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mpv/DOCS/man/ao.rst
2015-05-05 14:40:26 +02:00

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AUDIO OUTPUT DRIVERS
====================
Audio output drivers are interfaces to different audio output facilities. The
syntax is:
``--ao=<driver1[:suboption1[=value]:...],driver2,...[,]>``
Specify a priority list of audio output drivers to be used.
If the list has a trailing ',', mpv will fall back on drivers not contained
in the list. Suboptions are optional and can mostly be omitted.
You can also set defaults for each driver. The defaults are applied before the
normal driver parameters.
``--ao-defaults=<driver1[:parameter1:parameter2:...],driver2,...>``
Set defaults for each driver.
.. note::
See ``--ao=help`` for a list of compiled-in audio output drivers. The
driver ``--ao=alsa`` is preferred. ``--ao=pulse`` is preferred on systems
where PulseAudio is used. On Windows, ``--ao=wasapi`` is preferred,
though it might cause trouble sometimes, in which case ``--ao=dsound``
should be used. On BSD systems, ``--ao=oss`` or `--ao=sndio`` may work
(the latter being experimental). On OS X systems, use ``--ao=coreaudio``.
.. admonition:: Examples
- ``--ao=alsa,oss,`` Try the ALSA driver, then the OSS driver, then others.
- ``--ao=alsa:resample=yes:device=[plughw:0,3]`` Lets ALSA resample and
sets the device-name as first card, fourth device.
Available audio output drivers are:
``alsa`` (Linux only)
ALSA audio output driver
``device=<device>``
Sets the device name. For ac3 output via S/PDIF, use an "iec958" or
"spdif" device, unless you really know how to set it correctly.
``resample=yes``
Enable ALSA resampling plugin. (This is disabled by default, because
some drivers report incorrect audio delay in some cases.)
``mixer-device=<device>``
Set the mixer device used with ``--no-softvol`` (default: ``default``).
``mixer-name=<name>``
Set the name of the mixer element (default: ``Master``). This is for
example ``PCM`` or ``Master``.
``mixer-index=<number>``
Set the index of the mixer channel (default: 0). Consider the output of
"``amixer scontrols``", then the index is the number that follows the
name of the element.
``non-interleaved``
Allow output of non-interleaved formats (if the audio decoder uses
this format). Currently disabled by default, because some popular
ALSA plugins are utterly broken with non-interleaved formats.
``ingore-chmap``
Don't read or set the channel map of the ALSA device - only request the
required number of channels, and then pass the audio as-is to it. This
option most likely should not be used. It can be useful for debugging,
or for static setups with a specially engineered ALSA configuration (in
this case you should always force the same layout with ``--audio-channels``,
or it will work only for files which use the layout implicit to your
ALSA device).
.. note::
MPlayer and mplayer2 required you to replace any ',' with '.' and
any ':' with '=' in the ALSA device name. mpv does not do this anymore.
Instead, quote the device name:
``--ao=alsa:device=[plug:surround50]``
Note that the ``[`` and ``]`` simply quote the device name. With some
shells (like zsh), you have to quote the option string to prevent the
shell from interpreting the brackets instead of passing them to mpv.
Actually, you should use the ``--audio-device`` option, instead of
setting the device directly.
.. warning::
Handling of multichannel/surround audio changed in mpv 0.8.0 from the
behavior in MPlayer/mplayer2 and older versions of mpv.
The old behavior is that the player always downmixed to stereo by
default. The ``--audio-channels`` (or ``--channels`` before that) option
had to be set to get multichannel audio. Then playing stereo would
use the ``default`` device (which typically allows multiple programs
to play audio at the same time via dmix), while playing anything with
more channels would open one of the hardware devices, e.g. via the
``surround51`` alias (typically with exclusive access). Whether the
player would use exclusive access or not would depend on the file
being played.
The new behavior since mpv 0.8.0 always enables multichannel audio,
i.e. ``--audio-channels=auto`` is the default. However, since ALSA
provides no good way to play multichannel audio in a non-exclusive
way (without blocking other applications from using audio), the player
is restricted to the capabilities of the ``default`` device by default,
which means it supports only stereo and mono (at least with current
typical ALSA configurations). But if a hardware device is selected,
then multichannel audio will typically work.
