2003-04-01 09:20:36 +00:00
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
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2001-12-16 11:51:02 +00:00
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2002-05-10 11:17:41 +00:00
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<HEAD>
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2003-04-01 09:20:36 +00:00
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<TITLE>Sound - MPlayer - Odtwarzacz film<6C>w</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-8859-2">
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2002-05-10 11:17:41 +00:00
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2003-04-01 09:20:36 +00:00
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<BODY>
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2001-12-16 11:51:02 +00:00
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2003-04-01 09:20:36 +00:00
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<H3><A NAME="audio">2.3.2 Urz<72>dzenia wyj<79>cia: d<>wi<77>k</A></H3>
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2001-12-16 11:51:02 +00:00
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2003-04-01 09:20:36 +00:00
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<H4><A NAME="sync">2.3.2.1 Synchronizacja audio/video</A></H4>
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2001-12-16 11:51:02 +00:00
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2003-04-01 09:20:36 +00:00
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<P>Interfejs do d<>wi<77>ku w MPlayerze nazywa si<73> <I>libao2</I>. Aktualnie
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zawiera nast<73>puj<75>ce sterowniki:</P>
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2001-12-16 11:51:02 +00:00
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2003-04-01 09:20:36 +00:00
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<DL>
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<DT>oss</DT>
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<DD>sterownik OSS (ioctl) (obs<62>uguje sprz<72>towe AC3)</DD>
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<DT>sdl</DT>
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<DD>sterownik SDL (obs<62>uguje demony d<>wi<77>ku takie jak <B>ESD</B> i <B>ARTS</B>)</DD>
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2001-12-16 11:51:02 +00:00
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<DT>nas</DT>
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<DD>sterownik NAS (Network Audio System)</DD>
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<DT>alsa5</DT>
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<DD>natywny sterownik ALSA 0.5</DD>
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<DT>alsa9</DT>
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<DD>natywny sterownik ALSA 0.9 (obs<62>uguje sprz<72>towe AC3)</DD>
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<DT>sun</DT>
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<DD>sterownik d<>wi<77>ku SUN (<CODE>/dev/audio</CODE>) dla u<>ytkownik<69>w BSD i Solaris8</DD>
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<DT>arts</DT>
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<DD>natywny sterownik ARTS (g<><67>wnie dla u<>ytkownik<69>w KDE)</DD>
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<DT>esd</DT>
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<DD>natywny sterownik ESD (g<><67>wnie dla u<>ytkownik<69>w GNOME)</DD>
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</DL>
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2001-12-16 11:51:02 +00:00
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2003-04-01 09:20:36 +00:00
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<P>
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Sterowniki kart d<>wi<77>kowych w Linuxie maj<61> problemy z kompatybilno<6E>ci<63>.
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Wynika to z tego <20>e MPlayer polega na wbudowanych w<>a<EFBFBD>ciwo<77>ciach <EM>prawid<EFBFBD>owo</EM>
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napisanych sterownik<69>w kt<6B>re pozwalaj<61> utrzyma<6D> prawid<69>ow<6F> synchronizacj<63> audio/video.
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Niestety, niekt<6B>rzy autorzy sterownik<69>w nie wysilaj<61> si<73> z zaprogramowaniem tych w<>a<EFBFBD>ciwo<77>ci,
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gdy<64> nie s<> wymagane do odgrywania plik<69>w MP3. </P>
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2003-04-01 09:57:28 +00:00
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<P>Inne odgrywarki multimedialne jak <A HREF="http://avifile.sourceforge.net">aviplay</A>
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czy <A HREF="http://xine.sourceforge.net">xine</A> dzia<69>aj<61>
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prosto-z-pude<64>ka z tymi sterownikami, prawdopodobnie dlatego <20>e u<>ywaj<61> one
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'prostych' metod korzystaj<61>cych z wewn<77>trznego licznika.
