DOCS: remove en/ sub-directory

This additional sub-directory doesn't serve any purpose anymore. Get rid
of it.

11 files changed, 0 insertions, 8966 deletions

diff --git a/DOCS/man/en/af.rst b/DOCS/man/en/af.rst
deleted file mode 100644
index 0512376ff2..0000000000
--- a/DOCS/man/en/af.rst
+++ /dev/null
@@ -1,616 +0,0 @@
-AUDIO FILTERS
-=============
-
-Audio filters allow you to modify the audio stream and its properties. The
-syntax is:
-
-``--af=<filter1[=parameter1:parameter2:...],filter2,...>``
-    Setup a chain of audio filters.
-
-.. note::
-
-    To get a full list of available audio filters, see ``--af=help``.
-
-You can also set defaults for each filter. The defaults are applied before the
-normal filter parameters.
-
-``--af-defaults=<filter1[=parameter1:parameter2:...],filter2,...>``
-    Set defaults for each filter.
-
-Audio filters are managed in lists. There are a few commands to manage the
-filter list:
-
-``--af-add=<filter1[,filter2,...]>``
-    Appends the filters given as arguments to the filter list.
-
-``--af-pre=<filter1[,filter2,...]>``
-    Prepends the filters given as arguments to the filter list.
-
-``--af-del=<index1[,index2,...]>``
-    Deletes the filters at the given indexes. Index numbers start at 0,
-    negative numbers address the end of the list (-1 is the last).
-
-``--af-clr``
-    Completely empties the filter list.
-
-Available filters are:
-
-``lavrresample[=option1:option2:...]``
-    This filter uses libavresample (or libswresample, depending on the build)
-    to change sample rate, sample format, or channel layout of the audio stream.
-    This filter is automatically enabled if the audio output does not support
-    the audio configuration of the file being played.
-
-    It supports only the following sample formats: u8, s16, s32, float.
-
-    ``filter-size=<length>``
-        Length of the filter with respect to the lower sampling rate. (default:
-        16)
-    ``phase-shift=<count>``
-        Log2 of the number of polyphase entries. (..., 10->1024, 11->2048,
-        12->4096, ...) (default: 10->1024)
-    ``cutoff=<cutoff>``
-        Cutoff frequency (0.0-1.0), default set depending upon filter length.
-    ``linear``
-        If set then filters will be linearly interpolated between polyphase
-        entries. (default: no)
-    ``no-detach``
-        Do not detach if input and output audio format/rate/channels match.
-        (If you just want to set defaults for this filter that will be used
-        even by automatically inserted lavrresample instances, you should
-        prefer setting them with ``--af-defaults=lavrresample:...``.)
-    ``o=<string>``
-        Set AVOptions on the SwrContext or AVAudioResampleContext. These should
-        be documented by FFmpeg or Libav.
-
-``lavcac3enc[=tospdif[:bitrate[:minchn]]]``
-    Encode multi-channel audio to AC-3 at runtime using libavcodec. Supports
-    16-bit native-endian input format, maximum 6 channels. The output is
-    big-endian when outputting a raw AC-3 stream, native-endian when
-    outputting to S/PDIF. If the input sample rate is not 48 kHz, 44.1 kHz or
-    32 kHz, it will be resampled to 48 kHz.
-
-    ``tospdif=<yes|no>``
-        Output raw AC-3 stream if ``no``, output to S/PDIF for
-        passthrough if ``yes`` (default).
-
-    ``bitrate=<rate>``
-        The bitrate use for the AC-3 stream. Set it to 384 to get 384 kbps.
-
-        Valid values: 32, 40, 48, 56, 64, 80, 96, 112, 128,
-        160, 192, 224, 256, 320, 384, 448, 512, 576, 640.
-
-        The special value ``default`` selects a default bitrate based on the
-        input channel number:
-
-        :1ch: 96
-        :2ch: 192
-        :3ch: 224
-        :4ch: 384
-        :5ch: 448
-        :6ch: 448
-
-    ``minchn=<n>``
-        If the input channel number is less than ``<minchn>``, the filter will
-        detach itself (default: 5).
-
-``sweep[=speed]``
-    Produces a sine sweep.
-
-    ``<0.0-1.0>``
-        Sine function delta, use very low values to hear the sweep.
