If VO deinterlacing is unavailable, try to insert vf_yadif.
If vf_lavfi is available, actually use vf_yadif from libavfilter. The
libavfilter version of this filter is faster, more correct, etc., so it
is preferred. Unfortunately vf_yadif obviously doesn't support
VFCTRL_GET/SET_DEINTERLACE, and with the current state of the
libavfilter API, it doesn't look like there is any simple way to
emulate it. Instead, we simply insert the filter with a specific label,
and if deinterlacing is to be disabled, the filter is removed again by
label.
This won't do the right thing if the user inserts any deinterlacing
filter manually (except native vf_yadif, which understands the VFCTRL).
For example, with '-vf lavfi=yadif', pressing 'D' (toggle deinterlacing)
will just insert a second deinterlacer filter. In these cases, the user
is supposed to map a command that toggles his own filter instead of
using 'D' and the deinterlace property.
The same applies if the user wants to pass different parameters to the
deinterlacer filters.
Also add a "raw" prefix for commands, which prevents property expansion.
The idea is that if the commands are generated by a program, it doesn't
have to know whether the command expands properties or not.
Add the "vf" command, which allows changing the video filter chain at
runtime. For example, the 'y' key could be bound to toggle deinterlacing
by adding 'y vf toggle yadif' to the input.conf.
Reconfiguring the video filter chain normally resets the VO, so that it
will be "stuck" until a new video frame is rendered. To mitigate this, a
seek to the current position is issued when the filter chain is changed.
This is done only if playback is paused, because normal playback will
show an actual new frame quickly enough.
If vdpau hardware decoding is used, filter insertion (whether it fails
or not) will break the video for a while. This is because vo_vdpau
resets decoding related things on vo_config().
This helps passing the channel layout correctly from decoder to audio
filter chain. (Because that part "reuses" the demuxer level codec
parameters, which is very disgusting.)
Note that ffmpeg stuff already passed the channel layout via
mp_copy_lav_codec_headers(). So other than easier dealing with the
demuxer/decoder parameters mess, there's no real advantage to doing
this.
Make the --channels option accept a channel map. Since simple numbers
map to standard layouts with the given number of channels, this is
downwards compatible. Likewise for demux_rawaudio.
Makes it easier to understand... maybe. It's still pretty strange how
this function may either queue the seek or seek immediately. The way
it actually works doesn't change, queuing the seek is just moved into
the function.
Also add a execute_queued_seek() function, which resets the queue state
correctly.
A "watch later" command is now mapped to Shift+Q. This quits the player
and stores the playback state in a config file in ~/.mpv/watch_later/.
When calling the player with the same file again, playback is resumed
at that time position.
It's also possible to make mpv save playback state always on quit with
the --save-position-on-quit option. Likewise, resuming can be disabled
with the --no-resume-playback option.
This also attempts to save some playback parameters, like fullscreen
state or track selection. This will unconditionally override config
settings and command line options (which is probably not what you would
expect, but in general nobody will really care about this). Some things
are not backed up, because that would cause various problems. Additional
subtitle files, video filters, etc. are not stored because that would be
too hard and fragile. Volume/mute state are not stored because it would
mess up if the system mixer is used, or if the system mixer was
readjusted in the meantime.
Basically, the tradeoff between perfect state restoration and
complexity/fragility makes it not worth to attempt to implement
it perfectly, even if the result is a little bit inconsistent.
Now vid/aid/sid can be used as properties. video/audio/sub still work,
but they are aliases for the "real" properties.
This guarantees that options/properties use the same value range. One
consequence is that the video/audio/sub properties return "no" as value
if no track is selected instead of -1.
Rename the struct MPOpts "start_pause" field to "pause". Store the user-
pause state in that field, so that both runtime pause toggling and the
--pause switch change the same variable. Simplify the initialization of
pause so that using --pause and changing the file while paused is
exactly the same case (changing the file while paused doesn't unpause,
this has been always this way).