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Jak wskazuj<75> do<64>wiadczenia metody te nie s<> tak wydajne jak te u<>ywane przez MPlayera.</P>
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2003-04-01 09:20:36 +00:00
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2003-04-01 09:57:28 +00:00
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<P>U<EFBFBD>ywanie MPlayera z prawid<69>owo napisanymi sterownikami audio, nigdy nie spowoduje
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desynchronizacji A/V zwi<77>zanej z d<>wi<77>kiem, z wyj<79>tkiem wyj<79>tkowo <20>le stworzonych plik<69>w
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(zajrzyj do dokumentacji man by dowiedzie<69> si<73> jak sobie z nimi poradzi<7A>).</P>
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2003-04-01 09:20:36 +00:00
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2003-04-01 09:57:28 +00:00
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<P>Je<EFBFBD>li jeste<74> zmuszony u<>ywa<77> z<>ego sterownika, spr<70>buj opcji <CODE>-autosync</CODE>
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powinna ona rozwi<77>za<7A> Twoje problemy. Zajrzyj do manuala po dok<6F>adniejsze informacje.</P>
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2003-04-01 09:20:36 +00:00
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2003-04-01 09:57:28 +00:00
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<P>Informacje:</P>
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2002-05-10 11:17:41 +00:00
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<UL>
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2003-04-01 09:57:28 +00:00
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<LI>
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Je<4A>li u<>ywasz sterownika OSS, najpierw spr<70>buj opcji <CODE>-ao oss</CODE>
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(to jest ustawienie standardowe). Je<4A>li napotkasz problemy, trzaski, wstrzymania d<>wi<77>ku
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lub cokolwiek niespotykanego, spr<70>buj <CODE>-ao sdl</CODE> (UWAGA: bedziesz musial mie<69>
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zainstalowane biblioteki i nag<61><67>wki SDL). Sterownik SDL pomaga w wielu przypadkach, i
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dodatkowo obs<62>uguje ESD (GNOME) i ARTS (KDE).</LI>
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<LI>
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Je<4A>li u<>ywasz ALSA w wersji 0.5, to niemal zawsze musisz u<>ywa<77> <CODE>-ao alsa5</CODE> , gdy<64>
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ALSA 0.5 ma b<><62>dn<64> emulacj<63> OSS, i potrafi <B>wywr<EFBFBD>ci<EFBFBD> MPlayera</B> z komunikatem takim jak ten:
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<CODE>DEMUXER: Too many (945 in 8390980 bytes) video packets in the buffer!</CODE></LI>
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<LI>Pod Solarisem, u<>yj sterownika d<>wi<77>ku SUN, u<>ywaj<61>c opcji <CODE>-ao sun</CODE>,
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inaczej ani obraz ani d<>wi<77>k nie b<>dzie dzia<69>a<EFBFBD>.</LI>
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<LI>Je<EFBFBD>li d<>wi<77>k klika w czasie grania z CD-ROMu, w<><77>cz 'IRQ unmasking', na przyk<79>ad:
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<CODE>hdparm -u1 /dev/cdrom</CODE> (<CODE>man hdparm</CODE>).
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Ta opcja generalnie pomaga, i jest dok<6F>adniej opisana w <A HREF="cd-dvd.html#drives">sekcji o CD-ROM</A>.</LI>
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</UL>
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2003-04-01 09:57:28 +00:00
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<H4><A NAME="experiences">2.3.2.2 Do<44>wiadczenia z kartami d<>wi<77>kowymi, zalecenia</A></H4>
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2003-04-01 09:20:36 +00:00
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2003-04-01 09:57:28 +00:00
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<P>Na Linuxie zalecane jest j<>dro 2.4.x. J<>dro 2.2 nie by<62>o testowane.</P>
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2003-04-01 09:20:36 +00:00
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<P>Linux sound drivers are primarily provided by the free version of OSS. These
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drivers have been superceded by <A HREF="http://www.alsa-project.org">ALSA</A>
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(Advanced Linux Sound Architecture) in the 2.5 development series. If your
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distribution does not already use ALSA you may wish to try their drivers if
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you experience sound problems. ALSA drivers are generally superior to OSS in
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compatibility, performance and features. But some sound cards are only
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supported by the commercial OSS drivers from
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<A HREF="http://www.opensound.com/">4Front Technologies</A>. They also support
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several non-Linux systems.</P>
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<TABLE BORDER="1" WIDTH="100%">
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<TR>
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<TH ROWSPAN="2"><B>SOUND CARD</B></TH>
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<TH COLSPAN="4"><B>DRIVER</B></TH>
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<TH ROWSPAN="2"><B>Max kHz</B></TH>
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<TH ROWSPAN="2"><B>Max Channels</B></TH>
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<TH ROWSPAN="2"><B>Max Opens<FONT SIZE="-2"><A HREF=#note1>[1]</A></FONT></B></TH>
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</TR>
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<TR>
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<TH><B>OSS/Free</B></TH>
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<TH><B>ALSA</B></TH>
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<TH><B>OSS/Pro</B></TH>
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<TH><B>other</B></TH>
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</TR>
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<TR>
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<TD><B>VIA onboard (686/A/B, 8233, 8235)</B></TD>
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<TD><A HREF="http://sourceforge.net/project/showfiles.php?group_id=3242&release_id=59602">via82cxxx_audio</A></TD>
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<TD>snd-via82xx</TD>
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<TD> </TD>
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<TD> </TD>
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<TD>4-48 kHz or 48 kHz only, depending on the chipset</TD>
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<TD> </TD>
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<TD> </TD>
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</TR>
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<TR>
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<TD><B>Aureal Vortex 2</B></TD>
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<TD>none</TD>
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<TD>none</TD>
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<TD>OK</TD>
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<TD><A HREF="http://aureal.sourceforge.net">Linux Aureal Drivers</A><BR>
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<A HREF="http://makacs.poliod.hu/~pontscho/aureal/au88xx-1.1.3.tar.bz2">buffer size increased to 32k</A></TD>
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<TD>48</TD>
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<TD>4.1</TD>
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<TD>5+</TD>
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</TR>
