manpage: merge new manpage

About a year ago, ubitux converted most of the old manpage from the
hard to maintain nroff format to reStructuredText. This was not merged
back into the master repository immediately. The argument was that the
new manpage still required work to be done. However, progress was very
slow. Even worse: the old manpage wasn't updated, because it was
scheduled for deletion, and updating it would have meant useless work.

Now the situation is that the new manpage still isn't finished, and the
old manpage is grossly out of sync with the player. This is not helpful
for users. Additionally, keeping the new manpage in a separate branch,
while the normal development repository for code had the old manpage,
was very inconvenient, because you couldn't just update the
documentation in the same commit as the code.

Even though the new manpage isn't finished yet, merging it now seems to
be the best course of action. Squash-merge the manpage development
branch [1], revision e89f5dd3f2, which branches from the mplayer2
master branch after revision 159102e0cb.

Committers:

* Clément Bœsch <ubitux@gmail.com> (Initial conversion to RST.)
* Uoti Urpala <uau@mplayer2.org> (Many updates.)
* Myself (Minor edits.)

Most text of the manpage has been directly taken from the old manpage,
because this is a conversion, not a complete rewrite.

[1] http://git.mplayer2.org/uau/mplayer2.git/log/?h=man

7 files changed, 6088 insertions, 0 deletions

diff --git a/DOCS/man/en/af.rst b/DOCS/man/en/af.rst
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+.. _audio_filters:
+
+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``.
+
+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:
+
+resample[=srate[:sloppy[:type]]]
+    Changes the sample rate of the audio stream. Can be used if you have a
+    fixed frequency sound card or if you are stuck with an old sound card that
+    is only capable of max 44.1kHz. This filter is automatically enabled if
+    necessary. It only supports 16-bit integer and float in native-endian
+    format as input.
+
+    <srate>
+        output sample frequency in Hz. The valid range for this parameter is
+        8000 to 192000. If the input and output sample frequency are the same
+        or if this parameter is omitted the filter is automatically unloaded.
+        A high sample frequency normally improves the audio quality,
+        especially when used in combination with other filters.
+    <sloppy>
+        Allow (1) or disallow (0) the output frequency to differ slightly from
+        the frequency given by <srate> (default: 1). Can be used if the
+        startup of the playback is extremely slow.
+    <type>
+        Select which resampling method to use.
+
+        :0: linear interpolation (fast, poor quality especially when
+            upsampling)
+        :1: polyphase filterbank and integer processing
+        :2: polyphase filterbank and floating point processing
+            (slow, best quality)
+
+    *EXAMPLE*:
+
+    ``mplayer --af=resample=44100:0:0``
+        would set the output frequency of the resample filter to 44100Hz using
+        exact output frequency scaling and linear interpolation.
+
+lavcresample[=srate[:length[:linear[:count[:cutoff]]]]]
+    Changes the sample rate of the audio stream to an integer <srate> in Hz.
+    It only supports the 16-bit native-endian format.
+
+    <srate>
+        the output sample rate
+    <length>
+        length of the filter with respect to the lower sampling rate (default:
+        16)
+    <linear>
+        if 1 then filters will be linearly interpolated between polyphase
+        entries
+    <count>
+        log2 of the number of polyphase entries (..., 10->1024, 11->2048,
+        12->4096, ...) (default: 10->1024)
+    <cutoff>
+        cutoff frequency (0.0-1.0), default set depending upon filter length
+
+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. The output sample rate of this filter is same with
+    the input sample rate. When input sample rate is 48kHz, 44.1kHz, or 32kHz,
+    this filter directly use it. Otherwise a resampling filter is
+    auto-inserted before this filter to make the input and output sample rate
