mirror of https://github.com/mpv-player/mpv
733 lines
31 KiB
HTML
733 lines
31 KiB
HTML
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
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<HTML>
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<HEAD>
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<TITLE>Sound - MPlayer - The Movie Player for Linux</TITLE>
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<LINK REL="stylesheet" TYPE="text/css" HREF="default.css">
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<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-1">
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</HEAD>
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<BODY>
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<H3><A NAME="audio">2.3.2 Audio output devices</A></H3>
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<H4><A NAME="sync">2.3.2.1 Audio/Video synchronisation</A></H4>
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<P>MPlayer's audio interface is called <I>libao2</I>. It currently
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contains these drivers:</P>
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<DL>
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<DT>oss</DT>
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<DD>OSS (ioctl) driver (supports hardware AC3 passthrough)</DD>
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<DT>sdl</DT>
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<DD>SDL driver (supports <B>ESD</B>, <B>ARTS</B> etc)</DD>
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<DT>nas</DT>
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<DD>NAS (Network Audio System) driver</DD>
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<DT>alsa5</DT>
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<DD>native ALSA 0.5 driver</DD>
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<DT>alsa9</DT>
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<DD>native ALSA 0.9 driver (supports hardware AC3 passthrough)</DD>
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<DT>sun</DT>
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<DD>SUN audio driver (<CODE>/dev/audio</CODE>) for BSD and Solaris8 users</DD>
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<DT>arts</DT>
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<DD>native ARTS driver (mostly for KDE users)</DD>
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<DT>esd</DT>
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<DD>native ESD driver (mostly for GNOME users)</DD>
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</DL>
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<P>Fact is, Linux sound card drivers have compatibility problems. The cause
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is that MPlayer uses a feature that well coded audio drivers implement to
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maintain audio/video sync. Regrettably, some driver authors do not care about
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this function, it is not needed for playing MP3s or for sound effects.</P>
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<P>Other media players like aviplay or xine possibly work out-of-the-box with
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these drivers because they use "simple" methods with internal timing. A note:
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time showed their methods aren't AS efficient as MPlayer's.</P>
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<P>With a correctly written audio driver MPlayer will never create audio related
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A/V desynchronisation, unless your file is badly broken. Some options to work
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around these problems are described in the man page).</P>
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<P>If you happen to have a bad audio driver, try the <CODE>-autosync</CODE>
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option, it should sort out your problems. See the man page for detailed
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information.</P>
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<P>Some notes:</P>
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<UL>
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<LI>If you have an OSS driver, first try <CODE>-ao oss</CODE> (this is the
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default). If you experience glitches, halts or anything out of the
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ordinary, try <CODE>-ao sdl</CODE> (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 and ARTS. (ESD is the sound daemon
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from GNOME, ARTS is from KDE.)</LI>
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<LI>If you have ALSA version 0.5, then you almost always have to use
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<CODE>-ao alsa5</CODE> , since ALSA 0.5 has buggy OSS emulation code, and
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will <B>crash MPlayer</B> with a message like this:<BR>
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<CODE>DEMUXER: Too many (945 in 8390980 bytes) video packets in the buffer!</CODE></LI>
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</UL>
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<P>On <B>Solaris</B>, use the SUN audio driver with the
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<CODE>-ao sun</CODE> option, otherwise neither video nor audio will work.</P>
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<H4><A NAME="experiences">2.3.2.2 Sound Card experiences, recommendations</A></H4>
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<TABLE BORDER="0" WIDTH="100%">
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<TR><TD COLSPAN=3><B>VIA onboard chipset (via82cxxx) 48kHz only</B></TD></TR>
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<TR><TD></TD><TD>Driver:</TD><TD> from the