-
-``sinesuppress[=freq:decay]``
-    Remove a sine at the specified frequency. Useful to get rid of the 50/60Hz
-    noise on low quality audio equipment. It only works on mono input.
-
-    ``<freq>``
-        The frequency of the sine which should be removed (in Hz) (default:
-        50)
-    ``<decay>``
-        Controls the adaptivity (a larger value will make the filter adapt to
-        amplitude and phase changes quicker, a smaller value will make the
-        adaptation slower) (default: 0.0001). Reasonable values are around
-        0.001.
-
-``bs2b[=option1:option2:...]``
-    Bauer stereophonic to binaural transformation using libbs2b. Improves the
-    headphone listening experience by making the sound similar to that from
-    loudspeakers, allowing each ear to hear both channels and taking into
-    account the distance difference and the head shadowing effect. It is
-    applicable only to 2-channel audio.
-
-    ``fcut=<300-1000>``
-        Set cut frequency in Hz.
-    ``feed=<10-150>``
-        Set feed level for low frequencies in 0.1*dB.
-    ``profile=<value>``
-        Several profiles are available for convenience:
-
-        :default: will be used if nothing else was specified (fcut=700,
-                  feed=45)
-        :cmoy:    Chu Moy circuit implementation (fcut=700, feed=60)
-        :jmeier:  Jan Meier circuit implementation (fcut=650, feed=95)
-
-    If ``fcut`` or ``feed`` options are specified together with a profile, they
-    will be applied on top of the selected profile.
-
-``hrtf[=flag]``
-    Head-related transfer function: Converts multichannel audio to 2-channel
-    output for headphones, preserving the spatiality of the sound.
-
-    ==== ===================================
-    Flag Meaning
-    ==== ===================================
-    m    matrix decoding of the rear channel
-    s    2-channel matrix decoding
-    0    no matrix decoding (default)
-    ==== ===================================
-
-``equalizer=g1:g2:g3:...:g10``
-    10 octave band graphic equalizer, implemented using 10 IIR band-pass
-    filters. This means that it works regardless of what type of audio is
-    being played back. The center frequencies for the 10 bands are:
-
-    === ==========
-    No. frequency
-    === ==========
-    0    31.25  Hz
-    1    62.50  Hz
-    2   125.00  Hz
-    3   250.00  Hz
-    4   500.00  Hz
-    5     1.00 kHz
-    6     2.00 kHz
-    7     4.00 kHz
-    8     8.00 kHz
-    9    16.00 kHz
-    === ==========
-
-    If the sample rate of the sound being played is lower than the center
-    frequency for a frequency band, then that band will be disabled. A known
-    bug with this filter is that the characteristics for the uppermost band
-    are not completely symmetric if the sample rate is close to the center
-    frequency of that band. This problem can be worked around by upsampling
-    the sound using a resampling filter before it reaches this filter.
-
-    ``<g1>:<g2>:<g3>:...:<g10>``
-        floating point numbers representing the gain in dB for each frequency
-        band (-12-12)
-
-    .. admonition:: Example
-
-        ``mpv --af=equalizer=11:11:10:5:0:-12:0:5:12:12 media.avi``
-            Would amplify the sound in the upper and lower frequency region
-            while canceling it almost completely around 1kHz.
-
-``channels=nch[:routes]``
-    Can be used for adding, removing, routing and copying audio channels. If
-    only ``<nch>`` is given, the default routing is used. It works as follows:
-    If the number of output channels is greater than the number of input
-    channels, empty channels are inserted (except when mixing from mono to
-    stereo; then the mono channel is duplicated). If the number of output
-    channels is less than the number of input channels, the exceeding
-    channels are truncated.
-
-    ``<nch>``
-        number of output channels (1-8)
-    ``<routes>``
-        List of ``,`` separated routes, in the form ``from1-to1,from2-to2,...``.
-        Each pair defines where to route each channel. There can be at most
-        8 routes. Without this argument, the default routing is used. Since
-        ``,`` is also used to separate filters, you must quote this argument
-        with ``[...]`` or similar.
-
-    .. admonition:: Examples
-
-        ``mpv --af=channels=4:[0-1,1-0,0-2,1-3] media.avi``
-            Would change the number of channels to 4 and set up 4 routes that
-            swap channel 0 and channel 1 and leave channel 2 and 3 intact.
-            Observe that if media containing two channels were played back,
-            channels 2 and 3 would contain silence but 0 and 1 would still be
-            swapped.
