mirror of https://github.com/mpv-player/mpv
224 lines
9.4 KiB
ReStructuredText
224 lines
9.4 KiB
ReStructuredText
AUDIO OUTPUT DRIVERS
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====================
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Audio output drivers are interfaces to different audio output facilities. The
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syntax is:
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``--ao=<driver1,driver2,...[,]>``
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Specify a priority list of audio output drivers to be used.
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If the list has a trailing ',', mpv will fall back on drivers not contained
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in the list.
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.. note::
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See ``--ao=help`` for a list of compiled-in audio output drivers. The
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driver ``--ao=alsa`` is preferred. ``--ao=pulse`` is preferred on systems
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where PulseAudio is used. On BSD systems, ``--ao=oss`` is preferred.
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Available audio output drivers are:
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``alsa`` (Linux only)
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ALSA audio output driver
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See `ALSA audio output options`_ for options specific to this AO.
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.. warning::
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To get multichannel/surround audio, use ``--audio-channels=auto``. The
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default for this option is ``auto-safe``, which makes this audio output
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explicitly reject multichannel output, as there is no way to detect
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whether a certain channel layout is actually supported.
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You can also try `using the upmix plugin <http://git.io/vfuAy>`_.
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This setup enables multichannel audio on the ``default`` device
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with automatic upmixing with shared access, so playing stereo
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and multichannel audio at the same time will work as expected.
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``oss``
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OSS audio output driver
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``jack``
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JACK (Jack Audio Connection Kit) audio output driver.
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The following global options are supported by this audio output:
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``--jack-port=<name>``
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Connects to the ports with the given name (default: physical ports).
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``--jack-name=<client>``
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Client name that is passed to JACK (default: ``mpv``). Useful
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if you want to have certain connections established automatically.
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``--jack-autostart=<yes|no>``
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Automatically start jackd if necessary (default: disabled). Note that
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this tends to be unreliable and will flood stdout with server messages.
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``--jack-connect=<yes|no>``
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Automatically create connections to output ports (default: enabled).
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When enabled, the maximum number of output channels will be limited to
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the number of available output ports.
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``--jack-std-channel-layout=<waveext|any>``
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Select the standard channel layout (default: waveext). JACK itself has no
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notion of channel layouts (i.e. assigning which speaker a given
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channel is supposed to map to) - it just takes whatever the application
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outputs, and reroutes it to whatever the user defines. This means the
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user and the application are in charge of dealing with the channel
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layout. ``waveext`` uses WAVE_FORMAT_EXTENSIBLE order, which, even
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though it was defined by Microsoft, is the standard on many systems.
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The value ``any`` makes JACK accept whatever comes from the audio
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filter chain, regardless of channel layout and without reordering. This
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mode is probably not very useful, other than for debugging or when used
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with fixed setups.
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``coreaudio`` (macOS only)
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Native macOS audio output driver using AudioUnits and the CoreAudio
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sound server.
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Automatically redirects to ``coreaudio_exclusive`` when playing compressed
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formats.
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The following global options are supported by this audio output:
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``--coreaudio-change-physical-format=<yes|no>``
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Change the physical format to one similar to the requested audio format
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(default: no). This has the advantage that multichannel audio output
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will actually work. The disadvantage is that it will change the
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system-wide audio settings. This is equivalent to changing the ``Format``
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setting in the ``Audio Devices`` dialog in the ``Audio MIDI Setup``
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utility. Note that this does not affect the selected speaker setup.
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``--coreaudio-spdif-hack=<yes|no>``
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Try to pass through AC3/DTS data as PCM. This is useful for drivers
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which do not report AC3 support. It converts the AC3 data to float,
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and assumes the driver will do the inverse conversion, which means
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a typical A/V receiver will pick it up as compressed IEC framed AC3
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stream, ignoring that it's marked as PCM. This disables normal AC3
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passthrough (even if the device reports it as supported). Use with
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extreme care.
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``coreaudio_exclusive`` (macOS only)
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Native macOS audio output driver using direct device access and
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exclusive mode (bypasses the sound server).
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``openal``
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OpenAL audio output driver. This is broken and does not work.