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<A HREF="http://sourceforge.net/project/showfiles.php?group_id=3242&release_id=59602">gkernel project</A></TD></TR>
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<TR><TD COLSPAN=3><B>Aureal Vortex 2</B></TD></TR>
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<TR><TD> </TD><TD>OSS:</TD><TD>no driver</TD></TR>
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<TR><TD></TD><TD>OSS/Pro:</TD><TD>OK</TD></TR>
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<TR><TD></TD><TD>ALSA:</TD><TD>no driver</TD></TR>
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<TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR>
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<TR><TD></TD><TD>Driver:</TD><TD><A HREF="http://aureal.sourceforge.net">aureal.sourceforge.net</A></TD></TR>
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<TR><TD></TD><TD>Driver2:</TD><TD> from <A HREF="http://makacs.poliod.hu/~pontscho/aureal/au88xx-1.1.3.tar.bz2">Pontscho's page</A>
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(<I>buffer size increased to 32k</I>)</TD></TR>
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<TR><TD COLSPAN=3><B>GUS PnP</B></TD></TR>
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<TR><TD></TD><TD>OSS:</TD><TD>no driver</TD></TR>
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<TR><TD></TD><TD>OSS/Pro:</TD><TD>OK</TD></TR>
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<TR><TD></TD><TD>ALSA:</TD><TD>OK</TD></TR>
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<TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR>
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<TR><TD COLSPAN=3><B>SB Live!</B></TD></TR>
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<TR><TD></TD><TD>OSS:</TD><TD>Analog OK, SP/DIF not working</TD></TR>
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<TR><TD></TD><TD>ALSA:</TD><TD>Both OK</TD></TR>
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<TR><TD></TD><TD>Max kHz:</TD><TD>192</TD></TR>
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<TR><TD COLSPAN=3><B>SB AWE 64</B></TD></TR>
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<TR><TD></TD><TD>OSS:</TD><TD>max 44kHz</TD></TR>
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<TR><TD></TD><TD>ALSA:</TD><TD>48kHz sounds bad</TD></TR>
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<TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR>
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<TR><TD COLSPAN=3><B>Gravis UltraSound ACE</B></TD></TR>
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<TR><TD></TD><TD>OSS:</TD><TD>not OK</TD></TR>
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<TR><TD></TD><TD>ALSA:</TD><TD>OK</TD></TR>
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<TR><TD></TD><TD>Max kHz:</TD><TD>44</TD></TR>
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<TR><TD COLSPAN=3><B>Gravis UltraSound MAX</B></TD></TR>
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<TR><TD></TD><TD>OSS:</TD><TD>OK</TD></TR>
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<TR><TD></TD><TD>ALSA:</TD><TD>OK (?)</TD></TR>
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<TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR>
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<TR><TD COLSPAN=3><B>ESS 688</B></TD></TR>
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<TR><TD></TD><TD>OSS:</TD><TD>OK</TD></TR>
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<TR><TD></TD><TD>ALSA:</TD><TD>OK (?)</TD></TR>
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<TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR>
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<TR><TD COLSPAN=3><B>C-Media cards (which ones?)</B></TD></TR>
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<TR><TD></TD><TD>OSS:</TD><TD>not OK (hissing) (?)</TD></TR>
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<TR><TD></TD><TD>ALSA:</TD><TD>OK (?)</TD></TR>
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<TR><TD></TD><TD>Max kHz:</TD><TD>?</TD></TR>
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<TR><TD COLSPAN=3><B>Yamaha cards (*ymf*)</B></TD></TR>
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<TR><TD></TD><TD>OSS:</TD><TD>not OK (?) (maybe <CODE>-ao sdl</CODE>)</TD></TR>
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<TR><TD></TD><TD>ALSA:</TD><TD>OK only with ALSA 0.5 with OSS emulation <B>AND</B> <CODE>-ao sdl</CODE> (!) (?)</TD></TR>
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<TR><TD></TD><TD>Max kHz:</TD><TD>?</TD></TR>
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<TR><TD COLSPAN=3><B>Cards with envy24 chips (like Terratec EWS88MT)</B></TD></TR>
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<TR><TD></TD><TD>OSS:</TD><TD>?</TD></TR>
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<TR><TD></TD><TD>OSS/Pro:</TD><TD>OK</TD></TR>
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<TR><TD></TD><TD>ALSA:</TD><TD>?</TD></TR>
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<TR><TD></TD><TD>Max kHz:</TD><TD>?</TD></TR>
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<TR><TD COLSPAN=3><B>PC Speaker or DAC</B></TD></TR>
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<TR><TD></TD><TD>OSS:</TD><TD>OK (Use the SDL driver: <CODE>-ao sdl</CODE>)</TD></TR>
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<TR><TD></TD><TD>ALSA:</TD><TD>no driver</TD></TR>
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<TR><TD></TD><TD>Max kHz:</TD><TD>The driver emulates 44.1, maybe more.</TD></TR>
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<TR><TD></TD><TD>Driver:</TD><TD><A HREF="ftp://ftp.infradead.org/pub/pcsp">ftp://ftp.infradead.org/pub/pcsp</A></TD></TR>
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</TABLE>
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<P>On Linux, a 2.4.x kernel is highly recommended. Kernel 2.2 is not tested.</P>
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<P>If the sound clicks when playing from CD-ROM, turn on IRQ unmasking, e.g.