+    be 48kHz. You need to specify ``--channels=N`` to make the decoder decode
+    audio into N-channel, then the filter can encode the N-channel input to
+    AC-3.
+
+    <tospdif>
+        Output raw AC-3 stream if zero or not set, output to S/PDIF for
+        passthrough when <tospdif> is set non-zero.
+    <bitrate>
+        The bitrate to encode the AC-3 stream. Set it to either 384 or 384000
+        to get 384kbits.
+
+        Valid values: 32, 40, 48, 56, 64, 80, 96, 112, 128,
+        160, 192, 224, 256, 320, 384, 448, 512, 576, 640.
+
+        Default bitrate is based on the input channel number:
+
+        :1ch: 96
+        :2ch: 192
+        :3ch: 224
+        :4ch: 384
+        :5ch: 448
+        :6ch: 448
+
+    <minchn>
+        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 probably 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 the resample filter before it reaches this filter.
+
+    <g1>:<g2>:<g3>:...:<g10>
+        floating point numbers representing the gain in dB for each frequency
+        band (-12-12)
+
+    *EXAMPLE*:
+
+    ``mplayer --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[:nr:from1:to1:from2:to2:from3:to3:...]
+    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 bigger than the number of input channels
+    empty channels are inserted (except mixing from mono to stereo, then the
+    mono channel is repeated in both of the output channels). If the number of
+    output channels is smaller than the number of input channels the exceeding
+    channels are truncated.
+
+    <nch>
+        number of output channels (1-8)
+    <nr>
+        number of routes (1-8)
+    <from1:to1:from2:to2:from3:to3:...>
+        Pairs of numbers between 0 and 7 that define where to route each
+        channel.
+
+    *EXAMPLE*:
+
+    ``mplayer --af=channels=4:4:0:1:1:0:2:2:3:3 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 was played back, channels 2 and 3
+        would contain silence but 0 and 1 would still be swapped.
+
+    ``mplayer --af=channels=6:4:0:0:0:1:0:2:0:3 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.
+
+format[=format]
+    Convert between different sample formats. Automatically enabled when
+    needed by the sound card or another filter. See also ``--format``.
+
+    <format>
+        Sets the desired format. The general form is 'sbe', where 's' denotes
+        the sign (either 's' for signed or 'u' for unsigned), 'b' denotes the
+        number of bits per sample (16, 24 or 32) and 'e' denotes the
+        endianness ('le' means little-endian, 'be' big-endian and 'ne' the
+        endianness of the computer MPlayer is running on). Valid values
+        (amongst others) are: 's16le', 'u32be' and 'u24ne'. Exceptions to this
+        rule that are also valid format specifiers: u8, s8, floatle, floatbe,
+        floatne, mulaw, alaw, mpeg2, ac3 and imaadpcm.
+
+volume[=v[:sc]]
+    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 set the level for the PCM sound to max, leave this filter out and
+    control the output level to your speakers with the master volume control
+    of the mixer. In case your sound card has a digital PCM mixer instead of
+    an analog one, and you hear distortion, use the MASTER mixer instead. If
+    there is an external amplifier connected to the computer (this is almost
+    always the case), the noise level can be minimized by adjusting the master
+    level and the volume knob on the amplifier until the hissing noise in the
+    background is gone.
+
+    This filter has a second feature: It measures the overall maximum sound
+    level and prints out that level when MPlayer exits. This feature currently
+    only works with floating-point data, use e.g. ``--af-adv=force=5``, or use
+    ``--af=stats``.
+
+    *NOTE*: This filter is not reentrant and can therefore only be enabled
+    once for every audio stream.
+
+    <v>
+        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).
+    <sc>
+        Turns soft clipping on (1) or off (0). 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.
+
+    *EXAMPLE*:
+
+    ``mplayer --af=volume=10.1:0 media.avi``
+        Would amplify the sound by 10.1dB and hard-clip if the sound level is
+        too high.
+
