MPlayer - The Movie Player: http://www.mplayerhq.hu | ||
---|---|---|
Prev | Chapter 2. Features | Next |
MPlayer's audio interface is called libao2. It currently contains these drivers:
Driver | Comment |
---|---|
oss | OSS (ioctl) driver (supports hardware AC3 passthrough) |
sdl | SDL driver (supports sound daemons like ESD and ARTS) |
nas | NAS (Network Audio System) driver |
alsa5 | native ALSA 0.5 driver |
alsa | native ALSA 0.9/1.0 driver (supports hardware AC3 passthrough) |
sun | SUN audio driver (/dev/audio) for BSD and Solaris8 users |
macosx | native Mac OS X driver |
win32 | native Win32 driver |
arts | native ARTS driver (mostly for KDE users) |
esd | native ESD driver (mostly for GNOME users) |
jack | JACK (Jack Audio Connection Kit) driver |
polyp | polypaudio driver |
Linux sound card drivers have compatibility problems. This is because MPlayer relies on an in-built feature of properly coded sound drivers that enable them to maintain correct audio/video sync. Regrettably, some driver authors don't take the care to code this feature since it is not 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. Measuring showed that their methods are not as efficient as MPlayer's.
Using MPlayer with a properly written audio driver will never result in A/V desyncs related to the audio, except only with very badly created files (check the man page 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:
If you have an OSS driver, first try -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 (GNOME) and ARTS (KDE).
If you have ALSA version 0.5, then you almost always have to use
-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.
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.
On Linux, a 2.4.x kernel is highly recommended. Kernel 2.2 is not tested.
Linux sound drivers are primarily provided by the free version of OSS. These drivers have been superseded by ALSA (Advanced Linux Sound Architecture) in the 2.5 development series. If your distribution does not already use ALSA you may wish to try their drivers if you experience sound problems. ALSA drivers are generally superior to OSS in compatibility, performance and features. But some sound cards are only supported by the commercial OSS drivers from 4Front Technologies. They also support several non-Linux systems.
SOUND CARD | DRIVER | Max kHz | Max Channels | Max Opens [a] | |||
---|---|---|---|---|---|---|---|
OSS/Free | ALSA | OSS/Pro | other | ||||
VIA onboard (686/A/B, 8233, 8235) | via82cxxx_audio | snd-via82xx | 4-48 kHz or 48 kHz only, depending on the chipset | ||||
Aureal Vortex 2 | none | none | OK | Linux Aureal Drivers buffer size increased to 32k | 48 | 4.1 | 5+ |
SB Live! | Analog OK, S/PDIF not working | Both OK | Both OK | Creative's OSS driver (S/PDIF support) | 192 | 4.0/5.1 | 32 |
SB 128 PCI (es1371) | OK | ? | 48 | stereo | 2 | ||
SB AWE 64 | max 44kHz | 48kHz sounds bad | 48 | ||||
GUS PnP | none | OK | OK | 48 | |||
Gravis UltraSound ACE | |||||||
Gravis UltraSound MAX | OK | OK (?) | 48 | ||||
ESS 688 | OK | OK (?) | 48 | ||||
C-Media cards (CMI8338/8738) | OK | OK S/PDIF is supported with ALSA 0.9.x | ? | 44 | stereo | 1 | |
Yamaha cards (*ymf*) | not OK (?) (maybe -ao sdl ) | OK only with ALSA 0.5 with OSS emulation
AND -ao sdl (!) (?) | |||||
Cards with envy24 chips (like Terratec EWS88MT) | ? | ? | OK | ? | |||
PC Speaker or DAC | OK | none | Linux PC speaker OSS driver | The driver emulates 44.1, maybe more. | mono | 1 | |
Notes: a. the number of applications that are able to use the device at the same time. |
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,...
option 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.aviwould 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 options that change their behavior. These options are explained in detail in the sections below. A filter will execute using default settings if its options are omitted. Here is an example of how to use filters in combination with filter specific options:
mplayer -af resample=11025,pan=1:0.5:0.5 -channels 1 -srate 11025 media.aviwould set the output frequency of the resample filter to 11025Hz and downmix the audio to 1 channel using the pan filter.
The overall execution of the filter layer is controlled using the
-af-adv
option. This option has two suboptions:
force
is a bit field that controls how the filters
are inserted and what speed/accuracy optimizations they use:
0
Use automatic insertion of filters and optimize according to CPU speed.