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<CODE>hdparm -u1 /dev/cdrom</CODE> (<CODE>man hdparm</CODE>). This is
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generally beneficial and described in more detail in the
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<A HREF="cd-dvd.html#drives">CD-ROM section</A>.</P>
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<P>Feedback to this document is welcome. Please tell us how MPlayer
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and your sound card(s) worked together.</P>
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<H4><A NAME="af">2.3.2.3 Audio filters</A></H4>
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<P>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 <CODE>-af filter1,filter2,...</CODE>
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switch 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.</P>
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<P>Example:<BR>
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<CODE>mplayer -af resample,pan movie.avi </CODE></P>
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<P>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.</P>
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<P>The filters often have switches that change their behavior. These switches
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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 switches are omitted. Here is an example of how to use
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filters in combination with filter specific switches:</P>
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<P> <CODE>mplayer -af resample=11025,pan=1:0.5:0.5 -channels 1
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-srate 11025 media.avi</CODE></P>
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<P>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.</P>
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<P>The overall execution of the filter layer is controlled using the
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<CODE>-af-adv</CODE> switch. This switch has two suboptions:</P>
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<DL>
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<DT><CODE>force</CODE><DT>
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<DD>is a Bit field that controls how the filters are inserted and what
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speed/accuracy optimizations they use:
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<DL>
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<DT><CODE>0</CODE></DT>
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<DD>Use automatic insertion of filters and optimize according to CPU
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speed.</DD>
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<DT><CODE>1</CODE></DT>
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<DD>Use automatic insertion of filters and optimize for the highest
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speed.<BR>
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<EM>Warning:</EM> Some features in the audio filters may silently fail,
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and the sound quality may drop.</DD>
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<DT><CODE>2</CODE></DT>
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<DD>Use automatic insertion of filters and optimize for quality.</DD>
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<DT><CODE>3</CODE></DT>
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<DD>Use no automatic insertion of filters and no optimization.<BR>
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<I>Warning:</I> It may be possible to crash MPlayer using this
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setting.</DD>
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<DT><CODE>4</CODE></DT>
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<DD>Use automatic insertion of filters according to 0 above, but use
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floating point processing when possible.</DD>
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<DT><CODE>5</CODE></DT>
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<DD>Use automatic insertion of filters according to 1 above, but use
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floating point processing when possible.</DD>
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<DT><CODE>6</CODE></DT>
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<DD>Use automatic insertion of filters according to 2 above, but use
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floating point processing when possible.</DD>
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<DT><CODE>7</CODE></DT>
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<DD>Use no automatic insertion of filters according to 3 above, and use
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floating point processing when possible.</DD>
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</DL>
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</DD>
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<DT><CODE>list</CODE></DT>
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<DD>is an alias for the -af switch.</DD>
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</DL>
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<P>The filter layer is also affected by the following generic switches:
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<DL>
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<DT><CODE>-v</CODE></DT>
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<DD>Increases the verbosity level and makes most filters print out extra
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status messages.</DD>
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<DT><CODE>-channels</CODE></DT>
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<DD>This option sets the number of output channels your sound card is using.