Also make it a bit more consistent. Before, starting with --pause would
reset the pause state for every file, instead of following the usual
semantics for option switches (compare with behavior of --fs).
The core pauses and unpauses automatically to wait for the network
cache (also known as buffering). This conflicted with user pause
control, and was perceived as if the player was unresponsive and/or
the cache just overturned the user's decisions.
Change it so that the actual pause state and the pause state as
intended by the user never conflict. If the user toggles pause, the
pause state will be in the expected state as soon as the cache is
loaded.
Allows stepping back one frame via the frame_back_step inout command,
bound to "," by default.
This uses the precise seeking facility, and a perfect frame index built
on the fly. The index is built during playback and precise seeking, and
contains (as of this commit) the last 100 displayed or skipped frames.
This index is used to find the PTS of the previous frame, which is then
used as target for a precise seek. If no PTS is found, the core attempts
to do a seek before the current frame, and skip decoded frames until the
current frame is reached; this will create a sufficient index and the
normal backstep algorithm can be applied.
This can be rather slow. The worst case for backstepping is about the
same as the worst case for precise seeking if the previous frame can be
deduced from the index. If not, the worst case will be twice as slow.
There's also some minor danger that the index is incorrect in case
framedropping is involved. For framedropping due to --framedrop, this
problem is ignored (use of --framedrop is discouraged anyway). For
framedropping during precise seeking (done to make it faster), we try
to not add frames to the index that are produced when this can happen.
I'm not sure how well that works (or if the logic is sane), and it's
sure to break with some video filters. In the worst case, backstepping
might silently skip frames if you backstep after a user-initiated
precise seek. (Precise seeks to do indexing are not affected.)
Likewise, video filters that somehow change timing of frames and do not
do this in a deterministic way (i.e. if you seek to a position, frames
with different timings are produced than when the position is reached
during normal playback) will make backstepping silently jump to the
wrong frame. Enabling/disabling filters during playback (like for
example deinterlacing) will have similar bad effects.
Make sure automatically inserted filters are removed on full reinit
(they are re-added later if they are really needed). Automatically
inserted filters were never explicitly removed, instead, it was
expected that redundant conversion filters detach themselves. This
didn't work if there were several chained format conversion filters,
e.g. s16le->floatle->s16le, which could result from repeated filter
insertion and removal. (format filters detach only if input format and
output format are the same.)
Further, the dummy filter (which exists only because af.c can't handle
an empty filter chain for some reason) could introduce bad conversions
due to how the format negotiation works. Change the code so that the
dummy filter never takes part on format negotiation. (It would be better
to fix format negotiation, but that would be much more complicated and
would involving fixing all filters.)
Simplify af_reinit() and remove the start audio filter parameter. This
means format negotiation and filter initialization is run more often,
but should be harmless.
Add new properties "dwidth" and "dheight", which contain the video
size as known by the VO (not necessarily what the VO makes out of them,
i.e. without window scaling and panscan).
Move them into per-instance structs. This should get rid of all global
variables in mplayer.c (not counting those referenced by cfg-mplayer.h).
In core/input/ar.c, just remove checking the slave_mode variable. I'm
not sure what this code was supposed to achieve, but slave mode is
broken, slave mode is actually infeasible on OSX (ar.c is completely OSX
specific), and the correct way of doing this would be to disable this
input device per command line switch.
Separate the video output options from the big MPOpts structure and also only
pass the new mp_vo_opts structure to the vo backend.
Move video_driver_list into mp_vo_opts
Removes almost every global variabel in vo.h and puts them in a special struct
in MPOpts for video output related options.
Also we completly remove the options/globals pts and refresh rate because
they were unused.
The message reads: "Screenshot: filename", where the filename is what
mpv passes to fopen(). It will also show error messages when saving the
screenshot fails.
This means a commands like "seek 13:00 absolute" actually behaves like
"--start=13:00", instead of interpreting the argument as fraction as
with normal float options. This is probably slightly closer to what
you'd expect.
