mirror of https://github.com/mpv-player/mpv
239 lines
9.8 KiB
ReStructuredText
239 lines
9.8 KiB
ReStructuredText
AUDIO OUTPUT DRIVERS
|
|
====================
|
|
|
|
Audio output drivers are interfaces to different audio output facilities. The
|
|
syntax is:
|
|
|
|
``--ao=<driver1,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.
|
|
|
|
.. 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 BSD systems, ``--ao=oss`` is preferred.
|
|
|
|
Available audio output drivers are:
|
|
|
|
``alsa`` (Linux only)
|
|
ALSA audio output driver
|
|
|
|
See `ALSA audio output options`_ for options specific to this AO.
|
|
|
|
.. warning::
|
|
|
|
To get multichannel/surround audio, use ``--audio-channels=auto``. The
|
|
default for this option is ``auto-safe``, which makes this audio output
|
|
explicitly reject multichannel output, as there is no way to detect
|
|
whether a certain channel layout is actually supported.
|
|
|
|
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
|
|
|
|
``jack``
|
|
JACK (Jack Audio Connection Kit) audio output driver.
|
|
|
|
The following global options are supported by this audio output:
|
|
|
|
``--jack-port=<name>``
|
|
Connects to the ports with the given name (default: physical ports).
|
|
``--jack-name=<client>``
|
|
Client name that is passed to JACK (default: ``mpv``). Useful
|
|
if you want to have certain connections established automatically.
|
|
``--jack-autostart=<yes|no>``
|
|
Automatically start jackd if necessary (default: disabled). Note that
|
|
this tends to be unreliable and will flood stdout with server messages.
|
|
``--jack-connect=<yes|no>``
|
|
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.
|
|
``--jack-std-channel-layout=<waveext|any>``
|
|
Select the standard channel layout (default: waveext). 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. ``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`` (macOS only)
|
|
Native macOS audio output driver using AudioUnits and the CoreAudio
|
|
sound server.
|
|
|
|
Automatically redirects to ``coreaudio_exclusive`` when playing compressed
|
|
formats.
|
|
|
|
The following global options are supported by this audio output:
|
|
|
|
``--coreaudio-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 affect the selected speaker setup.
|
|
|
|
``--coreaudio-spdif-hack=<yes|no>``
|
|
Try to pass through AC3/DTS data as PCM. This is useful for drivers
|
|
which do not report AC3 support. It converts the AC3 data to float,
|
|
and assumes the driver will do the inverse conversion, which means
|
|
a typical A/V receiver will pick it up as compressed IEC framed AC3
|
|
stream, ignoring that it's marked as PCM. This disables normal AC3
|
|
passthrough (even if the device reports it as supported). Use with
|
|
extreme care.
|
|
|
|
|
|
``coreaudio_exclusive`` (macOS only)
|
|
Native macOS audio output driver using direct device access and
|
|
exclusive mode (bypasses the sound server).
|
|
|
|
``openal``
|
|
OpenAL audio output driver.
|
|
|
|
``--openal-num-buffers=<2-128>``
|
|
Specify the number of audio buffers to use. Lower values are better for
|
|
lower CPU usage. Default: 4.
|
|
|
|
``--openal-num-samples=<256-32768>``
|
|
Specify the number of complete samples to use for each buffer. Higher
|
|
values are better for lower CPU usage. Default: 8192.
|
|
|
|
``--openal-direct-channels=<yes|no>``
|
|
Enable OpenAL Soft's direct channel extension when available to avoid
|
|
tinting the sound with ambisonics or HRTF. Default: yes.
|
|
|
|
``pulse``
|
|
PulseAudio audio output driver
|
|
|
|
The following global options are supported by this audio output:
|
|
|
|
``--pulse-host=<host>``
|
|
Specify the host to use. An empty <host> string uses a local connection,
|
|
"localhost" uses network transfer (most likely not what you want).
|
|
|
|
``--pulse-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.
|
|
|
|
``--pulse-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 try to update PulseAudio.)
|
|
|
|
``--pulse-allow-suspended=<yes|no>``
|
|
Allow mpv to use PulseAudio even if the sink is suspended (default: no).
|
|
Can be useful if PulseAudio is running as a bridge to jack and mpv has its sink-input set to the one jack is using.
|
|
|
|
``pipewire``
|
|
PipeWire audio output driver
|
|
|
|
The following global options are supported by this audio output:
|
|
|
|
``--pipewire-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.
|
|
|
|
``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.
|
|
|
|
The following global options are supported by this audio output:
|
|
|
|
``--sdl-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.
|
|
|
|
``null``
|
|
Produces no audio output but maintains video playback speed. You can use
|
|
``--ao=null --ao-null-untimed`` for benchmarking.
|
|
|
|
The following global options are supported by this audio output:
|
|
|
|
``--ao-null-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.
|
|
|
|
``--ao-null-buffer``
|
|
Simulated buffer length in seconds.
|
|
|
|
``--ao-null-outburst``
|
|
Simulated chunk size in samples.
|
|
|
|
``--ao-null-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 to simulate this.
|
|
|
|
``--ao-null-latency``
|
|
Simulated device latency. This is additional to EOF.
|
|
|
|
``--ao-null-broken-eof``
|
|
Simulate broken audio drivers, which always add the fixed device
|
|
latency to the reported audio playback position.
|
|
|
|
``--ao-null-broken-delay``
|
|
Simulate broken audio drivers, which don't report latency correctly.
|
|
|
|
``--ao-null-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.
|
|
|
|
``--ao-null-format``
|
|
Force the audio output format the AO will accept. If unset accepts any.
|
|
|
|
``pcm``
|
|
Raw PCM/WAVE file writer audio output
|
|
|
|
The following global options are supported by this audio output:
|
|
|
|
``--ao-pcm-waveheader=<yes|no>``
|
|
Include or do not include the WAVE header (default: included). When
|
|
not included, raw PCM will be generated.
|
|
``--ao-pcm-file=<filename>``
|
|
Write the sound to ``<filename>`` instead of the default
|
|
``audiodump.wav``. If ``no-waveheader`` is specified, the default is
|
|
``audiodump.pcm``.
|
|
``--ao-pcm-append=<yes|no>``
|
|
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.
|
|
|
|
``sndio``
|
|
Audio output to the OpenBSD sndio sound system
|
|
|
|
(Note: only supports mono, stereo, 4.0, 5.1 and 7.1 channel
|
|
layouts.)
|
|
|
|
``wasapi``
|
|
Audio output to the Windows Audio Session API.
|