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It also affects the number of channels that are being decoded from the
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media. If the media contains less channels than requested the channels
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filter (see below) will automatically be inserted. The routing will be the
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default routing for the channels filter.</DD>
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<DT><CODE>-srate</CODE></DT>
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<DD>This option selects the sample rate of your sound card. If the sample
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frequency of your sound card is different from that of the current media,
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the resample filter (see below) will be inserted into the audio filter layer
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to compensate for the difference.</DD>
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<DT><CODE>-format</CODE><DT>
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<DD>This option sets the sample format of the audio filter layer and the sound
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card. If the requested sample format of your sound card is different from
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that of the current media, a format filter (see below) will be inserted to
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rectify the difference.</DD>
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</DL>
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<H5><A NAME="af_resample">2.3.2.3.1 Up/Down-sampling</A></H5>
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<P>MPlayer fully supports sound up/down-sampling through the
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<CODE>resample</CODE> filter. It can be used if you
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have a fixed frequency sound card or if you are stuck with an old sound card
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that is only capable of max 44.1kHz. This filter is automatically enabled if
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it is necessary, but it can also be explicitly enabled on the command line. It
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has three switches:</P>
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<DL>
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<DT><CODE>srate <8000-192000></CODE></DT>
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<DD>is an integer used for setting the output sample
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frequency in Hz. The valid range for this parameter is 8kHz to 192kHz. If
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the input and output sample frequency are the same or if this parameter is
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omitted the filter is automatically unloaded. A high sample frequency
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normally improves the audio quality, especially when used in combination
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with other filters.</DD>
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<DT><CODE>sloppy</CODE></DT>
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<DD>is an optional binary parameter that allows the output frequency to differ
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slightly from the frequency given by <CODE>srate</CODE>. This switch can be
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used if the startup of the playback is extremely slow. It is enabled by
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default.</DD>
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<DT><CODE>type <0-2></CODE><DT>
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<DD>is an optional integer between <CODE>0</CODE> and <CODE>2</CODE> that
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selects which resampling method to use. Here <CODE>0</CODE> represents
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linear interpolation as resampling method, <CODE>1</CODE> represents
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resampling using a poly-phase filter-bank and integer processing and
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<CODE>2</CODE> represents resampling using a poly-phase filter-bank and
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floating point processing. Linear interpolation is extremely fast, but
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suffers from poor sound quality especially when used for up-sampling. The
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best quality is given by <CODE>2</CODE> but this method also suffers from
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the highest CPU load.</DD>
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</DL>
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<P>Example:<BR>
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<CODE>mplayer -af resample=44100:0:0</CODE></P>
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<P>would set the output frequency of the resample filter to 44100Hz using exact
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output frequency scaling and linear interpolation.</P>
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<H5><A NAME="af_channels">2.3.2.3.2 Changing the number of channels</A></H5>
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<P>The <CODE>channels</CODE> filter can be used for adding and removing
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channels, it can also be used for routing or copying channels. It is
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automatically enabled when the output from the audio filter layer differs from
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the input layer or when it is requested by another filter. This filter unloads
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itself if not needed. The number of switches is dynamic:</P>
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<DL>
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<DT><CODE>nch <1-6></CODE></DT>
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<DD>is an integer between <CODE>1</CODE> and <CODE>6</CODE> that is used for
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setting the number of output channels. This switch is required, leaving it
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empty results in a runtime error.</DD>
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<DT><CODE>nr <1-6></CODE></DT>
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<DD>is an integer between <CODE>1</CODE> and <CODE>6</CODE> that is used for
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specifying the number of routes. This parameter is optional. If it is