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<TR>
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<TD><B>SB Live!</B></TD>
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<TD>Analog OK, SP/DIF not working</TD>
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<TD>Both OK</TD>
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<TD>Both OK</TD>
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<TD><A HREF="http://opensource.creative.com">Creative's OSS driver (SP/DIF support)</A></TD>
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<TD>192</TD>
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<TD>4.0/5.1</TD>
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<TD>32</TD>
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</TR>
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<TR>
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<TD><B>SB 128 PCI (es1371)</B></TD>
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<TD>OK</TD>
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<TD>?</TD>
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<TD> </TD>
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<TD> </TD>
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<TD>48</TD>
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<TD>stereo</TD>
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<TD>2</TD>
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</TR>
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<TR>
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<TD><B>SB AWE 64</B></TD>
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<TD>max 44kHz</TD>
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<TD>48kHz sounds bad</TD>
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<TD> </TD>
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<TD> </TD>
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<TD>48</TD>
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<TD> </TD>
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<TD> </TD>
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</TR>
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<TR>
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<TD><B>GUS PnP</B></TD>
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<TD>none</TD>
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<TD>OK</TD>
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<TD>OK</TD>
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<TD> </TD>
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<TD>48</TD>
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<TD> </TD>
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<TD> </TD>
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</TR>
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<TR>
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<TD><B>Gravis UltraSound ACE</B></TD>
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<TD>not OK</TD>
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<TD>OK</TD>
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<TD> </TD>
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<TD> </TD>
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<TD>44</TD>
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<TD> </TD>
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<TD> </TD>
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</TR>
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<TR>
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<TD><B>Gravis UltraSound MAX</B></TD>
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<TD>OK</TD>
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<TD>OK (?)</TD>
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<TD> </TD>
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<TD> </TD>
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<TD>48</TD>
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<TD> </TD>
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<TD> </TD>
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</TR>
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<TR>
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<TD><B>ESS 688</B></TD>
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<TD>OK</TD>
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<TD>OK (?)</TD>
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<TD> </TD>
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<TD> </TD>
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<TD>48</TD>
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<TD> </TD>
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<TD> </TD>
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</TR>
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<TR>
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<TD><B>C-Media cards (which ones?)</B></TD>
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<TD>not OK (hissing) (?)</TD>
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<TD>OK</TD>
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<TD> </TD>
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<TD> </TD>
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<TD>?</TD>
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<TD> </TD>
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<TD> </TD>
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</TR>
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<TR>
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<TD><B>Yamaha cards (*ymf*)</B></TD>
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<TD>not OK (?) (maybe <CODE>-ao sdl</CODE>)</TD>
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<TD>OK only with ALSA 0.5 with OSS emulation <B>AND</B>
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<CODE>-ao sdl</CODE> (!) (?)</TD>
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<TD> </TD>
|
|
|
|
|
|
<TD> </TD>
|
|
|
|
|
|
<TD>?</TD>
|
|
|
|
|
|
<TD> </TD>
|
|
|
|
|
|
<TD> </TD>
|
|
|
|
|
|
</TR>
|
|
|
|
|
|
|
|
|
|
|
|
<TR>
|
|
|
|
|
|
<TD><B>Cards with envy24 chips (like Terratec EWS88MT)</B></TD>
|
|
|
|
|
|
<TD>?</TD>
|
|
|
|
|
|
<TD>?</TD>
|
|
|
|
|
|
<TD>OK</TD>
|
|
|
|
|
|
<TD> </TD>
|
|
|
|
|
|
<TD>?</TD>
|
|
|
|
|
|
<TD> </TD>
|
|
|
|
|
|
<TD> </TD>
|
|
|
|
|
|
</TR>
|
|
|
|
|
|
|
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|
|
|
|
<TR>
|
|
|
|
|
|
<TD><B>PC Speaker or DAC</B></TD>
|
|
|
|
|
|
<TD>OK</TD>
|
|
|
|
|
|
<TD>none</TD>
|
|
|
|
|
|
<TD> </TD>
|
|
|
|
|
|
<TD><A HREF="http://www.geocities.com/stssppnn/pcsp.html">Linux PC speaker OSS driver</a></TD>
|
|
|
|
|
|
<TD>The driver emulates 44.1, maybe more.</TD>
|
|
|
|
|
|
<TD>mono</TD>
|
|
|
|
|
|
<TD>1</TD>
|
|
|
|
|
|
</TR>
|
|
|
|
|
|
|
2002-05-10 11:17:41 +00:00
|
|
|
|
</TABLE>
|
|
|
|
|
|
|
2003-04-01 09:20:36 +00:00
|
|
|
|
<P><A NAME="note1"><B>[1]</B></A>: the number of applications that are able to use the
|
|
|
|
|
|
device <I>at the same time</I>.</P>
|
|
|
|
|
|
|
|
|
|
|
|
<P>Feedback to this document is welcome. Please tell us how MPlayer
|
|
|
|
|
|
and your sound card(s) worked together.</P>
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
<H4><A NAME="af">2.3.2.3 Audio filters</A></H4>
|
|
|
|
|
|
|
|
|
|
|
|
<P>The old audio plugins have been superseded by a new audio filter layer. Audio
|
|
|
|
|
|
filters are used for changing the properties of the audio data before the
|
|
|
|
|
|
sound reaches the sound card. The activation and deactivation of the filters
|
|
|
|
|
|
is normally automated but can be overridden. The filters are activated when
|
|
|
|
|
|
the properties of the audio data differ from those required by the sound card
|
|
|
|
|
|
and deactivated if unnecessary. The <CODE>-af filter1,filter2,...</CODE>
|
|
|
|
|
|
option is used to override the automatic activation of filters or to insert
|
|
|
|
|
|
filters that are not automatically inserted. The filters will be executed as
|
|
|
|
|
|
they appear in the comma separated list.</P>
|
|
|
|
|
|
|
|
|
|
|
|
<P>Example:<BR>
|
|
|
|
|
|
<CODE>mplayer -af resample,pan movie.avi </CODE></P>
|
|
|
|
|
|
|
|
|
|
|
|
<P>would run the sound through the resampling filter followed by the pan filter.
