New audio filter documentation by Anders Johannsson with some structural

modifications by myself.


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@8751 b3059339-0415-0410-9bf9-f77b7e298cf2

2 files changed, 365 insertions, 35 deletions

diff --git a/DOCS/documentation.html b/DOCS/documentation.html
index bd2999f36b..d0873bba10 100644
--- a/DOCS/documentation.html
+++ b/DOCS/documentation.html
@@ -185,17 +185,28 @@
             </UL>
           <LI><A HREF="sound.html">2.3.2 Audio output devices</A>
             <UL>
-              <LI><A HREF="sound.html#sync">2.3.2.1 Description of MPlayer's A/V sync method</A></LI>
+              <LI><A HREF="sound.html#sync">2.3.2.1 Audio/Video synchronisation</A></LI>
               <LI><A HREF="sound.html#experiences">2.3.2.2 Sound card experiences, recommendations</A></LI>
-              <LI><A HREF="sound.html#plugins">2.3.2.3 Audio plugins</A>
+              <LI><A HREF="sound.html#af">2.3.2.3 Audio filters</A>
                 <UL>
-                  <LI><A HREF="sound.html#resample">2.3.2.3.1 Up/Downsampling</A></LI>
-                  <LI><A HREF="sound.html#surround_decoding">2.3.2.3.2 Surround Sound decoding</A></LI>
-                  <LI><A HREF="sound.html#format">2.3.2.3.3 Sample format converter</A></LI>
-                  <LI><A HREF="sound.html#delay">2.3.2.3.4 Delay</A></LI>
-                  <LI><A HREF="sound.html#volume">2.3.2.3.5 Software volume control</A></LI>
-                  <LI><A HREF="sound.html#extrastereo">2.3.2.3.6 Extrastereo</A></LI>
-                  <LI><A HREF="sound.html#normalizer">2.3.2.3.7 Volume Normalizer</A></LI>
+                  <LI><A HREF="sound.html#af_resample">2.3.2.3.1 Up/Downsampling</A></LI>
+                  <LI><A HREF="sound.html#af_channels">2.3.2.3.2 Changing the number of channels</A></LI>
+                  <LI><A HREF="sound.html#af_format">2.3.2.3.3 Sample format converter</A></LI>
+                  <LI><A HREF="sound.html#af_delay">2.3.2.3.4 Delay</A></LI>
+                  <LI><A HREF="sound.html#af_volume">2.3.2.3.5 Software volume control</A></LI>
+                  <LI><A HREF="sound.html#af_equalizer">2.3.2.3.6 Equalizer</A></LI>
+                  <LI><A HREF="sound.html#af_panning">2.3.2.3.7 Panning filter</A></LI>
+                </UL>
+              </LI>
+              <LI><A HREF="sound.html#plugins">2.3.2.4 Audio plugins (deprecated)</A>
+                <UL>
+                  <LI><A HREF="sound.html#resample">2.3.2.4.1 Up/Downsampling</A></LI>
+                  <LI><A HREF="sound.html#surround_decoding">2.3.2.4.2 Surround Sound decoding</A></LI>
+                  <LI><A HREF="sound.html#format">2.3.2.4.3 Sample format converter</A></LI>
+                  <LI><A HREF="sound.html#delay">2.3.2.4.4 Delay</A></LI>
+                  <LI><A HREF="sound.html#volume">2.3.2.4.5 Software volume control</A></LI>
+                  <LI><A HREF="sound.html#extrastereo">2.3.2.4.6 Extrastereo</A></LI>
+                  <LI><A HREF="sound.html#normalizer">2.3.2.4.7 Volume Normalizer</A></LI>
                 </UL>
               </LI>
             </UL>
diff --git a/DOCS/sound.html b/DOCS/sound.html
index 8fb383fa4b..ac439cd070 100644
--- a/DOCS/sound.html
+++ b/DOCS/sound.html
@@ -12,12 +12,12 @@
 
 <H3><A NAME="audio">2.3.2 Audio output devices</A></H3>
 
-<H4><A NAME="sync">2.3.2.1 Description of MPlayer's A/V sync method</A></H4>
+<H4><A NAME="sync">2.3.2.1 Audio/Video synchronisation</A></H4>
 
 <P>MPlayer's audio interface is called <I>libao2</I>. It currently
   contains these drivers:</P>
 
-<TABLE BORDER=0>
+<TABLE BORDER="0">
   <TR><TD COLSPAN=4><P><B>General:</B></P></TD></TR>
   <TR><TD>&nbsp;&nbsp;</TD><TD VALIGN=top>oss</TD><TD>&nbsp;&nbsp;</TD><TD>OSS (ioctl) driver (supports hardware AC3 passthrough)</TD></TR>
   <TR><TD></TD><TD VALIGN=top>sdl</TD><TD></TD><TD>SDL driver (supports <B>ESD</B>, <B>ARTS</B> etc)</TD></TR>
@@ -29,17 +29,17 @@
 </TABLE>
 
