Almost nothing was left of it.
The only thing this commit actually removes is support for reading
input commands from stdin. But you can emulate this via:
--input-file=/dev/stdin --input-terminal=no
However, this won't work on Windows. Just use a named pipe.
Some ALSA plugins take non-interleaved audio, but treat it as
interleaved, which results in various funny bugs. Users keep hitting
this issue, and it just doesn't seem worth the trouble.
CC: @mpv-player/stable
Add an option that enables using native PulseAudio auto-updated timing
information, instead of the manual calculations added in mplayer2 times.
You can use --ao=pulse:no-latency-hacks to enable the new code. The code
is almost the same as the code that was removed with commit de435ed5,
but I didn't readd some bits I didn't understand. Likewise, the option
will disable the code added with that commit.
In my tests this seemed to work well, though the A/V sync display looks
funny when seeking.
The default is still the old behavior.
See issue #959.
"loadfile filename append-play" will now always append the file to the
playlist, and if nothing is playing yet, start playback. I don't want to
change the semantics of "append" mode, so a new mode is needed.
Probably fixes issue #950.
Useful for Windows stuff. Actually, ENCA support should catch this, but,
well, whatever, everyone seems to hate ENCA.
Detection with BOM is trivial, although it needs some hackery to
integrate it with the existing autodetection support. For one, change
the default value of --sub-codepage to make this easier.
Probably fixes issue #937 (the second part).
The MPlayer style syntax ("-mf fps=10:type=png") was removed a while
ago, and now only the flat variants ("--mf-fps=10" etc.) work.
CC: @mpv-player/stable
This adds a thread to the demuxer which reads packets asynchronously.
It will do so until a configurable minimum packet queue size is
reached. (See options.rst additions.)
For now, the thread is disabled by default. There are some corner cases
that have to be fixed, such as fixing cache behavior with webradios.
Note that most interaction with the demuxer is still blocking, so if
e.g. network dies, the player will still freeze. But this change will
make it possible to remove most causes for freezing.
Most of the new code in demux.c actually consists of weird caches to
compensate for thread-safety issues (with the previously single-threaded
design), or to avoid blocking by having to wait on the demuxer thread.
Most of the changes in the player are due to the fact that we must not
access the source stream directly. the demuxer thread already accesses
it, and the stream stuff is not thread-safe.
For timeline stuff (like ordered chapters), we enable the thread for the
current segment only. We also clear its packet queue on seek, so that
the remaining (unconsumed) readahead buffer doesn't waste memory.
Keep in mind that insane subtitles (such as ASS typesetting muxed into
mkv files) will practically disable the readahead, because the total
queue size is considered when checking whether the minimum queue size
was reached.
The intention is to make it obvious which mpv releases certain changes
will apply to.
Also attempt to fix RST formatting of the list. This is not very proper,
but probably good enough.
These consult the vertical resolution, matching against 576 for
PAL and 480/486 for NTSC. The documentation has also been updated.
Signed-off-by: wm4 <wm4@nowhere>
Notably, we now conform to SMPTE 428-1-2006 when decoding XYZ12 input,
and we can support rendering intents other than colorimetric when
converting between BT.709 and BT.2020, like with :srgb or :icc-profile.
This add support for reading primary information from lavc, categorized
into BT.601-525, BT.601-625, BT.709 and BT.2020; and passes it on to the
vo. In vo_opengl, we always generate the 3dlut against the wider BT.2020
and transform our source into this colorspace in the shader.
For remarks, pretty much see the manpage additions. Could help with
network streams that require too much seeking (maybe), or might be
extended to help with the use case of watching and downloading a file
at the same time.
In general, it might be a useless feature and could be removed again.
Also clarify the semantics.
It seems --idx didn't do anything. Possibly it used to change how the
now removed legacy demuxers like demux_avi used to behave. Or maybe
it was accidental.
--forceidx basically becomes --index=force. It's possible that new
index modes will be added in the future, so I'm keeping it
extensible, instead of e.g. creating --force-index.
Similar to previous commits.
This also renames --doubleclick-time to --input-doubleclick-time, and
--key-fifo-size to --input-key-fifo-size. We could keep the old names,
but these options are very obscure, and renaming them seems better for
consistency.
Additionally to removing the global variables, this makes the options
more uniform. --ssf-... becomes --sws-..., and --sws becomes --sws-
scaler. For --sws-scaler, use choices instead of magic integer values.
Pretty much nothing changes, but using -tv-scan with suboptions doesn't
work anymore (instead of "-tv-scan x" it's "-tv scan-x" now). Flat
options ("-tv-scan-x") stay compatible.
Convert all these commands to properties. (Except tv_last_channel, not
sure what to do with this.) Also, internally, don't access stream
details directly, but dispatch commands with stream ctrls.
Many of the new properties are a bit strange, because they're write-
only. Also remove some OSD output these commands produced, because I
couldn't be bothered to port these.
In general, this makes everything much cleaner, and will also make it
easier to e.g. move the demuxer to its own thread.
Don't bother updating input.conf, but changes.rst documents how old
commands map to the new ones.
Mostly untested, due to lack of hardware.
Basically, this allows gapless playback with similar files (including
the ordered chapter case), while still being robust in general.
The implementation is quite simplistic on purpose, in order to avoid
all the weird corner cases that can occur when creating the filter
chain. The consequence is that it might do not-gapless playback in
more cases when needed, but if that bothers you, you still can use
the normal gapless mode.
Just using "--gapless-audio" or "--gapless-audio=yes" selects the old
mode.
A bit verbose, but less misleading. In most cases, the API user probably
actually wants mpv_terminate_destroy() instead, so the less-useful
function shouldn't have a simnpler name anyway.
