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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
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diego 2003-01-03 22:29:16 +00:00
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@ -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>

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@ -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>