The short story is: if you want multichannel audio with ALSA, use
``--audio-device`` to select the device (use ``--audio-device=help``
to get a list of all devices and their mpv name).
You can also try `using the upmix plugin <http://git.io/vfuAy>`_.
This setup enables multichannel audio on the ``default`` device
with automatic upmixing with shared access, so playing stereo
and multichannel audio at the same time will work as expected.
``oss``
OSS audio output driver
``<dsp-device>``
Sets the audio output device (default: ``/dev/dsp``).
``<mixer-device>``
Sets the audio mixer device (default: ``/dev/mixer``).
``<mixer-channel>``
Sets the audio mixer channel (default: ``pcm``). Other valid values
include **vol, pcm, line**. For a complete list of options look for
``SOUND_DEVICE_NAMES`` in ``/usr/include/linux/soundcard.h``.
``jack``
JACK (Jack Audio Connection Kit) audio output driver
``port=<name>``
Connects to the ports with the given name (default: physical ports).
``name=<client>``
Client name that is passed to JACK (default: ``mpv``). Useful
if you want to have certain connections established automatically.
``(no-)autostart``
Automatically start jackd if necessary (default: disabled). Note that
this tends to be unreliable and will flood stdout with server messages.
``(no-)connect``
Automatically create connections to output ports (default: enabled).
When enabled, the maximum number of output channels will be limited to
the number of available output ports.
``std-channel-layout=alsa|waveext|any``
Select the standard channel layout (default: alsa). JACK itself has no
notion of channel layouts (i.e. assigning which speaker a given
channel is supposed to map to) - it just takes whatever the application
outputs, and reroutes it to whatever the user defines. This means the
user and the application are in charge of dealing with the channel
layout. ``alsa`` uses the old MPlayer layout, which is inspired by
ALSA's standard layouts. In this mode, ao_jack will refuse to play 3
or 7 channels (because these do not really have a defined meaning in
MPlayer). ``waveext`` uses WAVE_FORMAT_EXTENSIBLE order, which, even
though it was defined by Microsoft, is the standard on many systems.
The value ``any`` makes JACK accept whatever comes from the audio
filter chain, regardless of channel layout and without reordering. This
mode is probably not very useful, other than for debugging or when used
with fixed setups.
``coreaudio`` (Mac OS X only)
Native Mac OS X audio output driver using AudioUnits and the CoreAudio
sound server.
Automatically redirects to ``coreaudio_exclusive`` when playing compressed
formats.
``change-physical-format=<yes|no>``
Change the physical format to one similar to the requested audio format
(default: no). This has the advantage that multichannel audio output
will actually work. The disadvantage is that it will change the
system-wide audio settings. This is equivalent to changing the ``Format``
setting in the ``Audio Devices`` dialog in the ``Audio MIDI Setup``
utility. Note that this does not effect the selected speaker setup.
``coreaudio_exclusive`` (Mac OS X only)
Native Mac OS X audio output driver using direct device access and
exclusive mode (bypasses the sound server).
Supports only compressed formats (AC3 and DTS).
``openal``
Experimental OpenAL audio output driver
.. note:: This driver is not very useful. Playing multi-channel audio with
it is slow.
``pulse``
PulseAudio audio output driver
``[<host>][:<output sink>]``
Specify the host and optionally output sink to use. An empty <host>
string uses a local connection, "localhost" uses network transfer
(most likely not what you want).
``buffer=<1-2000|native>``
Set the audio buffer size in milliseconds. A higher value buffers
more data, and has a lower probability of buffer underruns. A smaller
value makes the audio stream react faster, e.g. to playback speed
changes. Default: 250.
``latency-hacks=<yes|no>``
Enable hacks to workaround PulseAudio timing bugs (default: no). If
enabled, mpv will do elaborate latency calculations on its own. If
disabled, it will use PulseAudio automatically updated timing
information. Disabling this might help with e.g. networked audio or
some plugins, while enabling it might help in some unknown situations
(it used to be required to get good behavior on old PulseAudio versions).