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omitted the default routing is used.</DD>
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<DT><CODE>from1:to1:from2:to2:from3:to3...</CODE></DT>
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<DD>are pairs of numbers between <CODE>0</CODE> and <CODE>5</CODE> that define
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where each channel should be routed.</DD>
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</DL>
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<P>If only <CODE>nch</CODE> is given the default routing is used, it works as
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follows: If the number of output channels is bigger than the number of input
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channels empty channels are inserted (except mixing from mono to stereo, then
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the mono channel is repeated in both of the output channels). If the number of
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output channels is smaller than the number of input channels the exceeding
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channels are truncated.</P>
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<P>Example 1:<BR>
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<CODE>mplayer -af channels=4:4:0:1:1:0:2:2:3:3 media.avi </CODE></P>
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<P>would change the number of channels to 4 and set up 4 routes that swap
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channel 0 and channel 1 and leave channel 2 and 3 intact. Observe that if
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media containing two channels was played back, channels 2 and 3 would contain
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silence but 0 and 1 would still be swapped.</P>
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<P>Example 2:<BR>
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<CODE>mplayer -af channels=6:4:0:0:0:1:0:2:0:3 media.avi </CODE></P>
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<P>would change the number of channels to 6 and set up 4 routes that copy
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channel 0 to channels 0 to 3. Channel 4 and 5 will contain silence.</P>
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<H5><A NAME="af_format">2.3.2.3.3 Sample format converter</A></H5>
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<P>The <CODE>format</CODE> filter converts between different sample formats. It
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is automatically enabled when needed by the sound card or another filter.</P>
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<DL>
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<DT><CODE>bps <number></CODE></DT>
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<DD>can be <CODE>1</CODE>, <CODE>2</CODE> or <CODE>4</CODE> and denotes the
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number of bytes per sample. This switch is required, leaving it empty
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results in a runtime error.</DD>
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<DT><CODE>f <format></CODE></DT>
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<DD>is a text string describing the sample format. The string is a
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concatenated mix of: <CODE>alaw</CODE>, <CODE>mulaw</CODE> or
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<CODE>imaadpcm</CODE>, <CODE>float</CODE> or <CODE>int</CODE>,
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<CODE>unsigned</CODE> or <CODE>signed</CODE>, <CODE>le</CODE> or
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<CODE>be</CODE> (little or big endian). This switch is required, leaving it
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empty results in a runtime error.</DD>
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</DL>
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<P>Example:<BR>
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<CODE>mplayer -af format=4:float media.avi</CODE></P>
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<P>would set the output format to 4 bytes per sample floating point
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data.</P>
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<H5><A NAME="af_delay">2.3.2.3.4 Delay</A></H5>
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<P>The <CODE>delay</CODE> filter delays the sound to the loudspeakers such that
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the sound from the different channels arrives at the listening position
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simultaneously.
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It is only useful if you have more than 2 loudspeakers. This filter has a
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variable number of parameters:</P>
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<DL>
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<DT><CODE>d1:d2:d3...</CODE></DT>
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<DD>are floating point numbers representing the delays in ms that should be
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imposed on the different channels. The minimum delay is 0ms and the maximum
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is 1000ms.</DD>
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</DL>
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<P>To calculate the required delay for the different channels do as follows:</P>
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<OL>
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<LI>Measure the distance to the loudspeakers in meters in relation to your
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listening position, giving you the distances s1 to s5 (for a 5.1 system).
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There is no point in compensating for the sub-woofer (you will not hear the
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difference anyway).</LI>
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|
<LI>Subtract the distances s1 to s5 from the maximum distance i.e.<BR>
|
|
s[i] = max(s) - s[i]; i = 1...5</LI>
|
|
<LI>Calculated the required delays in ms as<BR>
|
|
d[i] = 1000*s[i]/342; i = 1...5 </LI>
|
|
</OL>
|
|
|
|
<P>Example:<BR>
|
|
<CODE>mplayer -af delay=10.5:10.5:0:0:7:0 media.avi</CODE></P>
|
|
|
|
<P>would delay front left and right by 10.5ms, the two rear channels and the sub
|
|
by 0ms and the center channel by 7ms.</P>
|
|
|
|
|
|
<H5><A NAME="af_volume">2.3.2.3.5 Software volume control</A></H5>
|
|
|
|
<P>Software volume control is implemented by the <CODE>volume</CODE> 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.
