
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native ARTS driver (mostly for KDE users)
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</entry></row>
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<row><entry>esd</entry><entry>
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native ESD driver (mostly for GNOME users)
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</entry></row>
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</tbody>
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</tgroup>
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</informaltable>
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<para>
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Linux sound card drivers have compatibility problems. This is because
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<application>MPlayer</application> relies on an in-built feature of
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<emphasis>properly</emphasis> coded sound drivers that enable them to
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maintain correct audio/video sync. Regrettably, some driver authors
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don't take the care to code this feature since it is not needed for
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playing MP3s or sound effects.
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</para>
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<para>
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Other media players like <ulink url="http://avifile.sourceforge.net">aviplay</ulink>
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or <ulink url="http://xine.sourceforge.net">xine</ulink> possibly work
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out-of-the-box with these drivers because they use "simple" methods
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with internal timing. Measuring showed that their methods are not as
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efficient as <application>MPlayer</application>'s.
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</para>
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<para>
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Using <application>MPlayer</application> with a properly written audio
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driver will never result in A/V desyncs related to the audio, except
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only with very badly created files (check the man page for workarounds).
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</para>
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<para>
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If you happen to have a bad audio driver, try the <option>-autosync</option>
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option, it should sort out your problems. See the man page for detailed
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information.
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</para>
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<itemizedlist>
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<title>Some notes:</title>
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<listitem><para>
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If you have an OSS driver, first try <option>-ao oss</option> (this is
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the default). If you experience glitches, halts or anything out of the
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ordinary, try <option>-ao sdl</option> (NOTE: you need to have SDL libraries
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and header files installed). The SDL audio driver helps in a lot of cases
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and also supports ESD (GNOME) and ARTS (KDE).
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</para></listitem>
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<listitem><para>
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If you have ALSA version 0.5, then you almost always have to use
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<option>-ao alsa5</option>, since ALSA 0.5 has buggy OSS emulation code,
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and will <emphasis role="bold">crash <application>MPlayer</application></emphasis>
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with a message like this:
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<screen>
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DEMUXER: Too many (945 in 8390980 bytes) video packets in the buffer!<!--
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--></screen>
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</para></listitem>
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<listitem><para>
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On Solaris, use the SUN audio driver with the <option>-ao sun</option> option,
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otherwise neither video nor audio will work.
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</para></listitem>
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<listitem><para>
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If the sound clicks when playing from CD-ROM, turn on IRQ unmasking, e.g.
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<command>hdparm -u1 /dev/cdrom</command> (<command>man hdparm</command>).
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This is generally beneficial and described in more detail in the
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<link linkend="drives">CD-ROM section</link>.
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</para></listitem>
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</itemizedlist>
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</sect3>
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<sect3 id="experiences">
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<title>Soundcard experiences, recommendations</title>
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<para>
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On Linux, a 2.4.x kernel is highly recommended. Kernel 2.2 is not tested.
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</para>
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<para>
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Linux sound drivers are primarily provided by the free version of OSS.
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These drivers have been superseded by <ulink url="http://www.alsa-project.org">ALSA</ulink>
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(Advanced Linux Sound Architecture) in the 2.5 development series. If
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your distribution does not already use ALSA you may wish to try their
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drivers if you experience sound problems. ALSA drivers are generally
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superior to OSS in compatibility, performance and features. But some
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sound cards are only supported by the commercial OSS drivers from
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<ulink url="http://www.opensound.com/">4Front Technologies</ulink>.
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They also support several non-Linux systems.
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</para>
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<informaltable>
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<tgroup cols="8">
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<colspec colname="c1"/>
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<colspec colname="c2"/>
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<colspec colname="c3"/>
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<colspec colname="c4"/>
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<colspec colname="c5"/>
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<colspec colname="c6"/>
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<colspec colname="c7"/>
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<colspec colname="c8"/>
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<spanspec spanname="driver" namest="c2" nameend="c5"/>
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<spanspec spanname="ossfree" namest="c2" nameend="c2"/>
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<spanspec spanname="alsa" namest="c3" nameend="c3"/>
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<spanspec spanname="osspro" namest="c4" nameend="c4"/>
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<spanspec spanname="others" namest="c5" nameend="c5"/>
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<thead>
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<row>
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<entry morerows="2" valign="middle">SOUND CARD</entry>
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<entry spanname="driver">DRIVER</entry>
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<entry morerows="2" valign="middle">Max kHz</entry>
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<entry morerows="2" valign="middle">Max Channels</entry>
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<entry morerows="2" valign="middle">Max Opens
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<footnote id="maxopens">
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<para>the number of applications that are able to use
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the device <emphasis>at the same time</emphasis>.</para>
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</footnote>
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</entry>
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</row>
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<row>
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<entry spanname="ossfree">OSS/Free</entry>
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<entry spanname="alsa">ALSA</entry>
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<entry spanname="osspro">OSS/Pro</entry>
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<entry spanname="others">other</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>VIA onboard (686/A/B, 8233, 8235)</entry>
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<entry>
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<ulink url="http://sourceforge.net/project/showfiles.php?group_id=3242&release_id=59602">via82cxxx_audio</ulink>
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</entry>
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<entry>snd-via82xx</entry>
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<entry></entry>
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<entry></entry>
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<entry>4-48 kHz or 48 kHz only, depending on the chipset</entry>
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<entry></entry>
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<entry></entry>
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</row>
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<row>
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<entry>Aureal Vortex 2</entry>
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<entry>none</entry>
