MPlayer's audio interface is called libao2. It currently contains these drivers:
General: | |||
oss | OSS (ioctl) driver (supports hardware AC3 passthrough) | ||
sdl | SDL driver (supports ESD, ARTS etc) | ||
nas | NAS (Network Audio System) driver | ||
alsa5 | native ALSA 0.5 driver | ||
alsa9 | native ALSA 0.9 driver (supports hardware AC3 passthrough) | ||
sun | SUN audio driver (/dev/audio ) for BSD and Solaris8 users | ||
arts | native ARTS driver (mostly for KDE users) |
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.
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.
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!).
If you happen to have a bad audio driver, try the -autosync
option, it should sort out your problems. See the man page for detailed
information.
Some notes:
-ao oss
(this is the
default). If you experience glitches, halts or anything out of the
ordinary, try -ao sdl
(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
from GNOME, ARTS is from KDE.)-ao alsa5
, since ALSA 0.5 has buggy OSS emulation code, and
will crash MPlayer with a message like this:DEMUXER: Too many (945 in 8390980 bytes) video packets in the buffer!
On Solaris, use the SUN audio driver with the
-ao sun
option, otherwise neither video nor audio will work.
VIA onboard chipset (via82cxxx) 48kHz only | ||
Driver: | from sourceforge.net | |
Aureal Vortex 2 | ||
OSS: | no driver | |
OSS/Pro: | OK | |
ALSA: | no driver | |
Max kHz: | 48 | |
Driver: | aureal.sourceforge.net | |
Driver2: | from Pontscho's page (buffer size increased to 32k) | |
GUS PnP | ||
OSS: | no driver | |
OSS/Pro: | OK | |
ALSA: | OK | |
Max kHz: | 48 | |
SB Live! | ||
OSS: | Analog OK, SP/DIF not working | |
ALSA: | Both OK | |
Max kHz: | 192 | |
SB AWE 64 | ||
OSS: | max 44kHz | |
ALSA: | 48kHz sounds bad | |
Max kHz: | 48 | |
Gravis UltraSound ACE | ||
OSS: | not OK | |
ALSA: | OK | |
Max kHz: | 44 | |
Gravis UltraSound MAX | ||
OSS: | OK | |
ALSA: | OK (?) | |
Max kHz: | 48 | |
ESS 688 | ||
OSS: | OK | |
ALSA: | OK (?) | |
Max kHz: | 48 | |
C-Media cards (which ones?) | ||
OSS: | not OK (hissing) (?) | |
ALSA: | OK (?) | |
Max kHz: | ? | |
Yamaha cards (*ymf*) | ||
OSS: | not OK (?) (maybe -ao sdl ) | |
ALSA: | OK only with ALSA 0.5 with OSS emulation AND -ao sdl (!) (?) | |
Max kHz: | ? | |
Cards with envy24 chips (like Terratec EWS88MT) | ||
OSS: | ? | |
OSS/Pro: | OK | |
ALSA: | ? | |
Max kHz: | ? | |
PC Speaker or DAC | ||
OSS: | OK (Use the SDL driver: -ao sdl ) | |
ALSA: | no driver | |
Max kHz: | The driver emulates 44.1, maybe more. | |
Driver: | ftp://ftp.infradead.org/pub/pcsp |
On Linux, a 2.4.x kernel is highly recommended. Kernel 2.2 is not tested.
If sound clicks when playing from CD-ROM, turn on IRQ unmasking, e.g.
hdparm -u1 /dev/cdrom
(man hdparm
). This is
generally beneficial and described more detailed in the CD-ROM section.
Sharing your sound card with another application like XMMS is strongly
discouraged! If the other sound application is using ESD, start
MPlayer with the -vo sdl:esd
option to combine both
sound streams. In fact, the option -vo sdl:esd
could be used
with ESD even when playing MPlayer alone.
Feedback to this document is welcome. Please tell us how MPlayer and your sound card(s) worked together.
MPlayer has support for audio plugins. Audio plugins can be used for
changing the properties of the audio data before the sound reaches the sound
card. They are enabled using the -aop
switch which takes a
list=plugin1,plugin2,...
argument. The list
argument
is required and determines which plugins should be used and in which order they
should be executed. Example:
mplayer media.avi -aop list=resample,format
would run the sound through the resampling plugin followed by the format plugin.
The plugins can also have switches that change their behavior. These switches are explained in detail in the sections below. A plugin will execute using default settings if its switches are omitted. Here is an example of how to use plugins in combination with plugin specific switches:
mplayer media.avi -aop
list=resample,format:fout=44100:format=0x8
would set the output frequency of the resample plugin to 44100Hz and the output format of the format plugin to AFMT_U8.
Currently audio plugins can not be used in MEncoder.
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 autodetected.
This plugin has one switch:
fout
which is used for setting the desired output sample
frequency. It defaults to 48kHz, and is given in
<Hz>.
Usage:
mplayer media.avi -aop list=resample:fout=<required
frequency in Hz, like 44100>
Note that the output frequency should not be scaled up from the default value. Scaling up will cause the audio and video streams to be played in slow motion in addition to audio distortion.
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.
Usage:
mplayer media.avi -aop list=surround
If your sound card driver does not support signed 16bit int
data type,
this plugin can
be used to change the format to one which your sound card can understand. It
has one switch, format
, which can be set to one of the numbers
found in libao2/afmt.h
. 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.
Usage:
mplayer media.avi -aop
list=format:format=<required output format>
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.
This plugin is a software replacement for the volume control, and
can be used on machines with a broken mixer device. It can also be
used if one wants to change the output volume of MPlayer
without changing the PCM volume setting in the mixer. It has one
switch volume
that is used for setting the initial
sound level. The initial sound level can be set to values between 0
and 255 and defaults to 101 which equals 0dB amplification. Use this
plugin with caution since it can reduce the signal to noise ratio of
the sound. In most cases it is best to set the level for the PCM
sound to max, leave this plugin out and control the output level to
your speakers with the master volume control of the mixer. If there is an
external amplifier connected to the computer (this is almost always
the case), the noise level can be minimized by adjusting the master
level and the volume knob on the amplifier until the hissing noise
in the background is gone.
Usage:
mplayer media.avi -aop
list=volume:volume=<0-255>
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.
Usage:
mplayer media.avi -aop
list=volume:softclip
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.
Usage:
mplayer media.avi -aop list=extrastereo
mplayer media.avi -aop list=extrastereo:mul=3.45
The default coefficient (mul
) 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.
This plugin maximizes the volume without distorting the sound.
Usage:
mplayer media.avi -aop list=volnorm