|
|
|
|
|
|
Observe that the list must not contain any spaces, else it will fail.</P>
|
|
|
|
|
|
|
|
|
|
|
|
<P>The filters often have options that change their behavior. These options
|
|
|
|
|
|
are explained in detail in the sections below. A filter will execute using
|
|
|
|
|
|
default settings if its options are omitted. Here is an example of how to use
|
|
|
|
|
|
filters in combination with filter specific options:</P>
|
|
|
|
|
|
|
|
|
|
|
|
<P> <CODE>mplayer -af resample=11025,pan=1:0.5:0.5 -channels 1
|
|
|
|
|
|
-srate 11025 media.avi</CODE></P>
|
|
|
|
|
|
|
|
|
|
|
|
<P>would set the output frequency of the resample filter to 11025Hz and downmix
|
|
|
|
|
|
the audio to 1 channel using the pan filter.</P>
|
|
|
|
|
|
|
|
|
|
|
|
<P>The overall execution of the filter layer is controlled using the
|
|
|
|
|
|
<CODE>-af-adv</CODE> option. This option has two suboptions:</P>
|
|
|
|
|
|
|
|
|
|
|
|
<DL>
|
|
|
|
|
|
<DT><CODE>force</CODE><DT>
|
|
|
|
|
|
<DD>is a Bit field that controls how the filters are inserted and what
|
|
|
|
|
|
speed/accuracy optimizations they use:
|
|
|
|
|
|
<DL>
|
|
|
|
|
|
<DT><CODE>0</CODE></DT>
|
|
|
|
|
|
<DD>Use automatic insertion of filters and optimize according to CPU
|
|
|
|
|
|
speed.</DD>
|
|
|
|
|
|
<DT><CODE>1</CODE></DT>
|
|
|
|
|
|
<DD>Use automatic insertion of filters and optimize for the highest
|
|
|
|
|
|
speed.<BR>
|
|
|
|
|
|
<EM>Warning:</EM> Some features in the audio filters may silently fail,
|
|
|
|
|
|
and the sound quality may drop.</DD>
|
|
|
|
|
|
<DT><CODE>2</CODE></DT>
|
|
|
|
|
|
<DD>Use automatic insertion of filters and optimize for quality.</DD>
|
|
|
|
|
|
<DT><CODE>3</CODE></DT>
|
|
|
|
|
|
<DD>Use no automatic insertion of filters and no optimization.<BR>
|
|
|
|
|
|
<I>Warning:</I> It may be possible to crash MPlayer using this
|
|
|
|
|
|
setting.</DD>
|
|
|
|
|
|
<DT><CODE>4</CODE></DT>
|
|
|
|
|
|
<DD>Use automatic insertion of filters according to 0 above, but use
|
|
|
|
|
|
floating point processing when possible.</DD>
|
|
|
|
|
|
<DT><CODE>5</CODE></DT>
|
|
|
|
|
|
<DD>Use automatic insertion of filters according to 1 above, but use
|
|
|
|
|
|
floating point processing when possible.</DD>
|
|
|
|
|
|
<DT><CODE>6</CODE></DT>
|
|
|
|
|
|
<DD>Use automatic insertion of filters according to 2 above, but use
|
|
|
|
|
|
floating point processing when possible.</DD>
|
|
|
|
|
|
<DT><CODE>7</CODE></DT>
|
|
|
|
|
|
<DD>Use no automatic insertion of filters according to 3 above, and use
|
|
|
|
|
|
floating point processing when possible.</DD>
|
|
|
|
|
|
</DL>
|
|
|
|
|
|
</DD>
|
|
|
|
|
|
|
|
|
|
|
|
<DT><CODE>list</CODE></DT>
|
|
|
|
|
|
<DD>is an alias for the -af option.</DD>
|
|
|
|
|
|
</DL>
|
|
|
|
|
|
|
|
|
|
|
|
<P>The filter layer is also affected by the following generic options:
|
|
|
|
|
|
|
|
|
|
|
|
<DL>
|
|
|
|
|
|
<DT><CODE>-v</CODE></DT>
|
|
|
|
|
|
<DD>Increases the verbosity level and makes most filters print out extra
|
|
|
|
|
|
status messages.</DD>
|
|
|
|
|
|
<DT><CODE>-channels</CODE></DT>
|
|
|
|
|
|
<DD>This option sets the number of output channels you would like your
|
|
|
|
|
|
sound card to use.