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<entry>none</entry>
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<entry>OK</entry>
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<entry>
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<ulink url="http://aureal.sourceforge.net">Linux Aureal Drivers</ulink>
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<ulink url="http://makacs.poliod.hu/~pontscho/aureal/au88xx-1.1.3.tar.bz2">buffer size increased to 32k</ulink>
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</entry>
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<entry>48</entry>
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<entry>4.1</entry>
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<entry>5+</entry>
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</row>
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<row>
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<entry>SB Live!</entry>
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<entry>Analog OK, SP/DIF not working</entry>
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<entry>Both OK</entry>
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<entry>Both OK</entry>
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<entry>
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<ulink url="http://opensource.creative.com">Creative's OSS driver (SP/DIF support)</ulink>
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</entry>
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<entry>192</entry>
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<entry>4.0/5.1</entry>
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<entry>32</entry>
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</row>
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<row>
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<entry>SB 128 PCI (es1371)</entry>
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<entry>OK</entry>
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<entry>?</entry>
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<entry></entry>
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<entry></entry>
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<entry>48</entry>
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<entry>stereo</entry>
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<entry>2</entry>
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</row>
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<row>
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<entry>SB AWE 64</entry>
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<entry>max 44kHz</entry>
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<entry>48kHz sounds bad</entry>
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<entry></entry>
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<entry></entry>
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<entry>48</entry>
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<entry></entry>
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<entry></entry>
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</row>
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<row>
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<entry>GUS PnP</entry>
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<entry>none</entry>
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<entry>OK</entry>
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<entry>OK</entry>
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<entry></entry>
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<entry>48</entry>
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<entry></entry>
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<entry></entry>
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</row>
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<row>
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<entry>Gravis UltraSound ACE</entry>
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<entry></entry>
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<entry></entry>
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<entry></entry>
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<entry></entry>
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<entry></entry>
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<entry></entry>
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<entry></entry>
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</row>
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<row>
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<entry>Gravis UltraSound MAX</entry>
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<entry>OK</entry>
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<entry>OK (?)</entry>
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<entry></entry>
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<entry></entry>
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<entry>48</entry>
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<entry></entry>
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<entry></entry>
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</row>
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<row>
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<entry>ESS 688</entry>
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<entry>OK</entry>
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<entry>OK (?)</entry>
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<entry></entry>
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<entry></entry>
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<entry>48</entry>
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<entry></entry>
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<entry></entry>
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</row>
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<row>
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<entry>C-Media cards (CMI8338/8738)</entry>
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<entry>OK</entry>
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<entry>OK SP/DIF is supported with ALSA 0.9.x</entry>
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<entry>?</entry>
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<entry></entry>
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<entry>44</entry>
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<entry>stereo</entry>
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<entry>1</entry>
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</row>
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<row>
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<entry>Yamaha cards (*ymf*)</entry>
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<entry>not OK (?) (maybe <option>-ao sdl</option>)</entry>
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<entry>OK only with ALSA 0.5 with OSS emulation
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<emphasis role="bold">AND</emphasis> <option>-ao sdl</option> (!) (?)</entry>
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<entry></entry>
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<entry></entry>
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<entry></entry>
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<entry></entry>
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<entry></entry>
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</row>
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<row>
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<entry>Cards with envy24 chips (like Terratec EWS88MT)</entry>
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<entry>?</entry>
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<entry>?</entry>
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<entry>OK</entry>
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<entry></entry>
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<entry>?</entry>
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<entry></entry>
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<entry></entry>
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</row>
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<row>
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<entry>PC Speaker or DAC</entry>
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<entry>OK</entry>
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<entry>none</entry>
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<entry></entry>
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<entry>
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<ulink url="http://www.geocities.com/stssppnn/pcsp.html">Linux PC speaker OSS driver</ulink>
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</entry>
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<entry>The driver emulates 44.1, maybe more.</entry>
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<entry>mono</entry>
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<entry>1</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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<para>
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Feedback to this document is welcome. Please tell us how
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<application>MPlayer</application> and your sound card(s) worked together.
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</para>
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</sect3>
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<sect3 id="af">
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<title>Audio filters</title>
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<para>
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The old audio plugins have been superseded by a new audio filter layer. Audio
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filters are used for changing the properties of the audio data before the
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sound reaches the sound card. The activation and deactivation of the filters
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is normally automated but can be overridden. The filters are activated when
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the properties of the audio data differ from those required by the sound card
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and deactivated if unnecessary. The <option>-af filter1,filter2,...</option>
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option is used to override the automatic activation of filters or to insert
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filters that are not automatically inserted. The filters will be executed as
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they appear in the comma separated list.
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</para>
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<para>
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Example:
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<screen>mplayer -af resample,pan movie.avi</screen>
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would run the sound through the resampling filter followed by the pan filter.
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Observe that the list must not contain any spaces, else it will fail.