-
-        ``mpv --af=channels=6:[0-0,0-1,0-2,0-3] media.avi``
-            Would change the number of channels to 6 and set up 4 routes that
-            copy channel 0 to channels 0 to 3. Channel 4 and 5 will contain
-            silence.
-
-    .. note::
-
-        You should probably not use this filter. If you want to change the
-        output channel layout, try the ``format`` filter, which can make mpv
-        automatically up- and downmix standard channel layouts.
-
-``format=format:srate:channels:out-format:out-srate:out-channels``
-    Force a specific audio format/configuration without actually changing the
-    audio data. Keep in mind that the filter system might auto-insert actual
-    conversion filters before or after this filter if needed.
-
-    All parameters are optional. The first 3 parameters restrict what the filter
-    accepts as input. The ``out-`` parameters change the audio format, without
-    actually doing a conversion. The data will be 'reinterpreted' by the
-    filters or audio outputs following this filter.
-
-    ``<format>``
-        Force conversion to this format. Use ``--af=format=format=help`` to get
-        a list of valid formats.
-
-    ``<srate>``
-        Force conversion to a specific sample rate. The rate is an integer,
-        48000 for example.
-
-    ``<channels>``
-        Force mixing to a specific channel layout. See ``--audio-channels`` option
-        for possible values.
-
-    ``<out-format>``
-
-    ``<out-srate>``
-
-    ``<out-channels>``
-
-    See also ``--audio-format``, ``--audio-samplerate``, and
-    ``--audio-channels`` for related options. Keep in mind that
-    ``--audio-channels`` does not actually force the number of
-    channels in most cases, while this filter can do this.
-
-    *NOTE*: this filter used to be named ``force``. Also, unlike the old
-    ``format`` filter, this does not do any actual conversion anymore.
-    Conversion is done by other, automatically inserted filters.
-
-``convert24``
-    Filter for internal use only. Converts between 24-bit and 32-bit sample
-    formats.
-
-``convertsignendian``
-    Filter for internal use only. Converts between signed/unsigned formats
-    and formats with different endian.
-
-``volume[=<volumedb>[:...]]``
-    Implements software volume control. Use this filter with caution since it
-    can reduce the signal to noise ratio of the sound. In most cases it is
-    best to use the *Master* volume control of your sound card or the volume
-    knob on your amplifier.
-
-    *NOTE*: This filter is not reentrant and can therefore only be enabled
-    once for every audio stream.
-
-    ``<volumedb>``
-        Sets the desired gain in dB for all channels in the stream from -200dB
-        to +60dB, where -200dB mutes the sound completely and +60dB equals a
-        gain of 1000 (default: 0).
-    ``replaygain-track``
-        Adjust volume gain according to the track-gain replaygain value stored
-        in the file metadata.
-    ``replaygain-album``
-        Like replaygain-track, but using the album-gain value instead.
-    ``replaygain-preamp``
-        Pre-amplification gain in dB to apply to the selected replaygain gain
-        (default: 0).
-    ``replaygain-clip=yes|no``
-        Prevent clipping caused by replaygain by automatically lowering the
-        gain (default). Use ``replaygain-clip=no`` to disable this.
-    ``softclip``
-        Turns soft clipping on. Soft-clipping can make the
-        sound more smooth if very high volume levels are used. Enable this
-        option if the dynamic range of the loudspeakers is very low.
-
-        *WARNING*: This feature creates distortion and should be considered a
-        last resort.
-    ``s16``
-        Force S16 sample format if set. Lower quality, but might be faster
-        in some situations.
-    ``detach``
-        Remove the filter if the volume is not changed at audio filter config
-        time. Useful with replaygain: if the current file has no replaygain
-        tags, then the filter will be removed if this option is enabled.
-        (If ``--softvol=yes`` is used and the player volume controls are used
-        during playback, a different volume filter will be inserted.)
-
-    .. admonition:: Example
-
-        ``mpv --af=volume=10.1 media.avi``
-            Would amplify the sound by 10.1dB and hard-clip if the sound level
-            is too high.
-
-``pan=n:[<matrix>]``
-    Mixes channels arbitrarily. Basically a combination of the volume and the
-    channels filter that can be used to down-mix many channels to only a few,
-    e.g. stereo to mono, or vary the "width" of the center speaker in a
-    surround sound system. This filter is hard to use, and will require some
-    tinkering before the desired result is obtained. The number of options for
-    this filter depends on the number of output channels. An example how to
-    downmix a six-channel file to two channels with this filter can be found
-    in the examples section near the end.