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``--openal-num-buffers=<2-128>``
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Specify the number of audio buffers to use. Lower values are better for
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lower CPU usage. Default: 4.
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``--openal-num-samples=<256-32768>``
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Specify the number of complete samples to use for each buffer. Higher
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values are better for lower CPU usage. Default: 8192.
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``--openal-direct-channels=<yes|no>``
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Enable OpenAL Soft's direct channel extension when available to avoid
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tinting the sound with ambisonics or HRTF.
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Channels are dropped when when they are not available as downmixing
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will be disabled. Default: no.
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``pulse``
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PulseAudio audio output driver
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The following global options are supported by this audio output:
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``--pulse-host=<host>``
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Specify the host to use. An empty <host> string uses a local connection,
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"localhost" uses network transfer (most likely not what you want).
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``--pulse-buffer=<1-2000|native>``
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Set the audio buffer size in milliseconds. A higher value buffers
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more data, and has a lower probability of buffer underruns. A smaller
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value makes the audio stream react faster, e.g. to playback speed
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changes.
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``--pulse-latency-hacks=<yes|no>``
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Enable hacks to workaround PulseAudio timing bugs (default: no). If
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enabled, mpv will do elaborate latency calculations on its own. If
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disabled, it will use PulseAudio automatically updated timing
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information. Disabling this might help with e.g. networked audio or
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some plugins, while enabling it might help in some unknown situations
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(it used to be required to get good behavior on old PulseAudio versions).
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If you have stuttering video when using pulse, try to enable this
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option. (Or try to update PulseAudio.)
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``--pulse-allow-suspended=<yes|no>``
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Allow mpv to use PulseAudio even if the sink is suspended (default: no).
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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.
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``sdl``
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SDL 1.2+ audio output driver. Should work on any platform supported by SDL
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1.2, but may require the ``SDL_AUDIODRIVER`` environment variable to be set
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appropriately for your system.
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.. note:: This driver is for compatibility with extremely foreign
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environments, such as systems where none of the other drivers
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are available.
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The following global options are supported by this audio output:
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``--sdl-buflen=<length>``
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Sets the audio buffer length in seconds. Is used only as a hint by the
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sound system. Playing a file with ``-v`` will show the requested and
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obtained exact buffer size. A value of 0 selects the sound system
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default.
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``null``
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Produces no audio output but maintains video playback speed. You can use
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``--ao=null --ao-null-untimed`` for benchmarking.
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The following global options are supported by this audio output:
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``--ao-null-untimed``
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Do not simulate timing of a perfect audio device. This means audio
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decoding will go as fast as possible, instead of timing it to the
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system clock.
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``--ao-null-buffer``
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Simulated buffer length in seconds.
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``--ao-null-outburst``
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Simulated chunk size in samples.
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``--ao-null-speed``
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Simulated audio playback speed as a multiplier. Usually, a real audio
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device will not go exactly as fast as the system clock. It will deviate
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just a little, and this option helps to simulate this.
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``--ao-null-latency``
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Simulated device latency. This is additional to EOF.
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``--ao-null-broken-eof``
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Simulate broken audio drivers, which always add the fixed device
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latency to the reported audio playback position.
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``--ao-null-broken-delay``
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Simulate broken audio drivers, which don't report latency correctly.
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``--ao-null-channel-layouts``
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If not empty, this is a ``,`` separated list of channel layouts the
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AO allows. This can be used to test channel layout selection.
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``--ao-null-format``
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Force the audio output format the AO will accept. If unset accepts any.
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``pcm``
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Raw PCM/WAVE file writer audio output
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The following global options are supported by this audio output:
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``--ao-pcm-waveheader=<yes|no>``
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Include or do not include the WAVE header (default: included). When
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not included, raw PCM will be generated.
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``--ao-pcm-file=<filename>``
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Write the sound to ``<filename>`` instead of the default
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``audiodump.wav``. If ``no-waveheader`` is specified, the default is
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``audiodump.pcm``.
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``--ao-pcm-append=<yes|no>``
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Append to the file, instead of overwriting it. Always use this with the
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``no-waveheader`` option - with ``waveheader`` it's broken, because
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it will write a WAVE header every time the file is opened.
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``wasapi``
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Audio output to the Windows Audio Session API.
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