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</para>
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<para>
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The filters often have options that change their behavior. These options
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are explained in detail in the sections below. A filter will execute using
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default settings if its options are omitted. Here is an example of how to use
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filters in combination with filter specific options:
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<screen>mplayer -af resample=11025,pan=1:0.5:0.5 -channels 1 -srate 11025 media.avi</screen>
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would set the output frequency of the resample filter to 11025Hz and downmix
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the audio to 1 channel using the pan filter.
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</para>
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<para>
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The overall execution of the filter layer is controlled using the
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<option>-af-adv</option> option. This option has two suboptions:
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</para>
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<para>
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<option>force</option> is a bit field that controls how the filters
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are inserted and what speed/accuracy optimizations they use:
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</para>
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<variablelist>
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<varlistentry>
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<term><option>0</option></term>
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<listitem><para>
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Use automatic insertion of filters and optimize according to CPU speed.
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</para></listitem>
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</varlistentry>
|
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<varlistentry>
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<term><option>1</option></term>
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<listitem><para>
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Use automatic insertion of filters and optimize for the highest speed.
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<emphasis>Warning:</emphasis> Some features in the audio filters may
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silently fail, and the sound quality may drop.
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</para></listitem>
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</varlistentry>
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<varlistentry>
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<term><option>2</option></term>
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<listitem><para>
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Use automatic insertion of filters and optimize for quality.
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</para></listitem>
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</varlistentry>
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<varlistentry>
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<term><option>3</option></term>
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<listitem><para>
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Use no automatic insertion of filters and no optimization.
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<emphasis>Warning:</emphasis> It may be possible to crash MPlayer
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using this setting.
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</para></listitem>
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</varlistentry>
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<varlistentry>
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<term><option>4</option></term>
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<listitem><para>
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Use automatic insertion of filters according to 0 above,
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but use floating point processing when possible.
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</para></listitem>
|
|
</varlistentry>
|
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|
|
<varlistentry>
|
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<term><option>5</option></term>
|
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<listitem><para>
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Use automatic insertion of filters according to 1 above,
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|
but use floating point processing when possible.
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</para></listitem>
|
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</varlistentry>
|
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|
|
<varlistentry>
|
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<term><option>6</option></term>
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<listitem><para>
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Use automatic insertion of filters according to 2 above,
|
|
but use floating point processing when possible.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
|
|
<varlistentry>
|
|
<term><option>7</option></term>
|
|
<listitem><para>
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|
Use no automatic insertion of filters according to 3 above,
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|
and use floating point processing when possible.
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|
</para></listitem>
|
|
</varlistentry>
|
|
</variablelist>
|
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|
|
<para>
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|
<option>list</option> is an alias for the -af option.
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</para>
|
|
|
|
<para>
|
|
The filter layer is also affected by the following generic options:
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|
</para>
|
|
|
|
<variablelist>
|
|
<varlistentry>
|
|
<term><option>-v</option></term>
|
|
<listitem><para>
|
|
Increases the verbosity level and makes most filters print out extra
|
|
status messages.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
|
|
<varlistentry>
|
|
<term><option>-channels</option></term>
|
|
<listitem><para>
|
|
This option sets the number of output channels you would like your
|
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sound card to use. It also affects the number of channels that are
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|
being decoded from the media. If the media contains less channels
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|
than requested the channels filter (see below) will automatically
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|
be inserted. The routing will be the default routing for the channels
|
|
filter.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
|
|
<varlistentry>
|
|
<term><option>-srate</option></term>
|
|
<listitem><para>
|
|
This option selects the sample rate you would like your sound card
|
|
to use (of course the cards have limits on this). If the sample frequency
|
|
of your sound card is different from that of the current media, the resample
|
|
filter (see below) will be inserted into the audio filter layer to compensate
|
|
for the difference.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
<varlistentry>
|
|
<term><option>-format</option></term>
|
|
<listitem><para>
|
|
This option sets the sample format between the audio filter layer and the
|
|
sound card. If the requested sample format of your sound card is different
|
|
from that of the current media, a format filter (see below) will be inserted
|
|
to rectify the difference.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
</variablelist>
|
|
|
|
<sect4 id="af_resample">
|
|
<title>Up/Downsampling</title>
|
|
|
|
<para>
|
|
MPlayer fully supports sound up/down-sampling through the
|
|
<option>resample</option> filter. It can be used if you
|
|
have a fixed frequency sound card or if you are stuck with an old sound card
|
|
that is only capable of max 44.1kHz. This filter is automatically enabled if
|
|
it is necessary, but it can also be explicitly enabled on the command line. It
|
|
has three options:
|
|
</para>
|
|
|
|
<variablelist>
|
|
<varlistentry>
|
|
<term><option>srate <8000-192000></option></term>
|
|
<listitem><para>
|
|
is an integer used for setting the output sample
|
|
frequency in Hz. The valid range for this parameter is 8kHz to 192kHz. If
|
|
the input and output sample frequency are the same or if this parameter is
|
|
omitted the filter is automatically unloaded. A high sample frequency
|
|
normally improves the audio quality, especially when used in combination
|
|
with other filters.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
|
|
<varlistentry>
|
|
<term><option>sloppy</option></term>
|
|
<listitem><para>
|
|
is an optional binary parameter that allows the output frequency to differ
|
|
slightly from the frequency given by <option>srate</option>. This option
|
|
can be used if the startup of the playback is extremely slow. It is enabled
|
|
by default.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
|
|
<varlistentry>
|
|
<term><option>type <0-2></option></term>
|
|
<listitem><para>
|
|
is an optional integer between <literal>0</literal> and <literal>2</literal> that
|
|
selects which resampling method to use. Here <literal>0</literal> represents
|
|
linear interpolation as resampling method, <literal>1</literal> represents
|
|
resampling using a poly-phase filter-bank and integer processing and
|
|
<literal>2</literal> represents resampling using a poly-phase filter-bank and
|
|
floating point processing. Linear interpolation is extremely fast, but
|
|
suffers from poor sound quality especially when used for up-sampling. The
|
|
best quality is given by <literal>2</literal> but this method also suffers from
|
|
the highest CPU load.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
</variablelist>
|
|
|
|
<para>Example:
|
|
<screen>mplayer -af resample=44100:0:0</screen>
|
|
would set the output frequency of the resample filter to 44100Hz using exact output
|
|
frequency scaling and linear interpolation.