(The old "force" choice of that option is renamed to "force-default".)
This allows overriding native ASS script subtitle styles with the style
provided by the --sub-text-* options (like --sub-text-font etc.). This
is disabled by default, and needs to be explicitly enabled with the
--ass-style-override=force option and input property.
This uses in fact exactly the same options (--sub-text-*) and semantics
as the ones used to configure unstyled text subtitles.
It's recommended to combine this with this in the mpv config file:
ass-force-style="ScaledBorderAndShadow=1" # work around dumb libass behavior
Also, adding a key binding to toggle this behavior should be added,
because overriding can easily break:
L cycle ass-style-override
This would cycle override behavior on Shift+L and allows quickly
disabling/enabling style overrides.
Note: ASS should be considered a vector format rather than a subtitle
format. There is no easy or reliable way to determine whether the style
of a given subtitle event can be changed without destroying visuals or
not. This patch relies on a simple heuristic, which often works and
often breaks.
This simply writes the file name as a comment to the top of the watch later
config file.
It can be useful to the user for determining whether a watch later config file
can be manually removed (e.g. in case the corresponding media file has been
deleted) or not.
If a single person complains, I will readd it. But I don't expect that
this will happen.
The main reason for removing this is that it's some of the most unclean
code remaining, it's unmaintained, and I've never ever heard of someone
using it.
Commit e2e450f9 started making use of luaL_register(), but OF COURSE
this function disappeared in Lua 5.2, and was replaced with a 5.2-only
alternative, slightly different mechanism.
So just NIH our own function. This is actually slightly more correct,
since it forces the user to call "require" to actually make the module
visible for builtin C-only modules other than "mp". Fix autoload.lua
accordingly.
We need this only because Lua's stdlib is so scarce. Lua doesn't intend
to include a complete stdlib - they confine themselves to standard C,
both for portability reasons and to keep the code minimal. But standard
C does not provide much either.
It would be possible to use Lua POSIX wrapper libraries, but that would
be a messy (and unobvious) dependency. It's better to implement the
missing functions ourselves, as long as they're small in number.
stream.start_pos was needed for optical media only, and (apparently) not
for very good reasons. Just get rid of it.
For stream_dvd, we don't need to do anything. Byte seeking was already
removed from it earlier.
For stream_cdda and stream_vcd, emulate the start_pos by offsetting the
stream pos as seen by the rest of mpv.
The bits in discnav.c and loadfile.c were for dealing with the code
seeking back to the start in demux.c. Handle this differently by
assuming the demuxer is always initialized with the stream at start
position, and instead seek back if initializing the demuxer fails.
Remove the --sb option, which worked by modifying stream.start_pos. If
someone really wants this option, it could be added back by creating a
"slice" stream (actually ffmpeg already has such a thing).
The quit command has an optional argument that is used as exit code.
Extend that to the quit_watch_later command. Actually, unify the
implementations of the two commands.
Requested in #798.
Some options change from percentages to number of kilobytes; there are
no cache options using percentages anymore.
Raise the default values. The cache is now 25000 kilobytes, although if
your connection is slow enough, the maximum is probably never reached.
(Although all the memory will still be used as seekback-cache.)
Remove the separate --audio-file-cache option, and use the cache default
settings for it.
Try to get the "new" code path (using NetWM/EWMH) free of hacks done for
the sake of old WMs or the no-WM case.
Implement --fs-screen using _NET_WM_FULLSCREEN_MONITORS.
VapourSynth won't just filter multiple frames at once on its own. You
have to request multiple frames at once manually. This is what this
commit introduces: a sub-option controls how many frames will be
requested at once. This also changes the semantics of the maxbuffer sub-
option, now renamed to buffered-frames.
The situation has changed a bit since the days of mplayer2, so we can
use more/less diplomatic wording. Merge the two sections listing
changes from MPlayer and mplayer2. Mention the client API and Lua
scripting as alternatives to slave mode.
I'm calling MPlayer code "horrible". This is not meant as an offense,
but after turning around almost every line of MPlayer code, I believe
I have a right to say this. Sorry. I would say that MPlayer has a
surprisingly sane and simple architecture (for what it is), but much
of it drowned under a load of evil hacks or not-cleaned-up-yet code.
These are now equivalent to combining commands with the "cycle pause" or
"set pause" commands, and thus are not needed anymore. They were also
obscure and undocumented.
The many "boxes" in this entry were causing rst2pdf to fail because it
couldn't figure out where to break the page. Make the boxes smaller by
removing semi-redundant examples. Also try and make surrounding text a
little shorter by rewording.
This is done after filters, so things like framerate-doubling
deinterlacing is accounted for.
Unfortunately, framedropping can cause inaccuracies (especially after
precise seeks), and we can't really know when that happens. Even though
we know that the decoder might drop a frame if we request it to do so,
we don't know when the dropped frame will start or stop affecting the
video filter chain. Video filters can have frames buffered, and we
can't tell at which point the dropped frame would have been output.
It's not even possible to mark a discontinuity after seek, because
again we don't know if the filter chain still has the discontinuity
within its buffers.
So we have to live with the fact that the output of this property can
be completely broken after seek, unless --no-hr-seek-framedrop is used.
This allows disabling of decoder framedrop during hr-seek.
It's basically another useless option, but it will help exploring
whether this framedropping really makes seeking faster, or whether
disabling it helps with precise seeking (especially frame backstepping).
Also remove MSGL_SMODE and friends.
Note: The indent in options.rst was added to work around a bug in
ReportLab that causes the PDF manual build to fail.