+pan=n[:L00:L01:L02:...L10:L11:L12:...Ln0:Ln1:Ln2:...]
+    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)
+    <Lij>
+        How much of input channel i is mixed into output channel j (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.
+
+    *EXAMPLE*:
+
+    ``mplayer --af=pan=1:0.5:0.5 media.avi``
+        Would down-mix from stereo to mono.
+
+    ``mplayer --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).
+
+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 DVDs with Dolby
+    Digital 5.1 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.
+
+    *EXAMPLE*:
+
+    ``mplayer --af=sub=100:4 --channels=5 media.avi``
+        Would add a sub-woofer 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. Many files
+    with 2 channel audio actually contain matrixed surround sound. Requires a
+    sound card supporting at least 4 channels.
+
+    <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.
+
+    *EXAMPLE*:
+
+    ``mplayer --af=surround=15 --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``.
+
+    *EXAMPLE*:
+
+    ``mplayer --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 sub 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 (default: ``~/.mplayer/mplayer-af_export``)
+    <nsamples>
+        number of samples per channel (default: 512)
+
+    *EXAMPLE*:
+
+    ``mplayer --af=export=/tmp/mplayer-af_export:1024 media.avi``
+        Would export 1024 samples per channel to ``/tmp/mplayer-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.
+
+volnorm[=method:target]
+    Maximizes the volume without distorting the sound.
+
+    <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).
+
+ladspa=file:label[:controls...]
+    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. If ``LADSPA_PATH`` is set,
+        it searches for the specified file. If it is not set, you must supply
+        a fully specified pathname.
+    <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.
+    <controls>
+        Controls are zero or more floating point values that determine the
+        behavior of the loaded plugin (for example delay, threshold or gain).
+        In verbose mode (add ``-v`` to the MPlayer command line), all
+        available controls and their valid ranges are printed. This eliminates
+        the use of 'analyseplugin' from the LADSPA SDK.
+
+comp
+    Compressor/expander filter usable for microphone input. Prevents artifacts
+    on very loud sound and raises the volume on very low sound. This filter is
+    untested, maybe even unusable.
+
+gate
+    Noise gate filter similar to the comp audio filter. This filter is
+    untested, maybe even unusable.
+
+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 value will
+        cause noticable 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.
+
+    *EXAMPLE*:
+
+    ``mplayer --af=scaletempo --speed=1.2 media.ogg``
+        Would playback media at 1.2x normal speed, with audio at normal pitch.
+        Changing playback speed, would change audio tempo to match.
+
+    ``mplayer --af=scaletempo=scale=1.2:speed=none --speed=1.2 media.ogg``
+        Would playback media at 1.2x normal speed, with audio at normal pitch,
+        but changing playback speed has no effect on audio tempo.
+
+    ``mplayer --af=scaletempo=stride=30:overlap=.50:search=10 media.ogg``
+        Would tweak the quality and performace parameters.
+
+    ``mplayer --af=format=floatne,scaletempo media.ogg``
+        Would make scaletempo use float code. Maybe faster on some platforms.
+
+    ``mplayer --af=scaletempo=scale=1.2:speed=pitch audio.ogg``
+        Would playback audio file at 1.2x normal speed, with audio at normal
+        pitch. Changing playback speed, would change pitch, leaving audio
+        tempo at 1.2x.
+
+stats
+    Collects and prints statistics about the audio stream, especially the
+    volume. These statistics are especially intended to help adjusting the
+    volume while avoiding clipping. The volumes are printed in dB and
+    compatible with the volume audio filter.
diff --git a/DOCS/man/en/ao.rst b/DOCS/man/en/ao.rst
new file mode 100644
index 0000000000..4f817774ed
--- /dev/null
+++ b/DOCS/man/en/ao.rst
@@ -0,0 +1,152 @@
+.. _audio_outputs:
+