|
|
</para>
|
|
</sect4>
|
|
|
|
<sect4 id="af_channels">
|
|
<title>Changing the number of channels</title>
|
|
<para>
|
|
The <option>channels</option> filter can be used for adding and removing
|
|
channels, it can also be used for routing or copying channels. It is
|
|
automatically enabled when the output from the audio filter layer differs from
|
|
the input layer or when it is requested by another filter. This filter unloads
|
|
itself if not needed. The number of options is dynamic:
|
|
</para>
|
|
|
|
<variablelist>
|
|
<varlistentry>
|
|
<term><option>nch <1-6></option></term>
|
|
<listitem><para>
|
|
is an integer between <literal>1</literal> and <literal>6</literal> that is used
|
|
for setting the number of output channels. This option is required, leaving it
|
|
empty results in a runtime error.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
|
|
<varlistentry>
|
|
<term><option>nr <1-6></option></term>
|
|
<listitem><para>
|
|
is an integer between <literal>1</literal> and <literal>6</literal> that is used
|
|
for specifying the number of routes. This parameter is optional. If it is
|
|
omitted the default routing is used.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
|
|
<varlistentry>
|
|
<term><option>from1:to1:from2:to2:from3:to3...</option></term>
|
|
<listitem><para>
|
|
are pairs of numbers between <literal>0</literal> and <literal>5</literal>
|
|
that define where each channel should be routed.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
</variablelist>
|
|
|
|
<para>
|
|
If only <option>nch</option> is given the default routing is used, it works
|
|
as follows: If the number of output channels is bigger than the number of input
|
|
channels empty channels are inserted (except mixing from mono to stereo, then
|
|
the mono channel is repeated in both of the output channels). If the number of
|
|
output channels is smaller than the number of input channels the exceeding
|
|
channels are truncated.
|
|
</para>
|
|
|
|
<para>
|
|
Example 1:
|
|
<screen>mplayer -af channels=4:4:0:1:1:0:2:2:3:3 media.avi</screen>
|
|
would change the number of channels to 4 and set up 4 routes that swap
|
|
channel 0 and channel 1 and leave channel 2 and 3 intact. Observe that
|
|
if media containing two channels was played back, channels 2 and 3 would
|
|
contain silence but 0 and 1 would still be swapped.
|
|
</para>
|
|
|
|
<para>
|
|
Example 2:
|
|
<screen>mplayer -af channels=6:4:0:0:0:1:0:2:0:3 media.avi</screen>
|
|
would change the number of channels to 6 and set up 4 routes that copy
|
|
channel 0 to channels 0 to 3. Channel 4 and 5 will contain silence.
|
|
</para>
|
|
</sect4>
|
|
|
|
<sect4 id="af_format">
|
|
<title>Sample format converter</title>
|
|
<para>
|
|
The <option>format</option> filter converts between different sample formats. It
|
|
is automatically enabled when needed by the sound card or another filter.
|
|
</para>
|
|
|
|
<variablelist>
|
|
<varlistentry>
|
|
<term><option>bps <number></option></term>
|
|
<listitem><para>
|
|
can be <literal>1</literal>, <literal>2</literal> or <literal>4</literal> and
|
|
denotes the number of bytes per sample. This option is required, leaving it empty
|
|
results in a runtime error.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
|
|
<varlistentry>
|
|
<term><option>f <format></option></term>
|
|
<listitem><para>
|
|
is a text string describing the sample format. The string is a
|
|
concatenated mix of: <option>alaw</option>, <option>mulaw</option> or
|
|
<option>imaadpcm</option>, <option>float</option> or <option>int</option>,
|
|
<option>unsigned</option> or <option>signed</option>, <option>le</option> or
|
|
<option>be</option> (little or big endian). This option is required,
|
|
leaving it empty results in a runtime error.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
</variablelist>
|
|
|
|
<para>
|
|
Example:
|
|
<screen>mplayer -af format=4:float media.avi</screen>
|
|
would set the output format to 4 bytes per sample floating point data.