|
|
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 switches:</P>
|
|
|
|
<DL>
|
|
<DT><CODE>v <-200 - +60></CODE></DT>
|
|
<DD>is a floating point number between <CODE>-200</CODE> and <CODE>+60</CODE>
|
|
which represents the volume level in dB. The default level is -10dB.</DD>
|
|
|
|
<DT><CODE>c</CODE></DT>
|
|
<DD>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
|
|
switch if the dynamic range of the loudspeakers is very low. Be aware that
|
|
this feature creates distortion and should be considered a last resort.</DD>
|
|
</DL>
|
|
|
|
<P>Example:<BR>
|
|
<CODE>mplayer -af volume=10.1:0 media.avi</CODE></P>
|
|
|
|
<P>would amplify the sound by 10.1dB and hard-clip if the sound level is too
|
|
high.</P>
|
|
|
|
<P>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.</P>
|
|
|
|
|
|
<H5><A NAME="af_equalizer">2.3.2.3.6 Equalizer</A></H5>
|
|
|
|
<P>The <CODE>equalizer</CODE> 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:</P>
|
|
|
|
<TABLE BORDER="0" WIDTH="100%">
|
|
<TR><TD>Band No.</TD><TD>Center frequency</TD></TR>
|
|
<TR><TD>0</TD><TD>31.25 Hz</TD></TR>
|
|
<TR><TD>1</TD><TD>62.50 Hz</TD></TR>
|
|
<TR><TD>2</TD><TD>125.0 Hz</TD></TR>
|
|
<TR><TD>3</TD><TD>250.0 Hz</TD></TR>
|
|
<TR><TD>4</TD><TD>500.0 Hz</TD></TR>
|
|
<TR><TD>5</TD><TD>1.000 kHz</TD></TR>
|
|
<TR><TD>6</TD><TD>2.000 kHz</TD></TR>
|
|
<TR><TD>7</TD><TD>4.000 kHz</TD></TR>
|
|
<TR><TD>8</TD><TD>8.000 kHz</TD></TR>
|
|
<TR><TD>9</TD><TD>16.00 kHz</TD></TR>
|
|
</TABLE>
|
|
|
|
<P>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. </P>
|
|
|
|
<P>This filter has 10 parameters:</P>
|
|
|
|
<DL>
|
|
<DT><CODE>g1:g2:g3...g10</CODE></DT>
|
|
<DD>are floating point numbers between <CODE>-12<CODE> and <CODE>+12</CODE>
|
|
representing the gain in dB for each frequency band.</DD>
|
|
</DL>
|
|
|
|
<P>Example:<BR>
|
|
<CODE>mplayer -af equalizer=11:11:10:5:0:-12:0:5:12:12 media.avi</CODE></P>
|
|
|
|
<P>would amplify the sound in the upper and lower frequency region while
|
|
canceling it almost completely around 1kHz.</P>
|
|
|
|
|
|
<H5><A NAME="af_panning">2.3.2.3.7 Panning filter </A></H5>
|
|
|
|
<P>Use the <CODE>pan</CODE> 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: </P>
|
|
|
|
<OL>
|
|
<LI>Down-mixing many channels to only a few, stereo to mono for example.</LI>
|
|
<LI>Varying the "width" of the center speaker in a surround sound system.</LI>
|
|
</OL>
|
|
|
|
<P>This filter is hard to use, and will require some tinkering before the
|
|
desired result is obtained. The number of switches for this filter depends on
|
|
the number of output channels:</P>
|
|
|
|
<DL>
|
|
<DT><CODE>nch <1-6></CODE></DT>
|
|
<DD>is an integer between <CODE>1</CODE> and <CODE>6</CODE> and is used for
|
|
setting the number of output channels. This switch is required, leaving it
|
|
empty results in a runtime error.</DD>
|
|
|
|
<DT><CODE>l00:l01:l02:..l10:l11:l12:...ln0:ln1:ln2:...</CODE></DT>
|
|
<DD>are floating point values between <CODE>0</CODE> and <CODE>1</CODE>.