|
|
|
|
|
|
It also affects the number of channels that are being decoded from the
|
|
|
|
|
|
media. If the media contains less channels than requested the channels
|
|
|
|
|
|
filter (see below) will automatically be inserted. The routing will be the
|
|
|
|
|
|
default routing for the channels filter.</DD>
|
|
|
|
|
|
<DT><CODE>-srate</CODE></DT>
|
|
|
|
|
|
<DD>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.</DD>
|
|
|
|
|
|
<DT><CODE>-format</CODE><DT>
|
|
|
|
|
|
<DD>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.</DD>
|
|
|
|
|
|
</DL>
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
<H4><A NAME="af_resample">2.3.2.3.1 Up/Down-sampling</A></H4>
|
|
|
|
|
|
|
|
|
|
|
|
<P>MPlayer fully supports sound up/down-sampling through the
|
|
|
|
|
|
<CODE>resample</CODE> 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:</P>
|
|
|
|
|
|
|
|
|
|
|
|
<DL>
|
|
|
|
|
|
<DT><CODE>srate <8000-192000></CODE></DT>
|
|
|
|
|
|
<DD>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.</DD>
|
|
|
|
|
|
|
|
|
|
|
|
<DT><CODE>sloppy</CODE></DT>
|
|
|
|
|
|
<DD>is an optional binary parameter that allows the output frequency to differ
|
|
|
|
|
|
slightly from the frequency given by <CODE>srate</CODE>. This option can be
|
|
|
|
|
|
used if the startup of the playback is extremely slow. It is enabled by
|
|
|
|
|
|
default.</DD>
|
|
|
|
|
|
|
|
|
|
|
|
<DT><CODE>type <0-2></CODE><DT>
|
|
|
|
|
|
<DD>is an optional integer between <CODE>0</CODE> and <CODE>2</CODE> that
|
|
|
|
|
|
selects which resampling method to use. Here <CODE>0</CODE> represents
|
|
|
|
|
|
linear interpolation as resampling method, <CODE>1</CODE> represents
|
|
|
|
|
|
resampling using a poly-phase filter-bank and integer processing and
|
|
|
|
|
|
<CODE>2</CODE> 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 <CODE>2</CODE> but this method also suffers from
|
|
|
|
|
|
the highest CPU load.</DD>
|
|
|
|
|
|
</DL>
|
|
|
|
|
|
|
|
|
|
|
|
<P>Example:<BR>
|
|
|
|
|
|
<CODE>mplayer -af resample=44100:0:0</CODE></P>
|
|
|
|
|
|
|
|
|
|
|
|
<P>would set the output frequency of the resample filter to 44100Hz using exact
|
|
|
|
|
|
output frequency scaling and linear interpolation.</P>
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
<H4><A NAME="af_channels">2.3.2.3.2 Changing the number of channels</A></H4>
|
|
|
|
|
|
|
|
|
|
|
|
<P>The <CODE>channels</CODE> 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:</P>
|
|
|
|
|
|
|
|
|
|
|
|
<DL>
|
|
|
|
|
|
<DT><CODE>nch <1-6></CODE></DT>
|
|
|
|
|
|
<DD>is an integer between <CODE>1</CODE> and <CODE>6</CODE> that is used for
|
|
|
|
|
|
setting the number of output channels. This option is required, leaving it
|
|
|
|
|
|
empty results in a runtime error.</DD>
|
|
|
|
|
|
|
|
|
|
|
|
<DT><CODE>nr <1-6></CODE></DT>
|
|
|
|
|
|
<DD>is an integer between <CODE>1</CODE> and <CODE>6</CODE> that is used for
|
|
|
|
|
|
specifying the number of routes. This parameter is optional. If it is
|
|
|
|
|
|
omitted the default routing is used.</DD>
|
|
|
|
|
|
|
|
|
|
|
|
<DT><CODE>from1:to1:from2:to2:from3:to3...</CODE></DT>
|
|
|
|
|
|
<DD>are pairs of numbers between <CODE>0</CODE> and <CODE>5</CODE> that define
|
|
|
|
|
|
where each channel should be routed.</DD>
|
|
|
|
|
|
</DL>
|
|
|
|
|
|
|
|
|
|
|
|
<P>If only <CODE>nch</CODE> 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.</P>
|
|
|
|
|
|
|
|
|
|
|
|
<P>Example 1:<BR>
|
|
|
|
|
|
<CODE>mplayer -af channels=4:4:0:1:1:0:2:2:3:3 media.avi </CODE></P>
|
|
|
|
|
|
|
|
|
|
|
|
<P>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.</P>
|
|
|
|
|
|
|
|
|
|
|
|
<P>Example 2:<BR>
|
|
|
|
|
|
<CODE>mplayer -af channels=6:4:0:0:0:1:0:2:0:3 media.avi </CODE></P>
|
|
|
|
|
|
|
|
|
|
|
|
<P>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.</P>
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
<H4><A NAME="af_format">2.3.2.3.3 Sample format converter</A></H4>
|
|
|
|
|
|
|
|
|
|
|
|
<P>The <CODE>format</CODE> filter converts between different sample formats. It
|
|
|
|
|
|
is automatically enabled when needed by the sound card or another filter.</P>
|
|
|
|
|
|
|
|
|
|
|
|
<DL>
|
|
|
|
|
|
<DT><CODE>bps <number></CODE></DT>
|
|
|
|
|
|
<DD>can be <CODE>1</CODE>, <CODE>2</CODE> or <CODE>4</CODE> and denotes the
|
|
|
|
|
|
number of bytes per sample. This option is required, leaving it empty
|
|
|
|
|
|
results in a runtime error.</DD>
|
|
|
|
|
|
|
|
|
|
|
|
<DT><CODE>f <format></CODE></DT>
|
|
|
|
|
|
<DD>is a text string describing the sample format. The string is a
|
|
|
|
|
|
concatenated mix of: <CODE>alaw</CODE>, <CODE>mulaw</CODE> or
|
|
|
|
|
|
<CODE>imaadpcm</CODE>, <CODE>float</CODE> or <CODE>int</CODE>,
|
|
|
|
|
|
<CODE>unsigned</CODE> or <CODE>signed</CODE>, <CODE>le</CODE> or
|
|
|
|
|
|
<CODE>be</CODE> (little or big endian). This option is required, leaving it
|
|
|
|
|
|
empty results in a runtime error.</DD>
|
|
|
|
|
|
</DL>
|
|
|
|
|
|
|
|
|
|
|
|
<P>Example:<BR>
|
|
|
|
|
|
<CODE>mplayer -af format=4:float media.avi</CODE></P>
|
|
|
|
|
|
|
|
|
|
|
|
<P>would set the output format to 4 bytes per sample floating point
|
|
|
|
|