|
|
</para>
|
|
</sect4>
|
|
|
|
<sect4 id="af_delay">
|
|
<title>Delay</title>
|
|
<para>
|
|
The <option>delay</option> filter delays the sound to the loudspeakers such that
|
|
the sound from the different channels arrives at the listening position
|
|
simultaneously.
|
|
It is only useful if you have more than 2 loudspeakers. This filter has a
|
|
variable number of parameters:
|
|
</para>
|
|
|
|
<variablelist>
|
|
<varlistentry>
|
|
<term><option>d1:d2:d3...</option></term>
|
|
<listitem><para>
|
|
are floating point numbers representing the delays in ms that should be
|
|
imposed on the different channels. The minimum delay is 0ms and the maximum
|
|
is 1000ms.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
</variablelist>
|
|
|
|
<para>
|
|
To calculate the required delay for the different channels do as follows:
|
|
</para>
|
|
|
|
<orderedlist>
|
|
<listitem><para>
|
|
Measure the distance to the loudspeakers in meters in relation to your
|
|
listening position, giving you the distances s1 to s5 (for a 5.1 system).
|
|
There is no point in compensating for the sub-woofer (you will not hear the
|
|
difference anyway).
|
|
</para></listitem>
|
|
<listitem><para>
|
|
Subtract the distances s1 to s5 from the maximum distance i.e.
|
|
s[i] = max(s) - s[i]; i = 1...5
|
|
</para></listitem>
|
|
<listitem><para>
|
|
Calculate the required delays in ms as
|
|
d[i] = 1000*s[i]/342; i = 1...5
|
|
</para></listitem>
|
|
</orderedlist>
|
|
|
|
<para>
|
|
Example:
|
|
<screen>mplayer -af delay=10.5:10.5:0:0:7:0 media.avi</screen>
|
|
would delay front left and right by 10.5ms, the two rear channels and the sub
|
|
by 0ms and the center channel by 7ms.
|
|
</para>
|
|
|
|
</sect4>
|
|
|
|
<sect4 id="af_volume">
|
|
<title>Software volume control</title>
|
|
<para>Software volume control is implemented by the <option>volume</option>
|
|
audio filter. Use this filter with caution since it can reduce the signal to
|
|
noise ratio of the sound. In most cases it is best to set the level for the
|
|
PCM sound to max, leave this filter out and control the output level to your
|
|
speakers with the master volume control of the mixer. In case your sound card
|
|
has a digital PCM mixer instead of an analog one, and you hear distortion,
|
|
use the MASTER mixer instead. If there is an external amplifier connected to
|
|
the computer (this is almost always the case), the noise level can be minimized
|
|
by adjusting the master level and the volume knob on the amplifier until the
|
|
hissing noise in the background is gone. This filter has two options:
|
|
</para>
|
|
|
|
<variablelist>
|
|
<varlistentry>
|
|
<term><option>v <-200 - +60></option></term>
|
|
<listitem><para>
|
|
is a floating point number between <literal>-200</literal> and <literal>+60</literal>
|
|
which represents the volume level in dB. The default level is 0dB.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
|
|
<varlistentry>
|
|
<term><option>c</option></term>
|
|
<listitem><para>
|
|
is a binary control that turns soft clipping on and off. Soft-clipping can
|
|
make the sound more smooth if very high volume levels are used. Enable this
|
|
option if the dynamic range of the loudspeakers is very low. Be aware that
|
|
this feature creates distortion and should be considered a last resort.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
</variablelist>
|
|
|
|
<para>
|
|
Example:
|
|
<screen>mplayer -af volume=10.1:0 media.avi</screen>
|
|
would amplify the sound by 10.1dB and hard-clip if the sound level is too high.
|
|
</para>
|
|
|
|
<para>
|
|
This filter has a second feature: It measures the overall maximum sound level
|
|
and prints out that level when MPlayer exits. This volume estimate can be used
|
|
for setting the sound level in MEncoder such that the maximum dynamic range is
|
|
utilized.
|
|
</para>
|
|
</sect4>
|
|
|
|
<sect4 id="af_equalizer">
|
|
<title>Equalizer</title>
|
|
<para>
|
|
The <option>equalizer</option> filter represents a 10 octave band graphic
|
|
equalizer, implemented using 10 IIR band pass filters. This means that
|
|
it works regardless of what type of audio is being played back. The center
|
|
frequencies for the 10 bands are:
|
|
</para>
|
|
|
|
<informaltable>
|
|
<tgroup cols="2">
|
|
<thead>
|
|
<row>
|
|
<entry>Band No.</entry><entry>Center frequency</entry>
|
|
</row>
|
|
</thead>
|
|
<tbody>
|
|
<row><entry>0</entry><entry>31.25 Hz</entry></row>
|
|
<row><entry>1</entry><entry>62.50 Hz</entry></row>
|
|
<row><entry>2</entry><entry>125.0 Hz</entry></row>
|
|
<row><entry>3</entry><entry>250.0 Hz</entry></row>
|
|
<row><entry>4</entry><entry>500.0 Hz</entry></row>
|
|
<row><entry>5</entry><entry>1.000 kHz</entry></row>
|
|
<row><entry>6</entry><entry>2.000 kHz</entry></row>
|
|
<row><entry>7</entry><entry>4.000 kHz</entry></row>
|
|
<row><entry>8</entry><entry>8.000 kHz</entry></row>
|
|
<row><entry>9</entry><entry>16.00 kHz</entry></row>
|
|
</tbody>
|
|
</tgroup>
|
|
</informaltable>
|
|
|
|
<para>
|
|
If the sample rate of the sound being played back is lower than the center
|
|
frequency for a frequency band, then that band will be disabled. A known
|
|
bug with this filter is that the characteristics for the uppermost band
|
|
are not completely symmetric if the sample rate is close to the center
|
|
frequency of that band. This problem can be worked around by up-sampling
|
|
the sound using the resample filter before it reaches this filter.