-
-    ``<n>``
-        Number of output channels (1-8).
-    ``<matrix>``
-        A list of values ``[L00,L01,L02,...,L10,L11,L12,...,Ln0,Ln1,Ln2,...]``,
-        where each element ``Lij`` means how much of input channel i is mixed
-        into output channel j (range 0-1). So in principle you first have n
-        numbers saying what to do with the first input channel, then n numbers
-        that act on the second input channel etc. If you do not specify any
-        numbers for some input channels, 0 is assumed.
-        Note that the values are separated by ``,``, which is already used
-        by the option parser to separate filters. This is why you must quote
-        the value list with ``[...]`` or similar.
-
-    .. admonition:: Examples
-
-        ``mpv --af=pan=1:[0.5,0.5] media.avi``
-            Would downmix from stereo to mono.
-
-        ``mpv --af=pan=3:[1,0,0.5,0,1,0.5] media.avi``
-            Would give 3 channel output leaving channels 0 and 1 intact, and mix
-            channels 0 and 1 into output channel 2 (which could be sent to a
-            subwoofer for example).
-
-    .. note::
-
-        If you just want to force remixing to a certain output channel layout,
-        it is easier to use the ``format`` filter. For example,
-        ``mpv '--af=format=channels=5.1' '--audio-channels=5.1'`` would always force
-        remixing audio to 5.1 and output it like this.
-
-``sub[=fc:ch]``
-    Adds a subwoofer channel to the audio stream. The audio data used for
-    creating the subwoofer channel is an average of the sound in channel 0 and
-    channel 1. The resulting sound is then low-pass filtered by a 4th order
-    Butterworth filter with a default cutoff frequency of 60Hz and added to a
-    separate channel in the audio stream.
-
-    .. warning::
-
-        Disable this filter when you are playing media with an LFE channel
-        (e.g. 5.1 surround sound), otherwise this filter will disrupt the sound
-        to the subwoofer.
-
-    ``<fc>``
-        cutoff frequency in Hz for the low-pass filter (20Hz to 300Hz)
-        (default: 60Hz) For the best result try setting the cutoff frequency
-        as low as possible. This will improve the stereo or surround sound
-        experience.
-    ``<ch>``
-        Determines the channel number in which to insert the sub-channel
-        audio. Channel number can be between 0 and 7 (default: 5). Observe
-        that the number of channels will automatically be increased to <ch> if
-        necessary.
-
-    .. admonition:: Example
-
-        ``mpv --af=sub=100:4 --audio-channels=5 media.avi``
-            Would add a subwoofer channel with a cutoff frequency of 100Hz to
-            output channel 4.
-
-``center``
-    Creates a center channel from the front channels. May currently be low
-    quality as it does not implement a high-pass filter for proper extraction
-    yet, but averages and halves the channels instead.
-
-    ``<ch>``
-        Determines the channel number in which to insert the center channel.
-        Channel number can be between 0 and 7 (default: 5). Observe that the
-        number of channels will automatically be increased to ``<ch>`` if
-        necessary.
-
-``surround[=delay]``
-    Decoder for matrix encoded surround sound like Dolby Surround. Some files
-    with 2-channel audio actually contain matrix encoded surround sound.
-
-    ``<delay>``
-        delay time in ms for the rear speakers (0 to 1000) (default: 20) This
-        delay should be set as follows: If d1 is the distance from the
-        listening position to the front speakers and d2 is the distance from
-        the listening position to the rear speakers, then the delay should be
-        set to 15ms if d1 <= d2 and to 15 + 5*(d1-d2) if d1 > d2.
-
-    .. admonition:: Example
-
-        ``mpv --af=surround=15 --audio-channels=4 media.avi``
-            Would add surround sound decoding with 15ms delay for the sound to
-            the rear speakers.
-
-``delay[=[ch1,ch2,...]]``
-    Delays the sound to the loudspeakers such that the sound from the
-    different channels arrives at the listening position simultaneously. It is
-    only useful if you have more than 2 loudspeakers.
-
-    ``[ch1,ch2,...]``
-        The delay in ms that should be imposed on each channel (floating point
-        number between 0 and 1000).
