There was a MPOpts fullscreen field, a mp_vo_opts.fs field, and
VOFLAG_FULLSCREEN. Remove all these and introduce a
mp_vo_opts.fullscreen flag instead.
When VOs receive VOCTRL_FULLSCREEN, they are supposed to set the
current fullscreen mode to the state in mp_vo_opts.fullscreen. They
also should do this implicitly on config().
VOs which are capable of doing so can update the mp_vo_opts.fullscreen
if the actual fullscreen mode changes (e.g. if the user uses the
window manager controls). If fullscreen mode switching fails, they
can also set mp_vo_opts.fullscreen to the actual state.
Note that the X11 backend does almost none of this, and it has a
private fs flag to store the fullscreen flag, instead of getting it
from the WM. (Possibly because it has to deal with broken WMs.)
The fullscreen option has to be checked on config() to deal with
the -fs option, especially with something like:
mpv --fs file1.mkv --{ --no-fs file2.mkv --}
(It should start in fullscreen mode, but go to windowed mode when
playing file2.mkv.)
Wayland changes by: Alexander Preisinger <alexander.preisinger@gmail.com>
Cocoa changes by: Stefano Pigozzi <stefano.pigozzi@gmail.com>
Instead of handling colorspaces with VFCTRLs/VOCTRLs, make them part of
the normal video format negotiation. The colorspace is passed down like
other video params with config/reconfig calls.
Forcing colorspaces (via the --colormatrix options and properties) is
handled differently too: if it's changed, completely reinit the video
chain. This is slower and requires a precise seek to the same position
to perform an update, but it's simpler and less bug-prone. Considering
switching the colorspace at runtime by user-interaction is a rather
obscure feature, this is a good change.
The colorspace VFCTRLs and VOCTRLs are still kept. The VOs rely on it,
and would have to be changed to get rid of them. We'll do that later,
and convert them incrementally instead of in one go.
Note that controlling the output range now always works on VO level.
Basically, this means you can't get vf_scale to output full-range YUV
for whatever reason. If that is really wanted, it should be a vf_scale
option. the previous behavior didn't make too much sense anyway.
This commit fixes a few bugs (such as playing RGB video and converting
that to YUV with vf_scale - a recent commit broke this and forced the
VO to display YUV as RGB if possible), and might introduce some new
ones.
From now on, usage of these macros is encouraged over using FFMAX and
FFMIN. FFMAX and FFMIN are perfectly fine, and the added macros are
actually exactly the same as the FFMAX and FFMIN definitions. But they
require including libavutil headers, and certain differences between
Libav and FFmpeg very often introduced breakages if these macros were
somehow not defined because a header was not recursively included.
Defining this macro on our own is the best way to escape from this
annoying issue.
DVD playback had some trouble with PTS resets: libavformat's genpts
feature would try reading until EOF (worst case) to find a new usable
PTS in case a packet's PTS is not set correctly. Especially with slow
DVD access, this would make the player to appear frozen.
Reimplement it partially in demux_lavf.c, and use that code in the DVD
case. This is heavily "inspired" by the code in av_read_frame from
libavformat/utils.c. The difference is that we stop reading if no PTS
has been found after 50 packets (consider this a heuristic). Also, we
don't bother with the PTS wrapping and last-frame-before-EOF handling.
Even with normal PTS wraps, the player frontend will go to hell for the
duration of a frame anyway, and should recover quickly after that.
The terribleness of this commit is mostly that we duplicate libavformat
functionality, and that we suddenly need a packet queue.
Move the reading loop from read_all_fd_events to read_events. If
got_new_events is set when calling read_events, don't actually wait
and set the timeout to 0.
(Note that not waiting is sort of transparent to the caller: the caller
is just supposed to execute the event loop again, and then it will
actually wait. mplayer.c handles this correctly.)
This might reduce latency with some input sources.
