2.3.2 Audio output devices

2.3.2.1 Audio/Video synchronisation

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

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)

esd

native ESD driver (mostly for GNOME 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:

• 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 and ARTS. (ESD is the sound daemon from GNOME, ARTS is from 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.

2.3.2.2 Sound Card experiences, recommendations

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.

Feedback to this document is welcome. Please tell us how MPlayer and your sound card(s) worked together.

2.3.2.3 Audio filters

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.

The overall execution of the filter layer is controlled using the -af-adv switch. This switch 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 switch.

The filter layer is also affected by the following generic switches:

-v

Increases the verbosity level and makes most filters print out extra status messages.

-channels

This option sets the number of output channels your sound card is using. 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 of your sound card. 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 of 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.

2.3.2.3.1 Up/Down-sampling

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 switches:

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 switch 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=44100:0:0

would set the output frequency of the resample filter to 44100Hz using exact output frequency scaling and linear interpolation.

2.3.2.3.2 Changing the number of channels

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 <1-6>

is an integer between 1 and 6 that is used for setting the number of output channels. This switch 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.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.

2.3.2.3.3 Sample format converter

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 switch 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 switch is required, leaving it empty results in a runtime error.

Example:
  mplayer -af format=4:float media.avi

would set the output format to 4 bytes per sample floating point data.

2.3.2.3.4 Delay

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:

1. 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).
2. Subtract the distances s1 to s5 from the maximum distance i.e.
   s[i] = max(s) - s[i]; i = 1...5
3. Calculated 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.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.

2.3.2.3.5 Software volume control

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. 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 <-200 - +60>

is a floating point number between -200 and +60 which represents the volume level in dB. The default level is -10dB.

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 switch 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.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.

2.3.2.3.6 Equalizer

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:

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.