-
-    To calculate the required delay for the different channels, do as follows:
-
-    1. Measure the distance to the loudspeakers in meters in relation to your
-       listening position, giving you the distances s1 to s5 (for a 5.1
-       system). There is no point in compensating for the subwoofer (you will
-       not hear the difference anyway).
-
-    2. Subtract the distances s1 to s5 from the maximum distance, i.e.
-       ``s[i] = max(s) - s[i]; i = 1...5``.
-
-    3. Calculate the required delays in ms as ``d[i] = 1000*s[i]/342; i =
-       1...5``.
-
-    .. admonition:: Example
-
-        ``mpv --af=delay=[10.5,10.5,0,0,7,0] media.avi``
-            Would delay front left and right by 10.5ms, the two rear channels
-            and the subwoofer by 0ms and the center channel by 7ms.
-
-``export=mmapped_file:nsamples]``
-    Exports the incoming signal to other processes using memory mapping
-    (``mmap()``). Memory mapped areas contain a header::
-
-        int nch                      /* number of channels */
-        int size                     /* buffer size */
-        unsigned long long counter   /* Used to keep sync, updated every time
-                                        new data is exported. */
-
-    The rest is payload (non-interleaved) 16-bit data.
-
-    ``<mmapped_file>``
-        File to map data to (required)
-    ``<nsamples>``
-        number of samples per channel (default: 512).
-
-    .. admonition:: Example
-
-        ``mpv --af=export=/tmp/mpv-af_export:1024 media.avi``
-            Would export 1024 samples per channel to ``/tmp/mpv-af_export``.
-
-``extrastereo[=mul]``
-    (Linearly) increases the difference between left and right channels which
-    adds some sort of "live" effect to playback.
-
-    ``<mul>``
-        Sets the difference coefficient (default: 2.5). 0.0 means mono sound
-        (average of both channels), with 1.0 sound will be unchanged, with
-        -1.0 left and right channels will be swapped.
-
-``drc[=method:target]``
-    Applies dynamic range compression. This maximizes the volume by compressing
-    the audio signal's dynamic range. (Formerly called ``volnorm``.)
-
-    ``<method>``
-        Sets the used method.
-
-        1
-            Use a single sample to smooth the variations via the standard
-            weighted mean over past samples (default).
-        2
-            Use several samples to smooth the variations via the standard
-            weighted mean over past samples.
-
-    ``<target>``
-        Sets the target amplitude as a fraction of the maximum for the sample
-        type (default: 0.25).
-
-    .. note::
-
-        This filter can cause distortion with audio signals that have a very
-        large dynamic range.
-
-``ladspa=file:label:[<control0>,<control1>,...]``
-    Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin. This
-    filter is reentrant, so multiple LADSPA plugins can be used at once.
-
-    ``<file>``
-        Specifies the LADSPA plugin library file.
-
-        .. note::
-
-            See also the note about the ``LADSPA_PATH`` variable in the
-            `ENVIRONMENT VARIABLES`_ section.
-    ``<label>``
-        Specifies the filter within the library. Some libraries contain only
-        one filter, but others contain many of them. Entering 'help' here
-        will list all available filters within the specified library, which
-        eliminates the use of 'listplugins' from the LADSPA SDK.
-    ``[<control0>,<control1>,...]``
-        Controls are zero or more ``,`` separated floating point values that
-        determine the behavior of the loaded plugin (for example delay,
-        threshold or gain).
-        In verbose mode (add ``-v`` to the mpv command line), all
-        available controls and their valid ranges are printed. This eliminates
-        the use of 'analyseplugin' from the LADSPA SDK.
-        Note that ``,`` is already used by the option parser to separate
-        filters, so you must quote the list of values with ``[...]`` or
-        similar.
-
-    .. admonition:: Example
-
-        ``mpv --af=ladspa='/usr/lib/ladspa/delay.so':delay_5s:[0.5,0.2] media.avi``
-            Does something.
-
-``karaoke``
-    Simple voice removal filter exploiting the fact that voice is usually
-    recorded with mono gear and later 'center' mixed onto the final audio
-    stream. Beware that this filter will turn your signal into mono. Works
-    well for 2 channel tracks; do not bother trying it on anything but 2
-    channel stereo.
-
-``scaletempo[=option1:option2:...]``
-    Scales audio tempo without altering pitch, optionally synced to playback
-    speed (default).