As a consequence, the seek argument's type changes from float to double
internally.
You can just use --wid=0 if you really want this.
This only worked/works for X11, and even then it might interact badly
with most desktop environments. All the option did was setting --wid to
0, and the property did nothing.
"End of file" was printed to the terminal instead of "Quit" when exiting
with the "quit" slave command (closing the window and such). Note that
it will still print EOF when it exists because the end of the playlist
is reached.
Do some other (not strictly related) simplifications.
Until now, setting a property that is not available (e.g. deinterlacing
if not using vdpau and no deinterlacing filter is inserted) silently
failed (except for a messager on the terminal). Instead show on the OSD
that the property is unavailable.
Use codec names instead of FourCCs to identify codecs. Rewrite how
codecs are selected and initialized. Now each decoder exports a list
of decoders (and the codec it supports) via add_decoders(). The order
matters, and the first decoder for a given decoder is preferred over
the other decoders. E.g. all ad_mpg123 decoders are preferred over
ad_lavc, because it comes first in the mpcodecs_ad_drivers array.
Likewise, decoders within ad_lavc that are enumerated first by
libavcodec (using av_codec_next()) are preferred. (This is actually
critical to select h264 software decoding by default instead of vdpau.
libavcodec and ffmpeg/avconv use the same method to select decoders by
default, so we hope this is sane.)
The codec names follow libavcodec's codec names as defined by
AVCodecDescriptor.name (see libavcodec/codec_desc.c). Some decoders
have names different from the canonical codec name. The AVCodecDescriptor
API is relatively new, so we need a compatibility layer for older
libavcodec versions for codec names that are referenced internally,
and which are different from the decoder name. (Add a configure check
for that, because checking versions is getting way too messy.)
demux/codec_tags.c is generated from the former codecs.conf (minus
"special" decoders like vdpau, and excluding the mappings that are the
same as the mappings libavformat's exported RIFF tables). It contains
all the mappings from FourCCs to codec name. This is needed for
demux_mkv, demux_mpg, demux_avi and demux_asf. demux_lavf will set the
codec as determined by libavformat, while the other demuxers have to do
this on their own, using the mp_set_audio/video_codec_from_tag()
functions. Note that the sh_audio/video->format members don't uniquely
identify the codec anymore, and sh->codec takes over this role.
Replace the --ac/--vc/--afm/--vfm with new --vd/--ad options, which
provide cover the functionality of the removed switched.
Note: there's no CODECS_FLAG_FLIP flag anymore. This means some obscure
container/video combinations (e.g. the sample Film_200_zygo_pro.mov)
are played flipped. ffplay/avplay doesn't handle this properly either,
so we don't care and blame ffmeg/libav instead.
This could write .edl files in MPlayer's format. Support for playing
these files has been removed from mplayer2 quite a while ago. (mplayer2
can play its own, "new" .edl format, but does not support writing it.)
Since this is a rather obscure functionality, and it's not really clear
how it should behave (e.g. what should it do if a new file is played),
and wasn't all that great to begin with (what if you made a mistake?
the "edl_mark" command sucks for editing), get rid of it.
Suggestions how to reimplement this in a nicer way are welcome. If it's
just about retrieving timecodes, this in input.conf will do:
KEY print_text "position: ${=time-pos}"
Explicitly advancing the playlist with input commands ("playlist_next")
didn't jump back to the first file, if the current file was the last on
the playlist and looping was enabled.
Fix this and make the behavior with explicit input and playback EOF the
same.
Also add a minor feature: if looping is enabled, and the current file is
the first on the playlist, going back one entry jumps to the last
playlist entry (without changing loop count).
Fixes#22.
This function sucks and apparently is not very portable (at least on
mingw, the configure check fails). Also remove the emulation of that
function from osdep/strsep*, and remove the configure check.
ffmpeg recently added a demuxer that can read vobsubs (pairs of .sub and
.idx files). Get rid of the internal vobsub reader, and use the ffmpeg
demuxer instead.