1
Use automatic insertion of filters and optimize for the highest speed. Warning: Some features in the audio filters may silently fail, and the sound quality may drop.
2
Use automatic insertion of filters and optimize for quality.
3
Use no automatic insertion of filters and no optimization. Warning: It may be possible to crash MPlayer using this setting.
4
Use automatic insertion of filters according to 0 above, but use floating point processing when possible.
5
Use automatic insertion of filters according to 1 above, but use floating point processing when possible.
6
Use automatic insertion of filters according to 2 above, but use floating point processing when possible.
7
Use no automatic insertion of filters according to 3 above, and use floating point processing when possible.
list
is an alias for the -af option.
The filter layer is also affected by the following generic options:
-v
Increases the verbosity level and makes most filters print out extra status messages.
-channels
This option sets the number of output channels you would like your sound card to use. It also affects the number of channels that are being decoded from the media. If the media contains less channels than requested the channels filter (see below) will automatically be inserted. The routing will be the default routing for the channels filter.
-srate
This option selects the sample rate you would like your sound card to use (of course the cards have limits on this). If the sample frequency of your sound card is different from that of the current media, the resample filter (see below) will be inserted into the audio filter layer to compensate for the difference.
-format
This option sets the sample format between the audio filter layer and the sound card. If the requested sample format of your sound card is different from that of the current media, a format filter (see below) will be inserted to rectify the difference.
MPlayer fully supports sound up/down-sampling through the
resample
filter. It 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 options:
srate <8000-192000>
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.
sloppy
is an optional binary parameter that allows the output frequency to differ
slightly from the frequency given by srate
. This option
can be used if the startup of the playback is extremely slow. It is enabled
by default.
type <0-2>
is an optional integer between 0 and 2 that selects which resampling method to use. Here 0 represents linear interpolation as resampling method, 1 represents resampling using a poly-phase filter-bank and integer processing and 2 represents resampling using a poly-phase filter-bank and floating point processing. Linear interpolation is extremely fast, but suffers from poor sound quality especially when used for up-sampling. The best quality is given by 2 but this method also suffers from the highest CPU load.
Example:
mplayer -af resample=44100:0:0would 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 options is dynamic:
nch <1-6>
is an integer between 1 and 6 that is used for setting the number of output channels. This option is required, leaving it empty results in a runtime error.
nr <1-6>
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.
from1:to1:from2:to2:from3:to3...
are pairs of numbers between 0 and 5 that define where each channel should be routed.
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.aviwould 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.aviwould 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.
The format
filter converts between different sample formats. It
is automatically enabled when needed by the sound card or another filter.
bps <number>
can be 1, 2 or 4 and denotes the number of bytes per sample. This option is required, leaving it empty results in a runtime error.
f <format>
is a text string describing the sample format. The string is a
concatenated mix of: alaw
, mulaw
or
imaadpcm
, float
or int
,
unsigned
or signed
, le
or
be
(little- or big-endian). This option is required,
leaving it empty results in a runtime error.
Example:
mplayer -af format=4:float media.aviwould set the output format to 4 bytes per sample floating point data.
The delay
filter delays the sound to the loudspeakers such that
the sound from the different channels arrives at the listening position
simultaneously.
It is only useful if you have more than 2 loudspeakers. This filter has a
variable number of parameters:
d1:d2:d3...
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.
To calculate the required delay for the different channels do as follows:
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).
Subtract the distances s1 to s5 from the maximum distance i.e. s[i] = max(s) - s[i]; i = 1...5
Calculate the required delays in ms as d[i] = 1000*s[i]/342; i = 1...5
Example:
mplayer -af delay=10.5:10.5:0:0:7:0 media.aviwould delay front left and right by 10.5ms, the two rear channels and the sub by 0ms and the center channel by 7ms.
Software volume control is implemented by the volume
audio filter. 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. In case your sound card
has a digital PCM mixer instead of an analog one, and you hear distortion,
use the MASTER mixer instead. 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 options:
v <-200 - +60>
is a floating point number between -200 and +60 which represents the volume level in dB. The default level is 0dB.
c
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 option if the dynamic range of the loudspeakers is very low. Be aware that this feature creates distortion and should be considered a last resort.
Example:
mplayer -af volume=10.1:0 media.aviwould 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.
The equalizer
filter represents 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
are floating point numbers between -12 and +12 representing the gain in dB for each frequency band.