Removes some code duplication. Also restructure the input waiting code
a bit: split the select() loop into a input_wait_read() function. On
systems which do not have POSIX select(), this function has an alternate
implementation, which waits unconditionally.
This was useless for anything but the raw demuxers. In most cases, this
would most likely lead to display of bogus duration values, because the
bitrates used are per-track, not the total file bitrate. There was
actually no case left where this code was helpful.
Note that demux_lavf has its own code for this using the total file
bitrate. Also, mplayer.c can calculate the playback percentage from
current file position / current file size. This is not removed.
All demuxers make a reasonable effort to set packet timestamps, and thus
support correct-pts mode. This commit also implicitly switches
demux_rawvideo to correct-pts mode.
We still allow demuxers to disable correct-pts mode in theory.
Get rid of the strange and messy reliance on DEMUXER_TYPE_ constants.
Instead of having two open functions for the demuxer callbacks (which
somehow are both optional, but you can also decide to implement both...),
just have one function. This function takes a parameter that tells the
demuxer how strictly it should check for the file headers. This is a
nice simplification and allows more flexibility.
Remove the file extension code. This literally did nothing (anymore).
Change demux_lavf so that we check our other builtin demuxers first
before libavformat tries to guess by file extension.
This removes the dependency on DEMUXER_TYPE_* and the file_format
parameter from the stream open functions.
Remove some of the playlist handling code. It looks like this was
needed only for loading linked mov files with demux_mov (which was
removed long ago).
Delete a minor bit of dead network-related code from stream.c as well.
The main problem is that this m_struct stuff uses pointers for offsets
(why...), so we mangle it by intptr_t. This stuff really should use ints
(or in theory ptrdiff_t) for offsets, but changing it would be too much
effort, and hopefully this m_struct stuff will go away and replaced by
the common option parser mechanism instead.
Shuts up warnings on Windows.
Patch suggested by jon_y and rossy on IRC.
Before this commit, we tried to play along with libavformat and tried
to pretend that attached pictures are video streams with a single
frame, and that the frame magically appeared at the seek position when
seeking. The playback core would then switch to a mode where the video
has ended, and the "remaining" audio is played.
This didn't work very well:
- we needed a hack in demux.c, because we tried to read more packets in
order to find the "next" video frame (libavformat doesn't tell us if
a stream has ended)
- switching the video stream didn't work, because we can't tell
libavformat to send the packet again
- seeking and resuming after was hacky (for some reason libavformat sets
the returned packet's PTS to that of the previously returned audio
packet in generic code not related to attached pictures, and this
happened to work)
- if the user did something stupid and e.g. inserted a deinterlacer by
default, a picture was never displayed, only an inactive VO window)
- same when using a command that reconfigured the VO (like switching
aspect or video filters)
- hr-seek didn't work
For this reason, handle attached pictures as separate case with a
separate video decoding function, which doesn't read packets. Also,
do not synchronize audio to video start in this case.
The code touched by this commit makes sure that DVD subtitle tracks
known by libdvdread but not known by demux_lavf can be selected and
displayed properly. These subtitle tracks have the first packet
some time late in the packet stream, so that libavformat won't
immediately recognize them, and will add the track as soon as the
first packet is seen during normal demuxing.
demux_mpg used to handle this elegantly: you just set the MPEG ID of
the stream you wanted. demux_lavf couldn't do this, so it was emulated
with a DEMUXER_CTRL. This commit changes it so that new streams are
selected by default (if autoselect is enabled), and the playloop
simply can take appropriate action before the lower layer throws away
the first packet.
This also changes the demux_lavf behavior that subtitle packets are
always demuxed, even if not needed. (They were immediately thrown away,
so there was no advantage to this.)
Further, this adds the ability to demux.c to deal with demuxing more
than one stream of a kind at once. (Though currently it's not useful.)
Generally remove all accesses to demux_stream from all the code, except
inside of demux.c. Make it completely private to demux.c.