|
|
</para>
|
|
|
|
<para>
|
|
This filter has 10 parameters:
|
|
</para>
|
|
|
|
<variablelist>
|
|
<varlistentry>
|
|
<term><option>g1:g2:g3...g10</option></term>
|
|
<listitem><para>
|
|
are floating point numbers between <literal>-12</literal> and <literal>+12</literal>
|
|
representing the gain in dB for each frequency band.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
</variablelist>
|
|
|
|
<para>
|
|
Example:
|
|
<screen>mplayer -af equalizer=11:11:10:5:0:-12:0:5:12:12 media.avi</screen>
|
|
would amplify the sound in the upper and lower frequency region while
|
|
canceling it almost completely around 1kHz.
|
|
</para>
|
|
</sect4>
|
|
|
|
<sect4 id="af_panning">
|
|
<title>Panning filter</title>
|
|
<para>
|
|
Use the <option>pan</option> filter to mix channels arbitrarily. It is
|
|
basically a combination of the volume control and the channels filter.
|
|
There are two major uses for this filter:
|
|
</para>
|
|
|
|
<orderedlist>
|
|
<listitem><para>
|
|
Down-mixing many channels to only a few, stereo to mono for example.
|
|
</para></listitem>
|
|
<listitem><para>
|
|
Varying the "width" of the center speaker in a surround sound system.
|
|
</para></listitem>
|
|
</orderedlist>
|
|
|
|
<para>
|
|
This filter is hard to use, and will require some tinkering before the
|
|
desired result is obtained. The number of options for this filter
|
|
depends on the number of output channels:
|
|
</para>
|
|
|
|
<variablelist>
|
|
<varlistentry>
|
|
<term><option>nch <1-6></option></term>
|
|
<listitem><para>
|
|
is an integer between <literal>1</literal> and <literal>6</literal> and is used
|
|
for setting the number of output channels. This option is required, leaving it
|
|
empty results in a runtime error.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
|
|
<varlistentry>
|
|
<term><option>l00:l01:l02:..l10:l11:l12:...ln0:ln1:ln2:...</option></term>
|
|
<listitem><para>
|
|
are floating point values between <literal>0</literal> and <literal>1</literal>.
|
|
<option>l[i][j]</option> determines how much of input channel j is mixed into
|
|
output channel i.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
</variablelist>
|
|
|
|
<para>
|
|
Example 1:
|
|
<screen>mplayer -af pan=1:0.5:0.5 -channels 1 media.avi</screen>
|
|
would down-mix from stereo to mono.
|
|
</para>
|
|
|
|
<para>
|
|
Example 2:
|
|
<screen>mplayer -af pan=3:1:0:1:0.5:0.5 -channels 3 media.avi</screen>
|
|
would give 3 channel output leaving channels 0 and 1 intact, and mix
|
|
channels 0 and 1 into output channel 2 (which could be sent to a
|
|
sub-woofer for example).
|
|
</para>
|
|
</sect4>
|
|
|
|
<sect4 id="af_sub">
|
|
<title>Sub-woofer</title>
|
|
<para>
|
|
The <option>sub</option> filter adds a sub woofer channel to the audio
|
|
stream. The audio data used for creating the sub-woofer channel is an
|
|
average of the sound in channel 0 and channel 1. The resulting sound is
|
|
then low-pass filtered by a 4th order Butterworth filter with a default
|
|
cutoff frequency of 60Hz and added to a separate channel in the audio
|
|
stream. Warning: Disable this filter when you are playing DVDs with Dolby
|
|
Digital 5.1 sound, otherwise this filter will disrupt the sound to the
|
|
sub-woofer. This filter has two parameters:
|
|
</para>
|
|
|
|
<variablelist>
|
|
<varlistentry>
|
|
<term><option>fc <20-300></option></term>
|
|
<listitem><para>
|
|
is an optional floating point number used for setting the cutoff frequency
|
|
for the filter in Hz. The valid range is 20Hz to 300Hz. For the best result
|
|
try setting the cutoff frequency as low as possible. This will improve the
|
|
stereo or surround sound experience. The default cutoff frequency is 60Hz.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
|
|
<varlistentry>
|
|
<term><option>ch <0-5></option></term>
|
|
<listitem><para>
|
|
is an optional integer between <literal>0</literal> and <literal>5</literal>
|
|
which determines the channel number in which to insert the sub-channel audio.
