mirror of https://github.com/mpv-player/mpv
314 lines
13 KiB
ReStructuredText
314 lines
13 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[:suboption1[=value]:...],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. Suboptions are optional and can mostly be omitted.
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You can also set defaults for each driver. The defaults are applied before the
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normal driver parameters.
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``--ao-defaults=<driver1[:parameter1:parameter2:...],driver2,...>``
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Set defaults for each driver.
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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 Windows, ``--ao=wasapi`` is preferred,
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though it might cause trouble sometimes, in which case ``--ao=dsound``
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should be used. On BSD systems, ``--ao=oss`` or `--ao=sndio`` may work
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(the latter being experimental). On OS X systems, use ``--ao=coreaudio``.
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.. admonition:: Examples
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- ``--ao=alsa,oss,`` Try the ALSA driver, then the OSS driver, then others.
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- ``--ao=alsa:resample=yes:device=[plughw:0,3]`` Lets ALSA resample and
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sets the device-name as first card, fourth device.
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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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``device=<device>``
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Sets the device name. For ac3 output via S/PDIF, use an "iec958" or
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"spdif" device, unless you really know how to set it correctly.
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``resample=yes``
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Enable ALSA resampling plugin. (This is disabled by default, because
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some drivers report incorrect audio delay in some cases.)
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``mixer-device=<device>``
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Set the mixer device used with ``--no-softvol`` (default: ``default``).
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``mixer-name=<name>``
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Set the name of the mixer element (default: ``Master``). This is for
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example ``PCM`` or ``Master``.
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``mixer-index=<number>``
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Set the index of the mixer channel (default: 0). Consider the output of
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"``amixer scontrols``", then the index is the number that follows the
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name of the element.
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``non-interleaved``
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Allow output of non-interleaved formats (if the audio decoder uses
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this format). Currently disabled by default, because some popular
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ALSA plugins are utterly broken with non-interleaved formats.
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.. note::
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MPlayer and mplayer2 required you to replace any ',' with '.' and
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any ':' with '=' in the ALSA device name. mpv does not do this anymore.
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Instead, quote the device name:
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``--ao=alsa:device=[plug:surround50]``
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Note that the ``[`` and ``]`` simply quote the device name. With some
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shells (like zsh), you have to quote the option string to prevent the
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shell from interpreting the brackets instead of passing them to mpv.
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Actually, you should use the ``--audio-device`` option, instead of
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setting the device directly.
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.. warning::
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Handling of multichannel/surround audio changed in mpv 0.8.0 from the
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behavior in MPlayer/mplayer2 and older versions of mpv.
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The old behavior is that the player always downmixed to stereo by
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default. The ``--audio-channels`` (or ``--channels`` before that) option
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had to be set to get multichannel audio. Then playing stereo would
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use the ``default`` device (which typically allows multiple programs
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to play audio at the same time via dmix), while playing anything with
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more channels would open one of the hardware devices, e.g. via the
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``surround51`` alias (typically with exclusive access). Whether the
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player would use exclusive access or not would depend on the file
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being played.
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The new behavior since mpv 0.8.0 always enables multichannel audio,
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i.e. ``--audio-channels=auto`` is the default. However, since ALSA
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provides no good way to play multichannel audio in a non-exclusive
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way (without blocking other applications from using audio), the player
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is restricted to the capabilities of the ``default`` device by default,
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which means it supports only stereo and mono (at least with current
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typical ALSA configurations). But if a hardware device is selected,
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then multichannel audio will typically work.
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The short story is: if you want multichannel audio with ALSA, use
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``--audio-device`` to select the device (use ``--audio-device=help``
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to get a list of all devices and their mpv name).
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You can also try
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`Using the upmix plugin <https://github.com/mpv-player/mpv/wiki/ALSA:-Surround-Sound-and-Upmixing>`_.
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``oss``
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OSS audio output driver
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``<dsp-device>``
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Sets the audio output device (default: ``/dev/dsp``).
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``<mixer-device>``
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Sets the audio mixer device (default: ``/dev/mixer``).
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``<mixer-channel>``
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Sets the audio mixer channel (default: ``pcm``). Other valid values
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include **vol, pcm, line**. For a complete list of options look for
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``SOUND_DEVICE_NAMES`` in ``/usr/include/linux/soundcard.h``.
