MPlayer's audio interface is called libao2. It currently contains these drivers :
General: |
|||
oss | OSS (ioctl) driver | ||
sdl | SDL driver (supports up/downsampling, ESD, ARTS etc) | ||
nas | NAS (Network Audio System) driver | ||
alsa5 | native ALSA 0.5 driver | ||
alsa9 | native ALSA 0.9 driver (works, but has problems -> use OSS) | ||
sun | SUN audio driver (/dev/audio) for BSD and Solaris8 users |
The fact is, Linux soundcard drivers are usually bad, and always as incompatible as they can be. It MAY take a while to find your optimal settings.
DEMUXER: Too many (945 in 8390980 bytes) video packets in the buffer!
On Solaris/FreeBSD systems, use the SUN audio driver with the -ao sun option, otherwise you'll have no video, nor audio playing.
2.3.2.1. Soundcard experiences, recommendations
VIA onboard chipset (via82cxxx) 48Khz only | ||
Driver: | from here | |
Aureal Vortex 2 | ||
OSS: | no driver | |
OSS/Pro: | OK | |
ALSA: | no driver | |
Max kHz: | 48 | |
Driver: | aureal.sourceforge.net | |
Driver2: | from here (buffer size increased to 32k) | |
GUS PnP | ||
OSS: | no driver | |
OSS/Pro: | OK | |
ALSA: | OK | |
Max kHz: | 48 | |
SB Live! | ||
OSS: | HW AC3 won't work | |
ALSA: | OK | |
Max kHz: | 48 | |
SB AWE 64 | ||
OSS: | max 44kHz | |
ALSA: | 48kHz sounds shit | |
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 | |
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 followed by the
list=plugin1,plugin2,...
switch. The list
switch is
required and determines which plugins that 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 changes their behaviour. These switches are explained in detail in the sections below. A plugin will execute using default settings if it's switches are omitted. Example of how to use plugins in combination with plugin specific switches:
mplayer media.avi -aop
list=resample,format:fout=48000: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/down sampling of the sound. This plugin can
for example be used if you have a fixed frequency sound card or if you are
stuck with an olqd sound card that is only capable of max 44.1kHz.
Limitations in your hardware are not auto detected, so you have to specify
the sample frequency explicitly. 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>
2.3.2.2.2. 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 will need a sound-card supporting at least 4 channels.
Usage :
mplayer media.avi -aop list=surround
2.3.2.2.3. Sample format converter
If your sound card driver doesn't support signed 16bit int, 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. Observe 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 users perspective and is mentioned here for the sake of completeness only. Do not use this plugin unless you are a developer.
2.3.2.2.5. Software volume control
This plugin is a software replacement for the volume control, and
can be used in machines with broken mixer device. It can also be
used if one wants to change the output volume from 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 255. 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 mixers
master volume control. 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 increases (linearly) the difference between left and right channels (as the XMMS extrastereo plugin) which has some of "live" effect on 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 a mono sound (average of both
channels), if you set it to 1.0, sound will be unchanged.