+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 ',' MPlayer will fall back on drivers not contained
+in the list. Suboptions are optional and can mostly be omitted.
+
+*NOTE*: See ``--ao=help`` for a list of compiled-in audio output drivers.
+
+*EXAMPLE*:
+
+    - ``--ao=alsa,oss,`` Try the ALSA driver, then the OSS driver, then others.
+    - ``--ao=alsa:noblock:device=hw=0.3`` Sets noblock-mode and the device-name
+      as first card, fourth device.
+
+Available audio output drivers are:
+
+alsa
+    ALSA 0.9/1.x audio output driver
+
+    noblock
+        Sets noblock-mode.
+    device=<device>
+        Sets the device name. Replace any ',' with '.' and any ':' with '=' in
+        the ALSA device name. For hwac3 output via S/PDIF, use an "iec958" or
+        "spdif" device, unless you really know how to set it correctly.
+
+alsa5
+    ALSA 0.5 audio output driver
+
+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).
+
+sdl (SDL only)
+    highly platform independent SDL (Simple Directmedia Layer) library audio
+    output driver
+
+    <driver>
+        Explicitly choose the SDL audio driver to use (default: let SDL
+        choose).
+
+jack
+    audio output through JACK (Jack Audio Connection Kit)
+
+    port=<name>
+        Connects to the ports with the given name (default: physical ports).
+    name=<client
+        Client name that is passed to JACK (default: MPlayer [<PID>]). Useful
+        if you want to have certain connections established automatically.
+    (no-)estimate
+        Estimate the audio delay, supposed to make the video playback smoother
+        (default: enabled).
+    (no-)autostart
+        Automatically start jackd if necessary (default: disabled). Note that
+        this seems unreliable and will spam stdout with server messages.
+
+nas
+    audio output through NAS
+
+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
+
+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).
+
+sun (Sun only)
+    native Sun audio output driver
+
+    <device>
+        Explicitly choose the audio device to use (default: ``/dev/audio``).
+
+win32 (Windows only)
+    native Windows waveout audio output driver
+
+dsound (Windows only)
+    DirectX DirectSound audio output driver
+
+    device=<devicenum>
+        Sets the device number to use. Playing a file with ``-v`` will show a
+        list of available devices.
+
+ivtv (IVTV only)
+    IVTV specific MPEG audio output driver. Works with ``--ac=hwmpa`` only.
+
+v4l2 (requires Linux 2.6.22+ kernel)
+    Audio output driver for V4L2 cards with hardware MPEG decoder.
+
+mpegpes (DVB only)
+    Audio output driver for DVB cards that writes the output to an MPEG-PES
+    file if no DVB card is installed.
+
+    card=<1-4>
+        DVB card to use if more than one card is present. If not specified
+        MPlayer will search the first usable card.
+    file=<filename>
+        output filename
+
+null
+    Produces no audio output but maintains video playback speed. Use
+    ``--nosound`` for benchmarking.
+
+pcm
+    raw PCM/wave file writer audio output
+
+    (no-)waveheader
+        Include or do not include the wave header (default: included). When
+        not included, raw PCM will be generated.
+    file=<filename>
+        Write the sound to <filename> instead of the default
+        ``audiodump.wav``. If nowaveheader is specified, the default is
+        ``audiodump.pcm``.
+
+rsound
+    audio output to an RSound daemon
+
+    host=<name/path>
+        Set the address of the server (default: localhost).  Can be either a
+        network hostname for TCP connections or a Unix domain socket path
+        starting with '/'.
+    port=<number>
+        Set the TCP port used for connecting to the server (default: 12345).
+        Not used if connecting to a Unix domain socket.
+
+plugin
+    plugin audio output driver
diff --git a/DOCS/man/en/mplayer.1 b/DOCS/man/en/mplayer-old.1
index 0e1ff45516..0e1ff45516 100644
--- a/DOCS/man/en/mplayer.1
+++ b/DOCS/man/en/mplayer-old.1
diff --git a/DOCS/man/en/mplayer.rst b/DOCS/man/en/mplayer.rst
new file mode 100644
index 0000000000..621df0431c
--- /dev/null
+++ b/DOCS/man/en/mplayer.rst
@@ -0,0 +1,685 @@
+mplayer2 manual page
+####################
+
+Synopsis