|
data.</P>
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
<H4><A NAME="af_delay">2.3.2.3.4 Delay</A></H4>
|
|
|
|
|
|
|
|
|
|
|
|
<P>The <CODE>delay</CODE> 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:</P>
|
|
|
|
|
|
|
|
|
|
|
|
<DL>
|
|
|
|
|
|
<DT><CODE>d1:d2:d3...</CODE></DT>
|
|
|
|
|
|
<DD>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.</DD>
|
|
|
|
|
|
</DL>
|
|
|
|
|
|
|
|
|
|
|
|
<P>To calculate the required delay for the different channels do as follows:</P>
|
|
|
|
|
|
|
|
|
|
|
|
<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>
|
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|
|
s[i] = max(s) - s[i]; i = 1...5</LI>
|
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|
|
<LI>Calculated the required delays in ms as<BR>
|
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|
d[i] = 1000*s[i]/342; i = 1...5 </LI>
|
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|
</OL>
|
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|
|
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|
|
|
|
|
<P>Example:<BR>
|
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|
|
|
<CODE>mplayer -af delay=10.5:10.5:0:0:7:0 media.avi</CODE></P>
|
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|
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<P>would delay front left and right by 10.5ms, the two rear channels and the sub
|
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|
by 0ms and the center channel by 7ms.</P>
|
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|
<H4><A NAME="af_volume">2.3.2.3.5 Software volume control</A></H4>
|
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|
<P>Software volume control is implemented by the <CODE>volume</CODE> audio
|
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|
|
filter. Use this filter with caution since
|
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|
|
it can reduce the signal to noise ratio of the sound. In most cases it is best
|
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|
|
to set the level for the PCM sound to max, leave this filter out and control
|
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|
|
the output level to your speakers with the master volume control of the mixer.
|
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|
In case your sound card has a digital PCM mixer instead of an analog one, and
|
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|
|
you hear distortion, use the MASTER mixer instead.
|
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|
If there is an external amplifier connected to the computer (this is almost
|
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|
|
|
always the case), the noise level can be minimized by adjusting the master
|
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|
level and the volume knob on the amplifier until the hissing noise in the
|
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|
background is gone. This filter has two options:</P>
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|
<DL>
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|
<DT><CODE>v <-200 - +60></CODE></DT>
|
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|
|
<DD>is a floating point number between <CODE>-200</CODE> and <CODE>+60</CODE>
|
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|
|
which represents the volume level in dB. The default level is 0dB.</DD>
|
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|
<DT><CODE>c</CODE></DT>
|
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|
|
<DD>is a binary control that turns soft clipping on and off. Soft-clipping can
|
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|
|
|
make the sound more smooth if very high volume levels are used. Enable this
|
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|
|
|
|
option if the dynamic range of the loudspeakers is very low. Be aware that
|
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|
|
|
this feature creates distortion and should be considered a last resort.</DD>
|
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|
</DL>
|
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|
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|
|
|
|
<P>Example:<BR>
|
|
|
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|
|
<CODE>mplayer -af volume=10.1:0 media.avi</CODE></P>
|
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|
|
|
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|
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|
<P>would amplify the sound by 10.1dB and hard-clip if the sound level is too
|
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|
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|
|
high.</P>
|
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|
|
|
|
|
|
|
|
|
<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
|
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|
|
|
|
utilized.</P>
|
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|
<H4><A NAME="af_equalizer">2.3.2.3.6 Equalizer</A></H4>
|
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|