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``jack``
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JACK (Jack Audio Connection Kit) audio output driver
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``port=<name>``
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Connects to the ports with the given name (default: physical ports).
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``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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``(no-)autostart``
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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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``(no-)connect``
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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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``std-channel-layout=alsa|waveext|any``
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Select the standard channel layout (default: alsa). 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. ``alsa`` uses the old MPlayer layout, which is inspired by
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ALSA's standard layouts. In this mode, ao_jack will refuse to play 3
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or 7 channels (because these do not really have a defined meaning in
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MPlayer). ``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`` (Mac OS X only)
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Native Mac OS X 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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``coreaudio_exclusive`` (Mac OS X only)
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Native Mac OS X audio output driver using direct device access and
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exclusive mode (bypasses the sound server).
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Supports only compressed formats (AC3 and DTS).
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``openal``
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Experimental OpenAL audio output driver
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.. note:: This driver is not very useful. Playing multi-channel audio with
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it is slow.
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``pulse``
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PulseAudio audio output driver
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``[<host>][:<output sink>]``
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Specify the host and optionally output sink to use. An empty <host>
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string uses a local connection, "localhost" uses network transfer
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(most likely not what you want).
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``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. Default: 250.
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``latency-hacks=<yes|no>``
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Enable hacks to workaround PulseAudio timing bugs (default: yes). 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.
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``portaudio``
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PortAudio audio output driver. This works on all platforms, and has
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extensive MS Windows support.
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.. note:: This driver is not very useful. It was added in the hope of
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providing portable audio API across all platforms, but turned
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out semi-broken and underfeatured.
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``device``
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Specify the subdevice to use. Giving ``help`` as device name lists all
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devices found by PortAudio. Devices can be given as numeric values,
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starting from ``1``.
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``dsound`` (Windows only)
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DirectX DirectSound audio output driver
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.. note:: This driver is for compatibility with old systems.
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``device=<devicenum>``
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Sets the device number to use. Playing a file with ``-v`` will show a
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list of available devices.
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``buffersize=<ms>``
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DirectSound buffer size in milliseconds (default: 200).
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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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``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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``bufcnt=<count>``
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Sets the number of extra audio buffers in mpv. Usually needs not be
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changed.
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``null``
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Produces no audio output but maintains video playback speed. Use
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``--ao=null:untimed`` for benchmarking.
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``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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``buffer``
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Simulated buffer length in seconds.
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``outburst``
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Simulated chunk size in samples.
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``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 simulating this.
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``latency``
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Simulated device latency. This is additional to EOF.
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``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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``pcm``
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Raw PCM/WAVE file writer audio output
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``(no-)waveheader``
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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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``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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``rsound``
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Audio output to an RSound daemon
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.. note:: Completely useless, unless you intend to run RSound. Not to be
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confused with RoarAudio, which is something completely
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different.
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``host=<name/path>``
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Set the address of the server (default: localhost). Can be either a
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network hostname for TCP connections or a Unix domain socket path
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starting with '/'.
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``port=<number>``
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Set the TCP port used for connecting to the server (default: 12345).
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Not used if connecting to a Unix domain socket.
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``sndio``
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Audio output to the OpenBSD sndio sound system
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.. note:: Experimental. There are known bugs and issues.
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(Note: only supports mono, stereo, 4.0, 5.1 and 7.1 channel
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layouts.)
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``device=<device>``
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sndio device to use (default: ``$AUDIODEVICE``, resp. ``snd0``).
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``wasapi``
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Audio output to the Windows Audio Session API.
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``device=<id>``
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Uses the requested endpoint instead of the system's default audio
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endpoint. Both the number and the ID String are valid; the ID String
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is guaranteed to not change unless the driver is uninstalled.
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Also supports searching active devices by name. If more than one
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device matches the name, refuses loading it.
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To get a list of the valid devices, give ``help`` as the id. The
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list is the same as the ``list`` suboption, but stops the player
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initialization.
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``exclusive``
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Requests exclusive, direct hardware access. By definition prevents
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sound playback of any other program until mpv exits.
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``list``
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Lists all audio endpoints (output devices) present in the system.
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