 <P>Fact is, Linux sound card drivers have compatibility problems. The cause
-  is that MPlayer uses a feature of normally coded audio drivers to maintain
-  audio/video sync. Regrettably, some driver authors don't care of this
-  function: it isn't needed for playing MP3s, or sound effects.</P>
+  is that MPlayer uses a feature that well coded audio drivers implement to
+  maintain audio/video sync. Regrettably, some driver authors do not care about
+  this function, it is not needed for playing MP3s or for sound effects.</P>
 
 <P>Other media players like aviplay or xine possibly work out-of-the-box with
   these drivers because they use "simple" methods with internal timing. A note:
   time showed their methods aren't AS efficient as MPlayer's.</P>
 
-<P>Using MPlayer with a correctly written audio driver won't ever give you A/V
-  desyncs related to the audio, only with very badly created files (check the
-  documentation for workarounds!).</P>
+<P>With a correctly written audio driver MPlayer will never create audio related
+  A/V desynchronisation, unless your file is badly broken. Some options to work
+  around these problems are described in the man page).</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
@@ -50,9 +50,9 @@
 <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
+    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, ARTS. (ESD is the sound daemon
+    and also supports ESD and ARTS. (ESD is the sound daemon
     from GNOME, ARTS is from 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
@@ -66,9 +66,10 @@
 
 <H4><A NAME="experiences">2.3.2.2 Sound Card experiences, recommendations</A></H4>
 
-<TABLE BORDER=0 WIDTH="100%">
+<TABLE BORDER="0" WIDTH="100%">
   <TR><TD COLSPAN=3><B>VIA onboard chipset (via82cxxx) 48kHz only</B></TD></TR>
-  <TR><TD></TD><TD>Driver:</TD><TD> from <A HREF="http://sourceforge.net/project/showfiles.php?group_id=3242&amp;release_id=59602">sourceforge.net</A></TD></TR>
+  <TR><TD></TD><TD>Driver:</TD><TD> from the
+    <A HREF="http://sourceforge.net/project/showfiles.php?group_id=3242&amp;release_id=59602">gkernel project</A></TD></TR>
 
   <TR><TD COLSPAN=3><B>Aureal Vortex 2</B></TD></TR>
   <TR><TD>&nbsp;&nbsp;&nbsp;&nbsp;</TD><TD>OSS:</TD><TD>no driver</TD></TR>
@@ -135,10 +136,10 @@
 
 <P>On Linux, a 2.4.x kernel is highly recommended. Kernel 2.2 is not tested.</P>
 
-<P>If sound clicks when playing from CD-ROM, turn on IRQ unmasking, e.g. 
+<P>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 more detailed in the <A
-  HREF="cd-dvd.html#drives">CD-ROM section</A>.</P>
+  generally beneficial and described in more detail in the
+  <A HREF="cd-dvd.html#drives">CD-ROM section</A>.</P>
 
 <P>Sharing your sound card with another application like XMMS is <B>strongly
   discouraged</B>!  If the other sound application is using ESD, start
@@ -150,7 +151,325 @@
   and your sound card(s) worked together.</P>
 