<P>The <CODE>equalizer</CODE> filter represents a 10 octave band graphic
|
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|
|
|
|
equalizer, implemented using 10 IIR
|
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|
|
|
|
band pass filters. This means that it works regardless of what type of audio
|
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|
|
|
|
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>
|
|
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|
|
<TR><TD>3</TD><TD>250.0 Hz</TD></TR>
|
|
|
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|
|
<TR><TD>4</TD><TD>500.0 Hz</TD></TR>
|
|
|
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|
|
<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>
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
<H4><A NAME="af_panning">2.3.2.3.7 Panning filter</A></H4>
|
|
|
|
|
|
|
|
|
|
|
|
<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 options 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 option 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>
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
<H4><A NAME="af_sub">2.3.2.3.8 Sub-woofer</A></H4>
|
|
|
|
|
|
|
|
|
|
|
|
<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 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>
|
|
|
|
|
|
|
|
|
|
|
|
<H4><A NAME="af_surround">2.3.2.3.9 Surround-sound decoder</A></H4>
|
|
|
|
|
|
|
|
|
|
|
|
<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>
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
<H4><A NAME="plugins">2.3.2.4 Audio plugins (deprecated)</A></H4>
|
|
|
|
|
|
|
|
|
|
|
|
<H2><STRONG>Note: Audio plugins have been deprecated by audio filters and will be
|
|
|
|
|
|
removed soon.</STRONG></H2>
|
|
|
|
|
|
|
|
|
|
|
|
<P>MPlayer has support for audio plugins. Audio plugins can be used to
|
|
|
|
|
|
change the properties of the audio data before it reaches the sound
|
|
|
|
|
|
card. They are enabled using the <CODE>-aop</CODE> option which takes a
|
2002-05-10 11:17:41 +00:00
|
|
|
|
<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
|
2003-04-01 09:20:36 +00:00
|
|
|
|
should be executed. Example:</P>
|
2002-05-10 11:17:41 +00:00
|
|
|
|
|
|
|
|
|
|
<P> <CODE>mplayer media.avi -aop list=resample,format</CODE></P>
|
|
|
|
|
|
|
|
|
|
|
|
<P>would run the sound through the resampling plugin followed by the format
|
2003-04-01 09:20:36 +00:00
|
|
|
|
plugin.</P>
|
2002-05-10 11:17:41 +00:00
|
|
|
|
|
2003-04-01 09:20:36 +00:00
|
|
|
|
<P>The plugins can also have options that change their behavior. These
|
|
|
|
|
|
options are explained in detail in the sections below. A plugin will execute
|
|
|
|
|
|
using default settings if its options are omitted. Here is an example of how
|
|
|
|
|
|
to use plugins in combination with plugin specific options:</P>
|
2002-05-10 11:17:41 +00:00
|
|
|
|
|
|
|
|
|
|
<P> <CODE>mplayer media.avi -aop
|
2003-04-01 09:20:36 +00:00
|
|
|
|
list=resample,format:fout=44100:format=0x8</CODE></P>
|
2002-05-10 11:17:41 +00:00
|
|
|
|
|
|
|
|
|
|
<P>would set the output frequency of the resample plugin to 44100Hz and the
|
2003-04-01 09:20:36 +00:00
|
|
|
|
output format of the format plugin to AFMT_U8.</P>
|
2002-05-10 11:17:41 +00:00
|
|
|
|
|
2003-04-01 09:20:36 +00:00
|
|
|
|
<P>Currently audio plugins cannot be used in MEncoder.</P>
|
2002-05-10 11:17:41 +00:00
|
|
|
|
|
|
|
|
|
|
|
2003-04-01 09:20:36 +00:00
|
|
|
|
<H4><A NAME="resample">2.3.2.4.1 Up/Downsampling</A></H4>
|
2002-05-10 11:17:41 +00:00
|
|
|
|
|
2003-04-01 09:20:36 +00:00
|
|
|
|
<P>MPlayer fully supports up/downsampling of the sound. This plugin can
|
2002-05-10 11:17:41 +00:00
|
|
|
|
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.
|
2003-04-01 09:20:36 +00:00
|
|
|
|
MPlayer <EM>autodetects</EM> whether or not usage of this plugin is necessary.
|
|
|
|
|
|
This plugin has one option, <CODE>fout</CODE>, which is used for setting the
|
|
|
|
|
|
desired output sample frequency. The value is given in Hz, and defaults to
|
|
|
|
|
|
48kHz.</P>
|
2002-05-10 11:17:41 +00:00
|
|
|
|
|
|
|
|
|
|
<P>Usage:<BR>
|
2003-04-01 09:20:36 +00:00
|
|
|
|
<CODE>mplayer media.avi -aop list=resample:fout=<required
|
2001-12-29 18:46:10 +00:00
|
|
|
|
frequency in Hz, like 44100></CODE></P>
|
|
|
|
|
|
|
2002-05-10 11:17:41 +00:00
|
|
|
|
<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
|
2003-04-01 09:20:36 +00:00
|
|
|
|
and cause audio distortion.</P>
|
|
|
|
|
|
|
2002-05-10 11:17:41 +00:00
|
|
|
|
|
2003-04-01 09:20:36 +00:00
|
|
|
|
<H4><A NAME="surround_decoding">2.3.2.4.2 Surround Sound decoding</A></H4>
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2002-05-10 11:17:41 +00:00
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2003-04-01 09:20:36 +00:00
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<P>MPlayer has an audio plugin that can decode matrix encoded
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surround sound. Dolby Surround is an example of a matrix encoded format.
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Many files with 2 channel audio actually contain matrixed surround sound.
