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|
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<HTML>
<HEAD>
<TITLE>Sound - MPlayer - Odtwarzacz filmów</TITLE>
<LINK REL="stylesheet" TYPE="text/css" HREF="../default.css">
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-2">
</HEAD>
<BODY>
<H3><A NAME="audio">2.3.2 Urządzenia wyjścia: dźwięk</A></H3>
<H4><A NAME="sync">2.3.2.1 Synchronizacja audio/video</A></H4>
<P>Interfejs do dźwięku w MPlayerze nazywa się <I>libao2</I>. Aktualnie
zawiera następujące sterowniki:</P>
<DL>
<DT>oss</DT>
<DD>sterownik OSS (ioctl) (obsługuje sprzętowe AC3)</DD>
<DT>sdl</DT>
<DD>sterownik SDL (obsługuje demony dźwięku takie jak <B>ESD</B> i <B>ARTS</B>)</DD>
<DT>nas</DT>
<DD>sterownik NAS (Network Audio System)</DD>
<DT>alsa5</DT>
<DD>natywny sterownik ALSA 0.5</DD>
<DT>alsa9</DT>
<DD>natywny sterownik ALSA 0.9 (obsługuje sprzętowe AC3)</DD>
<DT>sun</DT>
<DD>sterownik dźwięku SUN (<CODE>/dev/audio</CODE>) dla użytkowników BSD i Solaris8</DD>
<DT>arts</DT>
<DD>natywny sterownik ARTS (głównie dla użytkowników KDE)</DD>
<DT>esd</DT>
<DD>natywny sterownik ESD (głównie dla użytkowników GNOME)</DD>
</DL>
<P>
Sterowniki kart dźwiękowych w Linuxie mają problemy z kompatybilnością.
Wynika to z tego że MPlayer polega na wbudowanych właściwościach <EM>prawidłowo</EM>
napisanych sterowników które pozwalają utrzymać prawidłową synchronizację audio/video.
Niestety, niektórzy autorzy sterowników nie wysilają się z zaprogramowaniem tych właściwości,
gdyż nie są wymagane do odgrywania plików MP3. </P>
<P>Other media players like <A HREF="http://avifile.sourceforge.net">aviplay</A>
or <A HREF="http://xine.sourceforge.net">xine</A> possibly work
out-of-the-box with these drivers because they use "simple" methods with
internal timing. Measuring showed that their methods are not as efficient
as MPlayer's. </P>
<P>Using MPlayer with a properly written audio driver will never result
in A/V desyncs related to the audio, except only with very badly created
files (check the man page for workarounds).</P>
<P>If you happen to have a bad audio driver, try the <CODE>-autosync</CODE>
option, it should sort out your problems. See the man page for detailed
information.</P>
<P>Some notes:</P>
<UL>
<LI>If you have an OSS driver, first try <CODE>-ao oss</CODE> (this is the
default). If you experience glitches, halts or anything out of the
ordinary, try <CODE>-ao sdl</CODE> (NOTE: You need to have SDL libraries
and header files installed). The SDL audio driver helps in a lot of cases
and also supports ESD (GNOME) and ARTS (KDE).</LI>
<LI>If you have ALSA version 0.5, then you almost always have to use
<CODE>-ao alsa5</CODE> , since ALSA 0.5 has buggy OSS emulation code, and
will <B>crash MPlayer</B> with a message like this:<BR>
<CODE>DEMUXER: Too many (945 in 8390980 bytes) video packets in the buffer!</CODE></LI>
<LI>On Solaris, use the SUN audio driver with the <CODE>-ao sun</CODE> option,
otherwise neither video nor audio will work.</LI>
<LI>If the sound clicks when playing from CD-ROM, turn on IRQ unmasking, e.g.
<CODE>hdparm -u1 /dev/cdrom</CODE> (<CODE>man hdparm</CODE>). This is
generally beneficial and described in more detail in the
<A HREF="cd-dvd.html#drives">CD-ROM section</A>.</LI>
</UL>
<H4><A NAME="experiences">2.3.2.2 Sound Card experiences, recommendations</A></H4>
<P>On Linux, a 2.4.x kernel is highly recommended. Kernel 2.2 is not tested.</P>
<P>Linux sound drivers are primarily provided by the free version of OSS. These
drivers have been superceded by <A HREF="http://www.alsa-project.org">ALSA</A>
(Advanced Linux Sound Architecture) in the 2.5 development series. If your
distribution does not already use ALSA you may wish to try their drivers if
you experience sound problems. ALSA drivers are generally superior to OSS in
compatibility, performance and features. But some sound cards are only
supported by the commercial OSS drivers from
<A HREF="http://www.opensound.com/">4Front Technologies</A>. They also support
several non-Linux systems.</P>
<TABLE BORDER="1" WIDTH="100%">
<TR>
<TH ROWSPAN="2"><B>SOUND CARD</B></TH>
<TH COLSPAN="4"><B>DRIVER</B></TH>
<TH ROWSPAN="2"><B>Max kHz</B></TH>
<TH ROWSPAN="2"><B>Max Channels</B></TH>
<TH ROWSPAN="2"><B>Max Opens<FONT SIZE="-2"><A HREF=#note1>[1]</A></FONT></B></TH>
</TR>
<TR>
<TH><B>OSS/Free</B></TH>
<TH><B>ALSA</B></TH>
<TH><B>OSS/Pro</B></TH>
<TH><B>other</B></TH>
</TR>
<TR>
<TD><B>VIA onboard (686/A/B, 8233, 8235)</B></TD>
<TD><A HREF="http://sourceforge.net/project/showfiles.php?group_id=3242&release_id=59602">via82cxxx_audio</A></TD>
<TD>snd-via82xx</TD>
<TD> </TD>
<TD> </TD>
<TD>4-48 kHz or 48 kHz only, depending on the chipset</TD>
<TD> </TD>
<TD> </TD>
</TR>
<TR>
<TD><B>Aureal Vortex 2</B></TD>
<TD>none</TD>
<TD>none</TD>
<TD>OK</TD>