 
-<H4><A NAME="plugins">2.3.2.3 Audio plugins</A></H4>
+<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>
+  switch 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>
+  &nbsp;&nbsp;<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 switches that change their behavior. These switches
+  are explained in detail in the sections below. A filter will execute using
+  default settings if its switches are omitted. Here is an example of how to use
+  filters in combination with filter specific switches:</P>
+
+<P>&nbsp;&nbsp;<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>Most filters respond to the <CODE>-v</CODE> switch, which makes the filters
+  print out status messages.</P>
+
+<P>The overall execution of the filter layer is controlled using the
+  <CODE>-af-adv</CODE> switch. This switch has two suboptions:</P>
+
+<DL>
+  <DT><CODE>force</CODE><DT>
+  <DD>is an integer between 0 and 3 that controls how the filters are inserted
+    and what speed/accuracy optimizations they use:
+    <DL>
+      <DT>0</DT>
+      <DD>Use automatic insertion of filters and optimize according to CPU
+        speed.</DD>
+      <DT>1</DT>
+      <DD>Use automatic insertion of filters and optimize for the highest speed.
+        If this option is set the processing of the audio data will be done
+        using fix point arithmetics. Warning: Some features in the audio filters
+        will silently fail, and the sound quality may drop.</DD>
+      <DT>2</DT>
+      <DD>Use automatic insertion of filters and optimize for quality. If this
+        option is set the processing of the audio data will be done using
+        floating point instructions and is therefore quite CPU intensive, but
+        gives a lot higher sound quality than fix point processing.</DD>
+      <DT>3</DT>
+      <DD>Use no automatic insertion of filters and no optimization. Warning: It
+        may be possible to crash MPlayer using this setting.</DD>
+    </DL>
+
+  </DD>
+  
+  <DT><CODE>list</CODE></DT>
+  <DD>is an alias for the -af switch.</DD>
+</DL>
+
+
+<H5><A NAME="af_resample">2.3.2.3.1 Up/Down-sampling</A></H5>
+
+<P>MPlayer fully supports sound up/down-sampling. This filter 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 switches:</P>
+
+<DL>
+  <DT><CODE>srate</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 switch can be
+    used if the startup of the playback is extremely slow.</DD>
+
+  <DT><CODE>fast</CODE><DT>
+  <DD>is an optional binary parameter that enables linear interpolation as
+    resampling method. Linear interpolation is extremely fast, but suffers from
+    poor sound quality especially when used for up-sampling.</DD>
+</DL>
+
+<P>Example:<BR>
+  &nbsp;&nbsp;<CODE>mplayer -af resample=44100:0:1</CODE></P>
+
+<P>would set the output frequency of the resample filter to 44100Hz using exact
+  output frequency scaling and linear interpolation.</P>
+
+
+<H5><A NAME="af_channels">2.3.2.3.2 Changing the number of channels</A></H5>
+
+<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 switches is dynamic:</P>
+
+<DL>
+  <DT><CODE>nch</CODE></DT>
+  <DD>is an integer between 1 and 6 that is used for setting the number of
+    output channels. This switch is required, leaving it empty results in a
+    runtime error.</DD>
+
+  <DT><CODE>nr</CODE></DT>
+  <DD>is an integer between 1 and 6 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 0 and 5 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>
+  &nbsp;&nbsp;<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>
+  &nbsp;&nbsp;<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>
+
+
+<H5><A NAME="af_format">2.3.2.3.3 Sample format converter</A></H5>
+
+<P>This filter is a sample format converter. It is automatically enabled when
+  needed by the sound card or another filter.</P>
+
+<DL>
+  <DT><CODE>bps</CODE></DT>
+  <DD>can be 1, 2 or 4 and denotes the number of bytes per sample. This switch
+    is required, leaving it empty results in a runtime error.</DD>
+
+  <DT><CODE>f</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 switch is required, leaving it
+    empty results in a runtime error.</DD>
+</DL>
+
+<P>Example:<BR>
+  &nbsp;&nbsp;<CODE>mplayer media.avi -af format=4:float</CODE></P>
+
+<P>would set the output output format to 4 bytes per sample floating point
+  data.</P>
+
+
+<H5><A NAME="af_delay">2.3.2.3.4 Delay</A></H5>
+
+<P>This filter delays the sound to the loudspeakers in order to make the sound
+  in the different channels arrive at the same time to the listening position.
+  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>
+  &nbsp;&nbsp;<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>This filter is a software volume control. 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</CODE></DT>
+  <DD>is a floating point number between -200 and +60 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>
+  &nbsp;&nbsp;<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> This filter is 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 -12 to +12dB representing the gain in
+    dB for each frequency band.</DD>
+</DL>
+
+<P>Example:<BR>
+  &nbsp;&nbsp;<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>This filter can be used for mixing the 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</CODE></DT>
+  <DD>is an integer between 1 and 6 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 0 and 1 that determine the level
+    <CODE>l[i][j]</CODE> that the input channel j is mixed into output channel
+    i.</DD>
+</DL>
+
+<P>Example:<BR>
+  &nbsp;&nbsp;<CODE>mplayer -af pan=1:0.5:0.5 -channels 1 media.avi</CODE></P>
+
+<P>would down-mix from stereo to mono.</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
@@ -173,17 +492,17 @@
   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>                         
+  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">2.3.2.3.1 Up/Downsampling</A></H5>
+<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 neccessary or not, is <B>autodetected</B>.
+  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
@@ -198,7 +517,7 @@
   in addition to audio distortion.</P>
 
 
-<H5><A NAME="surround_decoding">2.3.2.3.2 Surround Sound decoding</A></H5>
+<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.
@@ -210,8 +529,8 @@
 
 
 <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, 
+
+<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
@@ -224,7 +543,7 @@
   list=format:format=&lt;required output format&gt;</CODE></P>
 
 
-<H5><A NAME="delay">2.3.2.3.4 Delay</A></H5>
+<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
@@ -232,7 +551,7 @@
   plugin unless you are a developer.</P>
 
 
-<H5><A NAME="volume">2.3.2.3.5 Software volume control</A></H5>
+<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
@@ -265,7 +584,7 @@
   list=volume:softclip</CODE></P>
 
 
-<H5><A NAME="extrastereo">2.3.2.3.6 Extrastereo</A></H5>
+<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"
@@ -281,7 +600,7 @@
   -1.0, left and right channels will be swapped.</P>
 
 
-<H5><A NAME="normalizer">2.3.2.3.7 Volume normalizer</A></H5>
+<H5><A NAME="normalizer">2.3.2.4.7 Volume normalizer</A></H5>
 
 <P>This plugin maximizes the volume without distorting the sound.</P>