|
|
<CODE>l[i][j]</CODE> determines how much of input channel j is mixed into
|
|
output channel i.</DD>
|
|
</DL>
|
|
|
|
<P>Example 1:<BR>
|
|
<CODE>mplayer -af pan=1:0.5:0.5 -channels 1 media.avi</CODE></P>
|
|
|
|
<P>would down-mix from stereo to mono.</P>
|
|
|
|
<P>Example 2:<BR>
|
|
<CODE>mplayer -af pan=3:1:0:1:0.5:0.5 -channels 3 media.avi</CODE></P>
|
|
|
|
<P>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).</P>
|
|
|
|
|
|
<H5><A NAME="af_sub">2.3.2.3.8 Sub-woofer</A></H5>
|
|
|
|
<P>The <CODE>sub</CODE> 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 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:</P>
|
|
|
|
<DL>
|
|
<DT><CODE>fc <20-300></CODE></DT>
|
|
<DD>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.</DD>
|
|
|
|
<DT><CODE>ch <0-5></CODE></DT>
|
|
<DD>is an optional integer between <CODE>0</CODE> and <CODE>5</CODE> which
|
|
determines the channel number in which to insert the sub-channel audio.
|
|
The default is channel number <CODE>5</CODE>. Observe that the number of
|
|
channels will automatically be increased to <CODE>ch</CODE> if
|
|
necessary.</DD>
|
|
</DL>
|
|
|
|
<P>Example:<BR>
|
|
<CODE>mplayer -af sub=100:4 -channels 5 media.avi</CODE></P>
|
|
|
|
<P>would add a sub-woofer channel with a cutoff frequency of 100Hz to output
|
|
channel 4.</P>
|
|
|
|
<H5><A NAME="af_surround">2.3.2.3.9 Surround-sound decoder</A></H5>
|
|
|
|
<P>Matrix encoded surround sound can be decoded by the <CODE>surround</CODE>
|
|
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:</P>
|
|
|
|
<DL>
|
|
<DT><CODE>d <0-1000></CODE></DT>
|
|
<DD>is an optional floating point number between <CODE>0</CODE> and
|
|
<CODE>1000</CODE> 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 <CODE>d</CODE>
|
|
should be set to 15ms if d1 <= d2 and to 15 + 5*(d1-d2) if d1 > d2.
|
|
The default value for <CODE>d</CODE> is 20ms.</DD>
|
|
</DL>
|
|
|
|
<P>Example:<BR>
|
|
<CODE>mplayer -af surround=15 -channels 4 media.avi</CODE></P>
|
|
|
|
<P>would add surround sound decoding with 15ms delay for the sound to the rear
|
|
speakers.</P>
|
|
|
|
|
|
|
|
|
|
<H2><STRONG>Note: Audio plugins have been deprecated by audio filters and will be
|
|
removed soon.</STRONG></H2>
|
|
|
|
<H4><A NAME="plugins">2.3.2.4 Audio plugins (deprecated)</A></H4>
|
|
|
|
<P>MPlayer 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 <CODE>-aop</CODE> switch which takes a
|
|
<CODE>list=plugin1,plugin2,...</CODE> argument. The <CODE>list</CODE> argument
|
|
is required and determines which plugins should be used and in which order they
|
|
should be executed. Example:</P>
|
|
|
|
<P> <CODE>mplayer media.avi -aop list=resample,format</CODE></P>
|
|
|
|
<P>would run the sound through the resampling plugin followed by the format
|
|
plugin.</P>
|
|
|
|
<P>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:</P>
|
|
|
|
<P> <CODE>mplayer media.avi -aop
|
|
list=resample,format:fout=44100:format=0x8</CODE></P>
|
|
|
|
<P>would set the output frequency of the resample plugin to 44100Hz and the
|
|
output format of the format plugin to AFMT_U8.</P>
|
|
|
|
<P>Currently audio plugins can not be used in MEncoder.</P>
|
|
|
|
|
|
<H5><A NAME="resample">2.3.2.4.1 Up/Downsampling</A></H5>
|
|
|
|
<P>MPlayer 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 44.1kHz.