Sneak in an unrelated manpage change (autosub default).
The playback status symbol in the OSD status display on video (such as
displayed when seeking or with the show_progress input command)
sometimes kept displaying the last seek, without resetting the symbol.
(For example: disable the OSD, seek, enable the OSD, run show_progress;
but also other cases.)
The main reason for that was the code clearing the OSD bar is also
responsible for clearing the osd_function (which stores the playback
symbol). If no OSD bar was set, the osd_function was never reset.
Fix by always setting the timer for clearing the OSD bar and the
osd_function whenever the osd_function is set. Clearing the OSD bar
when it wasn't set is OK. If the OSD bar is set some time after
osd_function is set, the timer is overwritten - that's a good thing,
as it makes both disappear from the screen at exactly the same time.
Always reset osd_function to 0 and determine the playback status
explicitly from mpctx->paused when displaying the status on screen.
Make more aspects of the OSD font customizable. This also affects the
font used for unstyled subtitles (such as SRT), or when using the
--no-ass option. This adds back some customizability that was lost with
commit 74e7a1 (osd: use libass for OSD rendering).
Removed options:
--ass-border-color
--ass-color
--font
--subfont
--subfont-text-scale
Added options:
--osd-color
--osd-border
--osd-back-color
--osd-shadow-color
--osd-font
--osd-font-size
--osd-border-size
--osd-margin-x
--osd-margin-y
--osd-shadow-offset
--osd-spacing
--sub-scale
The font size is now specified in pixels as it would be rendered on a
window with a height of 720 pixels. OSD and subtitles are always scaled
with the window height, so specifying or expecting an absolute font
size doesn't make sense.
Such scaled pixel units are used to specify font border etc. as well.
(Note: the font size is directly passed to libass. How the fonts are
actually rasterized is outside of our control, but in theory ASS font
sizes map to "script" pixels and then are scaled to screen size.)
The default settings should be about the same, with slight difference
due to rounding to the new scales.
The OSD and subtitle fonts are not separately configurable. It has
limited use and would double the number of newly added options, which
would be more confusing than helpful. It could be easily added later,
should the need arise.
Other small details that change:
- ASS_Style.Encoding is not set to -1 for subs anymore
(assuming subs use VSFilter direction in -no-ass mode too)
- use a different WrapStyle for OSD
- ASS forced styles are not applied to OSD
sub_remove remove an external subtitle track, for whatever this may be
needed.
sub_reload removes and re-adds an external subtitle track.
Also rename sub_load to sub_add, because that seems to be more in line
with sub_remove.
Finish renaming directories and moving files. Adjust all include
statements to make the previous commit compile.
The two commits are separate, because git is bad at tracking renames
and content changes at the same time.
Also take this as an opportunity to remove the separation between
"common" and "mplayer" sources in the Makefile. ("common" used to be
shared between mplayer and mencoder.)
Tis drops the silly lib prefixes, and attempts to organize the tree in
a more logical way. Make the top-level directory less cluttered as
well.
Renames the following directories:
libaf -> audio/filter
libao2 -> audio/out
libvo -> video/out
libmpdemux -> demux
Split libmpcodecs:
vf* -> video/filter
vd*, dec_video.* -> video/decode
mp_image*, img_format*, ... -> video/
ad*, dec_audio.* -> audio/decode
libaf/format.* is moved to audio/ - this is similar to how mp_image.*
is located in video/.
Move most top-level .c/.h files to core. (talloc.c/.h is left on top-
level, because it's external.) Park some of the more annoying files
in compat/. Some of these are relicts from the time mplayer used
ffmpeg internals.
sub/ is not split, because it's too much of a mess (subtitle code is
mixed with OSD display and rendering).
Maybe the organization of core is not ideal: it mixes playback core
(like mplayer.c) and utility helpers (like bstr.c/h). Should the need
arise, the playback core will be moved somewhere else, while core
contains all helper and common code.