2.3.2 Audio output devices

2.3.2.1 Description of MPlayer's A/V sync method

MPlayer's audio interface is called libao2. It currently contains these drivers:

General:

  oss  OSS (ioctl) driver (supports hardware AC3 passthrough)
sdlSDL driver (supports ESD, ARTS etc)
nasNAS (Network Audio System) driver
alsa5native ALSA 0.5 driver
alsa9native ALSA 0.9 driver (supports hardware AC3 passthrough)
sunSUN audio driver (/dev/audio) for BSD and Solaris8 users
artsnative 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:

On Solaris, use the SUN audio driver with the -ao sun option, otherwise neither video nor audio will work.

2.3.2.2 Sound Card experiences, recommendations

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.

2.3.2.3 Audio plugins

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.

2.3.2.3.1 Up/Downsampling

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.

2.3.2.3.2 Surround Sound decoding

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

2.3.2.3.3 Sample format converter

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>

2.3.2.3.4 Delay

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.

2.3.2.3.5 Software volume control

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

2.3.2.3.6 Extrastereo

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.

2.3.2.3.7 Volume normalizer

This plugin maximizes the volume without distorting the sound.

Usage:
  mplayer media.avi -aop list=volnorm