|
|
The default is channel number <literal>5</literal>. Observe that the number of
|
|
channels will automatically be increased to <replaceable>ch</replaceable> if
|
|
necessary.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
</variablelist>
|
|
|
|
<para>
|
|
Example:
|
|
<screen>mplayer -af sub=100:4 -channels 5 media.avi</screen>
|
|
would add a sub-woofer channel with a cutoff frequency of
|
|
100Hz to output channel 4.
|
|
</para>
|
|
</sect4>
|
|
|
|
<sect4 id="af_surround">
|
|
<title>Surround-sound decoder</title>
|
|
<para>
|
|
Matrix encoded surround sound can be decoded by the <option>surround</option>
|
|
filter. Dolby Surround is an example of a matrix encoded format. Many files
|
|
with 2 channel audio actually contain matrixed surround sound. To use this
|
|
feature you need a sound card supporting at least 4 channels. This filter has
|
|
one parameter:
|
|
</para>
|
|
|
|
<variablelist>
|
|
<varlistentry>
|
|
<term><option>d <0-1000></option></term>
|
|
<listitem><para>
|
|
is an optional floating point number between <literal>0</literal> and
|
|
<literal>1000</literal> used for setting the delay time in ms for the
|
|
rear speakers. This delay should be set as follows: if d1 is the distance
|
|
from the listening position to the front speakers and d2 is the distance
|
|
from the listening position to the rear speakers, then the delay d should
|
|
be set to 15ms if d1 <= d2 and to 15 + 5*(d1-d2) if d1 > d2. The default
|
|
value for d is 20ms.
|
|
</para></listitem>
|
|
</varlistentry>
|
|
</variablelist>
|
|
|
|
<para>
|
|
Example:
|
|
<screen>mplayer -af surround=15 -channels 4 media.avi</screen>
|
|
would add surround sound decoding with 15ms delay for the sound to the
|
|
rear speakers.
|
|
</para>
|
|
</sect4>
|
|
</sect3>
|
|
|
|
<sect3 id="audio-plugins">
|
|
<title>Audio plugins (deprecated)</title>
|
|
<note><para><emphasis role="bold">Audio plugins have been deprecated by audio filters and will
|
|
be removed soon.</emphasis></para></note>
|
|
|
|
<para>
|
|
<application>MPlayer</application> has support for audio plugins. Audio
|
|
plugins can be used for changing the properties of the audio data before
|
|
the sound reaches the sound card. They are enabled using the
|
|
<option>-aop</option> switch which takes a
|
|
<option>list=plugin1,plugin2,...</option> argument. The
|
|
<option>list</option> argument is required and determines which plugins
|
|
should be used and in which order they should be executed. Example:
|
|
|
|
<screen>mplayer media.avi -aop list=resample,format</screen>
|
|
|
|
would run the sound through the resampling plugin followed by the format
|
|
plugin.
|
|
</para>
|
|
|
|
<para>
|
|
The plugins can also have switches that change their behavior. These
|
|
switches are explained in detail in the sections below. A plugin will
|
|
execute using default settings if its switches are omitted. Here is an
|
|
example of how to use plugins in combination with plugin specific switches:
|
|
|
|
<screen>mplayer media.avi -aop list=resample,format:fout=44100:format=0x8</screen>
|
|
|
|
would set the output frequency of the resample plugin to 44100 Hz and the
|
|
output format of the format plugin to AFMT_U8.
|
|
</para>
|
|
|
|
<para>
|
|
Currently audio plugins can not be used in
|
|
<application>MEncoder</application>.
|
|
</para>
|
|
|
|
|
|
<sect4 id="updn-sampling">
|
|
<title>Up/Downsampling</title>
|
|
|
|
<para>
|
|
<application>MPlayer</application> fully supports up/downsampling of the sound. This plugin can be
|
|
used if you have a fixed frequency sound card or if you are stuck with an
|
|
old sound card that is only capable of max 44.1 kHz. Limitations in your
|
|
hardware are not auto detected, so you have to specify the sample frequency
|
|
explicitly. This plugin has one switch: <option>fout</option> which is used for setting the
|
|
desired output sample frequency. It defaults to 48 kHz, and is given in
|
|
Hz.
|
|
</para>
|
|
|
|
<para>
|
|
Usage:
|
|
|
|
<screen>
|
|
mplayer <replaceable>media.avi</replaceable> -aop list=resample:fout=<replaceable>freq</replaceable></screen>
|
|
|
|
where <literal><replaceable>freq</replaceable></literal> is the frequency
|
|
in Hz, like <literal>44100</literal>.
|
|
</para>
|
|
|
|
<note>
|
|
<para>
|
|
The output frequency should not be scaled up from the default value.
|
|
Scaling up will cause the audio and video streams to be played in slow
|
|
motion in addition to audio distortion.
|
|
</para>
|
|
</note>
|
|
</sect4>
|
|
|
|
|
|
<sect4 id="surround-decode">
|
|
<title>Surround Sound decoding</title>
|
|
|
|
<para>
|
|
<application>MPlayer</application> has an audio plugin that can decode
|
|
matrix encoded surround sound. Dolby Surround is an example of a matrix
|
|
encoded format. Many files with 2 channel audio actually contain matrixed
|
|
surround sound. To use this feature you need a sound card supporting at
|
|
least 4 channels.