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To use this feature you need a sound card supporting at least 4 channels.</P>
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2002-05-10 11:17:41 +00:00
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<P>Usage:<BR>
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2003-04-01 09:20:36 +00:00
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<CODE>mplayer media.avi -aop list=surround</CODE></P>
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2002-05-10 11:17:41 +00:00
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2003-04-01 09:20:36 +00:00
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<H4><A NAME="format">2.3.2.3.3 Sample format converter</A></H4>
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<P>If your sound card driver does not support signed 16bit <CODE>int</CODE> data type,
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2002-05-10 11:17:41 +00:00
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this plugin can
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be used to change the format to one which your sound card can understand. It
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2003-04-01 09:20:36 +00:00
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has one option, <CODE>format</CODE>, which can be set to one of the numbers
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2002-05-10 11:17:41 +00:00
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found in <CODE>libao2/afmt.h</CODE>. This plugin is hardly ever needed and is
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intended for advanced users. Keep in mind that this plugin only changes the
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2003-04-01 09:20:36 +00:00
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sample format and not the sample frequency or the number of channels.</P>
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2002-05-10 11:17:41 +00:00
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<P>Usage:<BR>
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<CODE>mplayer media.avi -aop
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2003-04-01 09:20:36 +00:00
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list=format:format=<required output format></CODE></P>
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2002-05-10 11:17:41 +00:00
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2003-04-01 09:20:36 +00:00
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<H4><A NAME="delay">2.3.2.4.4 Delay</A></H4>
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2002-05-10 11:17:41 +00:00
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<P>This plugin delays the sound and is intended as an example of how to develop
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new plugins. It can not be used for anything useful from a users perspective
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and is mentioned here for the sake of completeness only. Do not use this
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plugin unless you are a developer.</P>
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2003-04-01 09:20:36 +00:00
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<P>If you have a file with a consistent A/V sync fault, use the <CODE>+/-</CODE>
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keys to adjust timings on-the-fly instead. Usage of the OSD is recommended
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to make this easier.</P>
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<H4><A NAME="volume">2.3.2.4.5 Software volume control</A></H4>
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2002-05-10 11:17:41 +00:00
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<P>This plugin is a software replacement for the volume control, and
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can be used on machines with a broken mixer device. It can also be
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2003-04-01 09:20:36 +00:00
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used if one wants to change the output volume of MPlayer
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2002-05-10 11:17:41 +00:00
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without changing the PCM volume setting in the mixer. It has one
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2003-04-01 09:20:36 +00:00
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option <CODE>volume</CODE> that is used for setting the initial
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2002-05-10 11:17:41 +00:00
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sound level. The initial sound level can be set to values between 0
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and 255 and defaults to 101 which equals 0dB amplification. Use this
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plugin with caution since it can reduce the signal to noise ratio of
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the sound. In most cases it is best to set the level for the PCM
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sound to max, leave this plugin out and control the output level to
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2003-04-01 09:20:36 +00:00
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your speakers with the MASTER volume control of the mixer.
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In case your sound card has a digital PCM mixer instead of an analog one, and
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you hear distortion, use the MASTER mixer instead.
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2002-05-10 11:17:41 +00:00
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external amplifier connected to the computer (this is almost always
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the case), the noise level can be minimized by adjusting the master
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level and the volume knob on the amplifier until the hissing noise
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2003-04-01 09:20:36 +00:00
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in the background is gone.</P>
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2002-05-10 11:17:41 +00:00
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<P>Usage:<BR>
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<CODE>mplayer media.avi -aop
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2003-04-01 09:20:36 +00:00
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list=volume:volume=<0-255></CODE></P>
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2002-05-10 11:17:41 +00:00
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<P>This plugin also has compressor or "soft-clipping" capabilities.
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Compression can be used if the dynamic range of the sound is very
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high or if the dynamic range of the loudspeakers is very
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low. Be aware that this feature creates distortion and should be
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2003-04-01 09:20:36 +00:00
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considered a last resort.</P>
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2002-05-10 11:17:41 +00:00
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<P>Usage:<BR>
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<CODE>mplayer media.avi -aop
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2003-04-01 09:20:36 +00:00
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list=volume:softclip</CODE></P>
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2002-05-10 11:17:41 +00:00
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2003-04-01 09:20:36 +00:00
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<H4><A NAME="extrastereo">2.3.2.4.6 Extrastereo</A></H4>
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2002-05-10 11:17:41 +00:00
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<P>This plugin (linearly) increases the difference between left and right
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channels (like the XMMS extrastereo plugin) which gives some sort of "live"
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2003-04-01 09:20:36 +00:00
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effect to playback.</P>
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2002-05-10 11:17:41 +00:00
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<P>Usage:<BR>
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<CODE>mplayer media.avi -aop list=extrastereo</CODE><BR>
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<CODE>mplayer media.avi -aop list=extrastereo:mul=3.45</CODE></P>
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<P>The default coefficient (<CODE>mul</CODE>) is a float number that defaults
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to 2.5. If you set it to 0.0, you will have mono sound (average of both
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2003-04-01 09:20:36 +00:00
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channels). If you set it to 1.0, sound will be unchanged, if you set it to
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-1.0, left and right channels will be swapped.</P>
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2002-05-10 11:17:41 +00:00
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2003-04-01 09:20:36 +00:00
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<H4><A NAME="normalizer">2.3.2.4.7 Volume normalizer</A></H4>
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2002-05-10 11:17:41 +00:00
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<P>This plugin maximizes the volume without distorting the sound.</P>
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<P>Usage:<BR>
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<CODE>mplayer media.avi -aop list=volnorm</CODE><BR>
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2001-12-16 11:51:02 +00:00
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</BODY>
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</HTML>
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