Example:
mplayer -af equalizer=11:11:10:5:0:-12:0:5:12:12 media.aviwould amplify the sound in the upper and lower frequency region while canceling it almost completely around 1kHz.
Use the pan
filter to mix channels arbitrarily. It is
basically a combination of the volume control and the channels filter.
There are two major uses for this filter:
Down-mixing many channels to only a few, stereo to mono for example.
Varying the "width" of the center speaker in a surround sound system.
This filter is hard to use, and will require some tinkering before the desired result is obtained. The number of options for this filter depends on the number of output channels:
nch <1-6>
is an integer between 1 and 6 and is used for setting the number of input channels. This option is required, leaving it empty results in a runtime error.
l00:l01:l02:..l10:l11:l12:...ln0:ln1:ln2:...
are floating point values between 0 and 1.
l[i][j]
determines how much of input channel j is mixed into
output channel i.
Example 1:
mplayer -af pan=1:0.5:0.5 -channels 1 media.aviwould down-mix from stereo to mono.
Example 2:
mplayer -af pan=3:1:0:1:0.5:0.5 -channels 3 media.aviwould give 3 channel output leaving channels 0 and 1 intact, and mix channels 0 and 1 into output channel 2 (which could be sent to a sub-woofer for example).
The sub
filter adds a sub woofer channel to the audio
stream. The audio data used for creating the sub-woofer channel is an
average of the sound in channel 0 and channel 1. The resulting sound is
then low-pass filtered by a 4th order Butterworth filter with a default
cutoff frequency of 60Hz and added to a separate channel in the audio
stream. Warning: Disable this filter when you are playing DVDs with Dolby
Digital 5.1 sound, otherwise this filter will disrupt the sound to the
sub-woofer. This filter has two parameters:
fc <20-300>
is an optional floating point number used for setting the cutoff frequency for the filter in Hz. The valid range is 20Hz to 300Hz. For the best result try setting the cutoff frequency as low as possible. This will improve the stereo or surround sound experience. The default cutoff frequency is 60Hz.
ch <0-5>
is an optional integer between 0 and 5 which determines the channel number in which to insert the sub-channel audio. The default is channel number 5. Observe that the number of channels will automatically be increased to ch if necessary.
Example:
mplayer -af sub=100:4 -channels 5 media.aviwould add a sub-woofer channel with a cutoff frequency of 100Hz to output channel 4.
Matrix encoded surround sound can be decoded by the surround
filter. 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. This filter has
one parameter:
d <0-1000>
is an optional floating point number between 0 and 1000 used for setting the delay time in ms for the rear speakers. This delay should be set as follows: if d1 is the distance from the listening position to the front speakers and d2 is the distance from the listening position to the rear speakers, then the delay d should be set to 15ms if d1 <= d2 and to 15 + 5*(d1-d2) if d1 > d2. The default value for d is 20ms.
Example:
mplayer -af surround=15 -channels 4 media.aviwould add surround sound decoding with 15ms delay for the sound to the rear speakers.
This audio filter exports the incoming signal to other processes using memory mapping (mmap()). Memory mapped areas contain a header:
int nch /*number of channels*/ int size /*buffer size*/ unsigned long long counter /*Used to keep sync, it's updated every time new data is exported.*/The rest is payload (non-interleaved) 16bit data.
mmapped_file
The file you want this filter to export to. The default is to map to ~/.mplayer/mplayer-af_export.
nsamples
Number of samples per channel. The default is 512 samples.
Example:
mplayer -af export=/tmp/mplayer-af_export:1024 media.aviwould export 1024 samples per channel to /tmp/mplayer-af_export.
This audio filter (linearly) increases the difference between left and right channels (like the XMMS extrastereo plugin) which adds some sort of "live" effect to playback. This filter has one parameter:
mul
is the difference coefficient, an optional floating point 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.
Usage:
mplayer -af extrastereo media.avi mplayer -af extrastereo=3.45 media.avi
This audio filter maximizes the volume without distorting the sound.
Usage:
mplayer -af volnorm media.avi
Note: Audio plugins have been deprecated by audio filters and will be removed soon.
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,formatwould 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=0x8would set the output frequency of the resample plugin to 44100 Hz 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 44.1 kHz. 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 48 kHz, and is given in
Hz.
Usage:
mplayer media.avi -aop list=resample:fout=freqwhere freq is the frequency in Hz, like 44100.
Note: 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 16-bit 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=outfmtwhere outfmt is the 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=3.45The 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