|
|
</para>
|
|
|
|
<para>
|
|
Usage:
|
|
<screen>mplayer <replaceable>media.avi</replaceable> -aop list=surround</screen>
|
|
</para>
|
|
</sect4>
|
|
|
|
|
|
<sect4 id="sample-format-conv">
|
|
<title>Sample format converter</title>
|
|
|
|
<para>
|
|
If your sound card driver does not support signed 16-bit int data type,
|
|
this plugin can be used to change the format to one which your sound card
|
|
can understand. It has one switch, <option>format</option>, which can be
|
|
set to one of the numbers found in <filename>libao2/afmt.h</filename>. This
|
|
plugin is hardly ever needed and is intended for advanced users. Keep in
|
|
mind that this plugin only changes the sample format and not the sample
|
|
frequency or the number of channels.
|
|
</para>
|
|
|
|
<para>
|
|
Usage:
|
|
|
|
<screen>
|
|
mplayer <replaceable>media.avi</replaceable> -aop list=format:format=<replaceable>outfmt</replaceable></screen>
|
|
|
|
where <literal><replaceable>outfmt</replaceable></literal> is the
|
|
required output format.
|
|
</para>
|
|
</sect4>
|
|
|
|
|
|
<sect4 id="audio-delay">
|
|
<title>Delay</title>
|
|
<para>
|
|
This plugin delays the sound and is intended as an example of how to
|
|
develop new plugins. It can not be used for anything useful from a users
|
|
perspective and is mentioned here for the sake of completeness only. Do not
|
|
use this plugin unless you are a developer.
|
|
</para>
|
|
</sect4>
|
|
|
|
|
|
<sect4 id="sw-volume">
|
|
<title>Software volume control</title>
|
|
|
|
<para>
|
|
This plugin is a software replacement for the volume control, and can be
|
|
used on machines with a broken mixer device. It can also be used if one
|
|
wants to change the output volume of <application>MPlayer</application>
|
|
without changing the PCM volume setting in the mixer. It has one switch
|
|
<option>volume</option> that is used for setting the initial sound level.
|
|
The initial sound level can be set to values between 0 and 255 and defaults
|
|
to 101 which equals 0dB amplification. Use this plugin with caution since
|
|
it can reduce the signal to noise ratio of the sound. In most cases it is
|
|
best to set the level for the PCM sound to max, leave this plugin out and
|
|
control the output level to your speakers with the master volume control of
|
|
the mixer. If there is an external amplifier connected to the computer
|
|
(this is almost always the case), the noise level can be minimized by
|
|
adjusting the master level and the volume knob on the amplifier until the
|
|
hissing noise in the background is gone.
|
|
</para>
|
|
|
|
<para>
|
|
Usage:
|
|
<screen>
|
|
mplayer <replaceable>media.avi</replaceable> -aop list=volume:volume=<replaceable>0-255</replaceable></screen>
|
|
</para>
|
|
|
|
<para>
|
|
This plugin also has compressor or "soft-clipping" capabilities.
|
|
Compression can be used if the dynamic range of the sound is very high or
|
|
if the dynamic range of the loudspeakers is very low. Be aware that this
|
|
feature creates distortion and should be considered a last resort.
|
|
</para>
|
|
|
|
<para>
|
|
Usage:
|
|
<screen>
|
|
mplayer <replaceable>media.avi</replaceable> -aop list=volume:softclip</screen>
|
|
</para>
|
|
</sect4>
|
|
|
|
<sect4 id="extrastereo">
|
|
<title>Extrastereo</title>
|
|
|
|
<para>
|
|
This plugin (linearly) increases the difference between left and right
|
|
channels (like the <application>XMMS</application> extrastereo plugin) which gives some
|
|
sort of "live" effect to playback.
|
|
</para>
|
|
|
|
<para>
|
|
Usage:
|
|
<screen>
|
|
mplayer <replaceable>media.avi</replaceable> -aop list=extrastereo
|
|
mplayer <replaceable>media.avi</replaceable> -aop list=extrastereo:mul=3.45<!--
|
|
--></screen>
|
|
|
|
The default coefficient (<option>mul</option>) is a float number that
|
|
defaults to 2.5. If you set it to <literal>0.0</literal>, you will have
|
|
mono sound (average of both channels). If you set it to
|
|
<literal>1.0</literal>, sound will be unchanged, if you set it to
|
|
<literal>-1.0</literal>, left and right channels will be swapped.
|
|
</para>
|
|
</sect4>
|
|
|
|
|
|
<sect4 id="volnorm">
|
|
<title>Volume normalizer</title>
|
|
|
|
<para>
|
|
This plugin maximizes the volume without distorting the sound.
|
|
</para>
|
|
|
|
<para>
|
|
Usage:
|
|
<screen>mplayer <replaceable>media.avi</replaceable> -aop list=volnorm</screen>
|
|
</para>
|
|
</sect4>
|
|
</sect3>
|
|
</sect2>
|