mirror of https://github.com/mpv-player/mpv
323 lines
14 KiB
ReStructuredText
323 lines
14 KiB
ReStructuredText
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.
|
|
|
|
``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.
|
|
|
|
``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.
|