<TD><A HREF="http://aureal.sourceforge.net">Linux Aureal Drivers</A><BR>
<A HREF="http://makacs.poliod.hu/~pontscho/aureal/au88xx-1.1.3.tar.bz2">buffer size increased to 32k</A></TD>
<TD>48</TD>
<TD>4.1</TD>
<TD>5+</TD>
</TR>
<TR>
<TD><B>SB Live!</B></TD>
<TD>Analog OK, SP/DIF not working</TD>
<TD>Both OK</TD>
<TD>Both OK</TD>
<TD><A HREF="http://opensource.creative.com">Creative's OSS driver (SP/DIF support)</A></TD>
<TD>192</TD>
<TD>4.0/5.1</TD>
<TD>32</TD>
</TR>
<TR>
<TD><B>SB 128 PCI (es1371)</B></TD>
<TD>OK</TD>
<TD>?</TD>
<TD> </TD>
<TD> </TD>
<TD>48</TD>
<TD>stereo</TD>
<TD>2</TD>
</TR>
<TR>
<TD><B>SB AWE 64</B></TD>
<TD>max 44kHz</TD>
<TD>48kHz sounds bad</TD>
<TD> </TD>
<TD> </TD>
<TD>48</TD>
<TD> </TD>
<TD> </TD>
</TR>
<TR>
<TD><B>GUS PnP</B></TD>
<TD>none</TD>
<TD>OK</TD>
<TD>OK</TD>
<TD> </TD>
<TD>48</TD>
<TD> </TD>
<TD> </TD>
</TR>
<TR>
<TD><B>Gravis UltraSound ACE</B></TD>
<TD>not OK</TD>
<TD>OK</TD>
<TD> </TD>
<TD> </TD>
<TD>44</TD>
<TD> </TD>
<TD> </TD>
</TR>
<TR>
<TD><B>Gravis UltraSound MAX</B></TD>
<TD>OK</TD>
<TD>OK (?)</TD>
<TD> </TD>
<TD> </TD>
<TD>48</TD>
<TD> </TD>
<TD> </TD>
</TR>
<TR>
<TD><B>ESS 688</B></TD>
<TD>OK</TD>
<TD>OK (?)</TD>
<TD> </TD>
<TD> </TD>
<TD>48</TD>
<TD> </TD>
<TD> </TD>
</TR>
<TR>
<TD><B>C-Media cards (which ones?)</B></TD>
<TD>not OK (hissing) (?)</TD>
<TD>OK</TD>
<TD> </TD>
<TD> </TD>
<TD>?</TD>
<TD> </TD>
<TD> </TD>
</TR>
<TR>
<TD><B>Yamaha cards (*ymf*)</B></TD>
<TD>not OK (?) (maybe <CODE>-ao sdl</CODE>)</TD>
<TD>OK only with ALSA 0.5 with OSS emulation <B>AND</B>
<CODE>-ao sdl</CODE> (!) (?)</TD>
<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>
<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>
</TABLE>
<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>
<LI>Measure the distance to the loudspeakers in meters in relation to your
listening position, giving you the distances s1 to s5 (for a 5.1 system).
There is no point in compensating for the sub-woofer (you will not hear the
difference anyway).</LI>
<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>
<H4><A NAME="af_volume">2.3.2.3.5 Software volume control</A></H4>
<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.
In case your sound card has a digital PCM mixer instead of an analog one, and
you hear distortion, use the MASTER mixer instead.
If there is an external amplifier connected to the computer (this is almost
always the case), the noise level can be minimized by adjusting the master
level and the volume knob on the amplifier until the hissing noise in the
background is gone. This filter has two options:</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 0dB.</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
option if the dynamic range of the loudspeakers is very low. Be aware that
this feature creates distortion and should be considered a last resort.</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>
<H4><A NAME="af_equalizer">2.3.2.3.6 Equalizer</A></H4>
<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>
<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
<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 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>
<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 cannot be used in MEncoder.</P>
<H4><A NAME="resample">2.3.2.4.1 Up/Downsampling</A></H4>
<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.
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>
<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
and cause audio distortion.</P>
<H4><A NAME="surround_decoding">2.3.2.4.2 Surround Sound decoding</A></H4>
<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>
<H4><A NAME="format">2.3.2.3.3 Sample format converter</A></H4>
<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 option, <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>
<H4><A NAME="delay">2.3.2.4.4 Delay</A></H4>
<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>
<P>If you have a file with a consistent A/V sync fault, use the <CODE>+/-</CODE>
keys to adjust timings on-the-fly instead. Usage of the OSD is recommended
to make this easier.</P>
<H4><A NAME="volume">2.3.2.4.5 Software volume control</A></H4>
<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
option <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.
In case your sound card has a digital PCM mixer instead of an analog one, and
you hear distortion, use the MASTER mixer instead.
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>
<H4><A NAME="extrastereo">2.3.2.4.6 Extrastereo</A></H4>
<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>
<H4><A NAME="normalizer">2.3.2.4.7 Volume normalizer</A></H4>
<P>This plugin maximizes the volume without distorting the sound.</P>
<P>Usage:<BR>
<CODE>mplayer media.avi -aop list=volnorm</CODE><BR>
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