|
|
Whether is usage of this plugin is necessary or not, is <B>autodetected</B>.
|
|
This plugin has one switch:
|
|
<CODE>fout</CODE> which is used for setting the desired output sample
|
|
frequency. It defaults to 48kHz, and is given in
|
|
<Hz>.</P>
|
|
|
|
<P>Usage:<BR>
|
|
<CODE>mplayer media.avi -aop list=resample:fout=<required
|
|
frequency in Hz, like 44100></CODE></P>
|
|
|
|
<P>Note that 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.</P>
|
|
|
|
|
|
<H5><A NAME="surround_decoding">2.3.2.4.2 Surround Sound decoding</A></H5>
|
|
|
|
<P>MPlayer 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.</P>
|
|
|
|
<P>Usage:<BR>
|
|
<CODE>mplayer media.avi -aop list=surround</CODE></P>
|
|
|
|
|
|
<H5><A NAME="format">2.3.2.3.3 Sample format converter</A></H5>
|
|
|
|
<P>If your sound card driver does not support signed 16bit <CODE>int</CODE> data type,
|
|
this plugin can
|
|
be used to change the format to one which your sound card can understand. It
|
|
has one switch, <CODE>format</CODE>, which can be set to one of the numbers
|
|
found in <CODE>libao2/afmt.h</CODE>. 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.</P>
|
|
|
|
<P>Usage:<BR>
|
|
<CODE>mplayer media.avi -aop
|
|
list=format:format=<required output format></CODE></P>
|
|
|
|
|
|
<H5><A NAME="delay">2.3.2.4.4 Delay</A></H5>
|
|
|
|
<P>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.</P>
|
|
|
|
|
|
<H5><A NAME="volume">2.3.2.4.5 Software volume control</A></H5>
|
|
|
|
<P>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 MPlayer
|
|
without changing the PCM volume setting in the mixer. It has one
|
|
switch <CODE>volume</CODE> 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.</P>
|
|
|
|
<P>Usage:<BR>
|
|
<CODE>mplayer media.avi -aop
|
|
list=volume:volume=<0-255></CODE></P>
|
|
|
|
<P>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.</P>
|
|
|
|
<P>Usage:<BR>
|
|
<CODE>mplayer media.avi -aop
|
|
list=volume:softclip</CODE></P>
|
|
|
|
|
|
<H5><A NAME="extrastereo">2.3.2.4.6 Extrastereo</A></H5>
|
|
|
|
<P>This plugin (linearly) increases the difference between left and right
|
|
channels (like the XMMS extrastereo plugin) which gives some sort of "live"
|
|
effect to playback.</P>
|
|
|
|
<P>Usage:<BR>
|
|
<CODE>mplayer media.avi -aop list=extrastereo</CODE><BR>
|
|
<CODE>mplayer media.avi -aop list=extrastereo:mul=3.45</CODE></P>
|
|
|
|
<P>The default coefficient (<CODE>mul</CODE>) is a float number that defaults
|
|
to 2.5. If you set it to 0.0, you will have mono sound (average of both
|
|
channels). If you set it to 1.0, sound will be unchanged, if you set it to
|
|
-1.0, left and right channels will be swapped.</P>
|
|
|
|
|
|
<H5><A NAME="normalizer">2.3.2.4.7 Volume normalizer</A></H5>
|
|
|
|
<P>This plugin maximizes the volume without distorting the sound.</P>
|
|
|
|
<P>Usage:<BR>
|
|
<CODE>mplayer media.avi -aop list=volnorm</CODE><BR>
|
|
|
|
|
|
</BODY>
|
|
</HTML>
|