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 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.
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.
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).
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 and 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 the gkernel project | |
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 the 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 in more detail 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.
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 -af filter1,filter2,...
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.
Example:
mplayer -af resample,pan movie.avi
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.
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:
mplayer -af resample=11025,pan=1:0.5:0.5 -channels 1
-srate 11025 media.avi
would set the output frequency of the resample filter to 11025Hz and downmix the audio to 1 channel using the pan filter.
Most filters respond to the -v
switch, which makes the filters
print out status messages.
The overall execution of the filter layer is controlled using the
-af-adv
switch. This switch has two suboptions:
force
list
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:
srate
sloppy
srate
. This switch can be
used if the startup of the playback is extremely slow.fast
Example:
mplayer -af resample=44100:0:1
would set the output frequency of the resample filter to 44100Hz using exact output frequency scaling and linear interpolation.
The channels
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:
nch
nr
from1:to1:from2:to2:from3:to3...
If only nch
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.
Example 1:
mplayer -af channels=4:4:0:1:1:0:2:2:3:3 media.avi
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.
Example 2:
mplayer -af channels=6:4:0:0:0:1:0:2:0:3 media.avi
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.
This filter is a sample format converter. It is automatically enabled when needed by the sound card or another filter.
bps
f
alaw
, mulaw
or
imaadpcm
, float
or int
,
unsigned
or signed
, le
or
be
(little or big endian). This switch is required, leaving it
empty results in a runtime error.Example:
mplayer media.avi -af format=4:float
would set the output output format to 4 bytes per sample floating point data.
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:
d1:d2:d3...
To calculate the required delay for the different channels do as follows:
Example:
mplayer -af delay=10.5:10.5:0:0:7:0 media.avi
would delay front left and right by 10.5ms, the two rear channels and the sub by 0ms and the center channel by 7ms.
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:
v
c
Example:
mplayer -af volume=10.1:0 media.avi
would amplify the sound by 10.1dB and hard-clip if the sound level is too high.
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.
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:
Band No. | Center frequency |
0 | 31.25 Hz |
1 | 62.50 Hz |
2 | 125.0 Hz |
3 | 250.0 Hz |
4 | 500.0 Hz |
5 | 1.000 kHz |
6 | 2.000 kHz |
7 | 4.000 kHz |
8 | 8.000 kHz |
9 | 16.00 kHz |
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.
This filter has 10 parameters:
g1:g2:g3...g10
Example:
mplayer -af equalizer=11:11:10:5:0:-12:0:5:12:12 media.avi
would amplify the sound in the upper and lower frequency region while canceling it almost completely around 1kHz.
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:
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:
nch
l00:l01:l02:..l10:l11:l12:...ln0:ln1:ln2:...
l[i][j]
that the input channel j is mixed into output channel
i.Example:
mplayer -af pan=1:0.5:0.5 -channels 1 media.avi
would down-mix from stereo to mono.
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 necessary 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