If you have stuttering video when using pulse, try to enable this
option. (Or alternatively, try to update PulseAudio.)
``dsound`` (Windows only)
DirectX DirectSound audio output driver
.. note:: This driver is for compatibility with old systems.
``device=<devicenum>``
Sets the device number to use. Playing a file with ``-v`` will show a
list of available devices.
``buffersize=<ms>``
DirectSound buffer size in milliseconds (default: 200).
``sdl``
SDL 1.2+ audio output driver. Should work on any platform supported by SDL
1.2, but may require the ``SDL_AUDIODRIVER`` environment variable to be set
appropriately for your system.
.. note:: This driver is for compatibility with extremely foreign
environments, such as systems where none of the other drivers
are available.
``buflen=<length>``
Sets the audio buffer length in seconds. Is used only as a hint by the
sound system. Playing a file with ``-v`` will show the requested and
obtained exact buffer size. A value of 0 selects the sound system
default.
``bufcnt=<count>``
Sets the number of extra audio buffers in mpv. Usually needs not be
changed.
``null``
Produces no audio output but maintains video playback speed. Use
``--ao=null:untimed`` for benchmarking.
``untimed``
Do not simulate timing of a perfect audio device. This means audio
decoding will go as fast as possible, instead of timing it to the
system clock.
``buffer``
Simulated buffer length in seconds.
``outburst``
Simulated chunk size in samples.
``speed``
Simulated audio playback speed as a multiplier. Usually, a real audio
device will not go exactly as fast as the system clock. It will deviate
just a little, and this option helps simulating this.
``latency``
Simulated device latency. This is additional to EOF.
``broken-eof``
Simulate broken audio drivers, which always add the fixed device
latency to the reported audio playback position.
``broken-delay``
Simulate broken audio drivers, which don't report latency correctly.
``channel-layouts``
If not empty, this is a ``,`` separated list of channel layouts the
AO allows. This can be used to test channel layout selection.
``pcm``
Raw PCM/WAVE file writer audio output
``(no-)waveheader``
Include or do not include the WAVE header (default: included). When
not included, raw PCM will be generated.
``file=<filename>``
Write the sound to ``<filename>`` instead of the default
``audiodump.wav``. If ``no-waveheader`` is specified, the default is
``audiodump.pcm``.
``(no-)append``
Append to the file, instead of overwriting it. Always use this with the
``no-waveheader`` option - with ``waveheader`` it's broken, because
it will write a WAVE header every time the file is opened.
``rsound``
Audio output to an RSound daemon
.. note:: Completely useless, unless you intend to run RSound. Not to be
confused with RoarAudio, which is something completely
different.
``host=<name/path>``
Set the address of the server (default: localhost). Can be either a
network hostname for TCP connections or a Unix domain socket path
starting with '/'.
``port=<number>``
Set the TCP port used for connecting to the server (default: 12345).
Not used if connecting to a Unix domain socket.
``sndio``
Audio output to the OpenBSD sndio sound system
.. note:: Experimental. There are known bugs and issues.
(Note: only supports mono, stereo, 4.0, 5.1 and 7.1 channel
layouts.)
``device=<device>``
sndio device to use (default: ``$AUDIODEVICE``, resp. ``snd0``).
``wasapi``
Audio output to the Windows Audio Session API.
``exclusive``
Requests exclusive, direct hardware access. By definition prevents
sound playback of any other program until mpv exits.
``device=<id>``
Uses the requested endpoint instead of the system's default audio
endpoint. Both an ordinal number (0,1,2,...) and the GUID
String are valid; the GUID string is guaranteed to not change
unless the driver is uninstalled.
Also supports searching active devices by human readable name. If more
than one device matches the name, refuses loading it.
This option is mostly deprecated in favour of the more general
``--audio-device`` option. That said, ``--audio-device=help`` will give
a list of valid device GUIDs (prefixed with ``wasapi/``), as well as
their human readable names, which should work here.