mirror of
https://github.com/mpv-player/mpv
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dc2d0539c7
This used to be required to workaround PulseAudio bugs. Even later, when the bugs were (partially?) fixed in PulseAudio, I had the feeling the hacks gave better behavior. On the other hand, I couldn't actually reproduce any bad behavior without the hacks lately. On top of this, it seems our hacks sometimes perform much worse than PulseAudio's native implementation (see #1430). So disable the hacks by default, but still leave the code and the option in case it still helps somewhere. Also, being able to blame PulseAudio's code by using its native API is much easier than trying to debug our own (mplayer2-derived) hacks.
306 lines
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306 lines
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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: 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 alternatively, try to update PulseAudio.)
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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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