+========
+
+| **mplayer** [options] [file|URL|playlist|-]
+| **mplayer** [options] file1 [specific options] [file2] [specific options]
+| **mplayer** [options] {group of files and options} [group-specific options]
+| **mplayer** [br]://[title][/device] [options]
+| **mplayer** [dvd|dvdnav]://[title|[start\_title]-end\_title][/device] [options]
+| **mplayer** \vcd://track[/device]
+| **mplayer** \tv://[channel][/input_id] [options]
+| **mplayer** radio://[channel|frequency][/capture] [options]
+| **mplayer** \pvr:// [options]
+| **mplayer** \dvb://[card\_number@]channel [options]
+| **mplayer** \mf://[filemask|\@listfile] [-mf options] [options]
+| **mplayer** [cdda|cddb]://track[-endtrack][:speed][/device] [options]
+| **mplayer** \cue://file[:track] [options]
+| **mplayer** [file|mms[t]|http|http\_proxy|rt[s]p|ftp|udp|unsv|icyx|noicyx|smb]:// [user:pass\@]URL[:port] [options]
+| **mplayer** \sdp://file [options]
+| **mplayer** \mpst://host[:port]/URL [options]
+| **mplayer** \tivo://host/[list|llist|fsid] [options]
+
+
+DESCRIPTION
+===========
+
+**mplayer** is a movie player for Linux (runs on many other platforms and CPU
+architectures, see the documentation). It supports a wide variety of video
+file formats, audio and video codecs, and subtitle types. Special input URL
+types are available to read input from a variety of sources other than disk
+files. Depending on platform, a variety of different video and audio output
+methods are supported.
+
+Usage examples to get you started quickly can be found at the end of this man
+page.
+
+
+INTERACTIVE CONTROL
+===================
+
+MPlayer has a fully configurable, command-driven control layer which allows you
+to control MPlayer using keyboard, mouse, joystick or remote control (with
+LIRC). See the ``--input`` option for ways to customize it.
+
+keyboard control
+----------------
+
+LEFT and RIGHT
+    Seek backward/forward 10 seconds. Shift+arrow does a 1 second exact seek
+    (see ``--hr-seek``; currently modifier keys like shift only work if used in
+    an X output window).
+
+UP and DOWN
+    Seek forward/backward 1 minute. Shift+arrow does a 5 second exact seek (see
+    ``--hr-seek``; currently modifier keys like shift only work if used in an X
+    output window).
+
+PGUP and PGDWN
+    Seek forward/backward 10 minutes.
+
+[ and ]
+    Decrease/increase current playback speed by 10%.
+
+{ and }
+    Halve/double current playback speed.
+
+BACKSPACE
+    Reset playback speed to normal.
+
+< and >
+    Go backward/forward in the playlist.
+
+ENTER
+    Go forward in the playlist, even over the end.
+
+HOME and END
+    next/previous playtree entry in the parent list
+
+INS and DEL (ASX playlist only)
+    next/previous alternative source.
+
+p / SPACE
+    Pause (pressing again unpauses).
+
+.
+    Step forward. Pressing once will pause movie, every consecutive press will
+    play one frame and then go into pause mode again.
+
+q / ESC
+    Stop playing and quit.
+
+U
+    Stop playing (and quit if ``--idle`` is not used).
+
+\+ and -
+    Adjust audio delay by +/- 0.1 seconds.
+
+/ and *
+    Decrease/increase volume.
+
+9 and 0
+    Decrease/increase volume.
+
+( and )
+    Adjust audio balance in favor of left/right channel.
+
+m
+    Mute sound.
+
+\_ (MPEG-TS, AVI and libavformat only)
+    Cycle through the available video tracks.
+
+\# (DVD, Blu-ray, MPEG, Matroska, AVI and libavformat only)
+    Cycle through the available audio tracks.
+
+TAB (MPEG-TS and libavformat only)
+    Cycle through the available programs.
+
+f
+    Toggle fullscreen (see also ``--fs``).
+
+T
+    Toggle stay-on-top (see also ``--ontop``).
+
+w and e
+    Decrease/increase pan-and-scan range.
+
+o
+    Toggle OSD states: none / seek / seek + timer / seek + timer + total time.
+
+d
+    Toggle frame dropping states: none / skip display / skip decoding (see
+    ``--framedrop`` and ``--hardframedrop``).
+
+v
+    Toggle subtitle visibility.
+
+j and J
+    Cycle through the available subtitles.
+
+y and g
+    Step forward/backward in the subtitle list.
+
+F
+    Toggle displaying "forced subtitles".
+
+a
+    Toggle subtitle alignment: top / middle / bottom.
+
+x and z
+    Adjust subtitle delay by +/- 0.1 seconds.
+