-
-    This works by playing 'stride' ms of audio at normal speed then consuming
-    'stride*scale' ms of input audio. It pieces the strides together by
-    blending 'overlap'% of stride with audio following the previous stride. It
-    optionally performs a short statistical analysis on the next 'search' ms
-    of audio to determine the best overlap position.
-
-    ``scale=<amount>``
-        Nominal amount to scale tempo. Scales this amount in addition to
-        speed. (default: 1.0)
-    ``stride=<amount>``
-        Length in milliseconds to output each stride. Too high of a value will
-        cause noticeable skips at high scale amounts and an echo at low scale
-        amounts. Very low values will alter pitch. Increasing improves
-        performance. (default: 60)
-    ``overlap=<percent>``
-        Percentage of stride to overlap. Decreasing improves performance.
-        (default: .20)
-    ``search=<amount>``
-        Length in milliseconds to search for best overlap position. Decreasing
-        improves performance greatly. On slow systems, you will probably want
-        to set this very low. (default: 14)
-    ``speed=<tempo|pitch|both|none>``
-        Set response to speed change.
-
-        tempo
-             Scale tempo in sync with speed (default).
-        pitch
-             Reverses effect of filter. Scales pitch without altering tempo.
-             Add ``[ speed_mult 0.9438743126816935`` and ``] speed_mult
-             1.059463094352953`` to your ``input.conf`` to step by musical
-             semi-tones.
-
-             .. warning::
-
-                Loses sync with video.
-        both
-            Scale both tempo and pitch.
-        none
-            Ignore speed changes.
-
-    .. admonition:: Examples
-
-        ``mpv --af=scaletempo --speed=1.2 media.ogg``
-            Would play media at 1.2x normal speed, with audio at normal
-            pitch. Changing playback speed would change audio tempo to match.
-
-        ``mpv --af=scaletempo=scale=1.2:speed=none --speed=1.2 media.ogg``
-            Would play media at 1.2x normal speed, with audio at normal
-            pitch, but changing playback speed would have no effect on audio
-            tempo.
-
-        ``mpv --af=scaletempo=stride=30:overlap=.50:search=10 media.ogg``
-            Would tweak the quality and performace parameters.
-
-        ``mpv --af=format=float,scaletempo media.ogg``
-            Would make scaletempo use float code. Maybe faster on some
-            platforms.
-
-        ``mpv --af=scaletempo=scale=1.2:speed=pitch audio.ogg``
-            Would play media at 1.2x normal speed, with audio at normal pitch.
-            Changing playback speed would change pitch, leaving audio tempo at
-            1.2x.
-
-``lavfi=graph``
-    Filter audio using ffmpeg's libavfilter.
-
-    ``<graph>``
-        Libavfilter graph. See ``lavfi`` video filter for details - the graph
-        syntax is the same.
-
-        .. warning::
-
-            Don't forget to quote libavfilter graphs as described in the lavfi
-            video filter section.
-
-    ``o=<string>``
-        AVOptions.
-
diff --git a/DOCS/man/en/ao.rst b/DOCS/man/en/ao.rst
deleted file mode 100644
index 64c9e308d3..0000000000
--- a/DOCS/man/en/ao.rst
+++ /dev/null
@@ -1,263 +0,0 @@
-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 OSX systems, use ``--ao=coreaudio``.
-
-.. admonition:: Examples
-
-    - ``--ao=alsa,oss,`` Try the ALSA driver, then the OSS driver, then others.
-    - ``--ao=alsa:no-block:device=[hw:0,3]`` Sets noblock-mode and 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.
-    ``no-block``
-        Sets noblock-mode.
-    ``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.
-
-    .. 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.
-
-``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
-
-    ``device_id=<id>``
-        ID of output device to use (0 = default device)
-    ``help``
-        List all available output devices with their IDs.
-
-``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.
-
-``portaudio``
-    PortAudio audio output driver. This works on all platforms, and has
-    extensive MS Windows support.
-
-    .. note:: This driver is not very useful. It was added in the hope of
-              providing portable audio API across all platforms, but turned
-              out semi-broken and underfeatured.
-
-    ``device``
-        Specify the subdevice to use. Giving ``help`` as device name lists all
-        devices found by PortAudio. Devices can be given as numeric values,
-        starting from ``1``.
-
-``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.
-
-``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