This simplifies the code because it removes an extra concept. In demux.c
it is reduced to a simple packet queue. There were other uses of
demux_stream, but they were removed or are removed with this commit.
Remove the extra "ds" argument to demux fill_buffer callback. It was
used by demux_avi and the TV pseudo-demuxer only.
Remove usage of d_video->last_pts from the no-correct-pts code. This
field contains the last PTS retrieved after a packet that is not NOPTS.
We can easily get this value manually because we read the packets
ourselves. Reuse sh_video->last_pts to store the packet PTS values. It
was used only by the correct-pts code before, and like d_video->last_pts,
it is reset on seek. The behavior should be exactly the same.
This is not directly related to the handling of format changes itself,
but playing audio normally after the change. This was broken: the output
byte rate was not recalculated, so audio-video sync was simply broken.
Fix this by calculating the byte rate on the fly, instead of storing it
in sh_audio.
Format changes are relatively common (switches between stereo and 5.1
in TV recordings), so this fixes a somewhat critical bug.
Removing this code doesn't change anything. All remaining audio decoders
are well-behaved enough to not overwrite sh_audio->pts if they don't
know the PTS. And if they don't know the PTS, the d_audio->last_pts
field can't contain any usable value either, because both fields contain
theame value: the last known valid PTS found in an audio packet.
As the comment n the removed code says, this was once needed for
something subtitle related. This code has been cleaned up long ago,
so at least the original reason for it is gone.
Partial packet reads were needed because the video/audio parsers were
working on top of them. So it could happen that a parser read a part of
a packet, and returned that to the decoder. With libavformat/libavcodec,
packets are already parsed, and everything is much simpler.
Most of the simplifications in ad_spdif could have been done earlier.
Remove some other stuff as well, like the questionable slave mode start
time reporting (could be replaced by proper code, but we don't bother).
Remove the unused skip_audio_frame() functionality as well (it was used
by old demuxers). Some functions become private to demux.c, like
demux_fill_buffer(). Introduce new packet read functions, which have
simpler semantics. Packets returned from them are owned by the caller,
and all packets in the demux.c packet queue are considered unread.
Remove special code that dropped subtitle packets with size 0. This
used to be needed because it caused special cases in the old code.
Add this option, which lets users set the cache size without forcing it
even when playing from the local filesystem.
Also document the default value explicitly.
The Matroska linked segments case is slightly simplified: they can
never come from network (mostly because it'd be insane, and we can't
even list files from network sources), so the cache will never be
enabled automatically.
This code used to be part of the demux_mpg and vobsub specific code
path. Then (just recently) the different code paths for subtitles were
merged, so this code became active even for demux_lavf and demux_mkv.
As far as I can tell, this code won't help much, and at least sd_lavc
(which will be used for DVD subs and other potentially weird things) can
deal with NOPTS values.
Remove the special handling for mng/mkv. These don't profit at all from
no-correct-pts mode, and even removing the mkv specific code makes mkv
work better (wow!).
Don't check for (int)fps == 1000. I don't know where this value comes
from. Maybe it was once a special value which triggered certain
behavior, but the code for that must have gone away. The only way to
trigger this value would be by coincidence if two frames are 1 ms apart.
Otherwise, the behavior should be exactly the same, except for some
removed messages.
This was missing from the previous commit. It worked by luck, because
the sub-commands weren't freed either (as long as the original command
was around), but this is proper.
Also, set the original string for command lists (needed for input-test
only).
The demux_open as well as demux_open_withparams calls don't use the
stream selection parameters anymore, so remove them everywhere.
Completes the previous commit.
These separate arrays were used by the old demuxers and are not needed
anymore. We can simplify track switching as well.
One interesting thing is that stream/tv.c (which is a demuxer) won't
respect --no-audio anymore. It will probably work as expected, but it
will still open an audio device etc. - this is because track selection
is now always done with the runtime track switching mechanism. Maybe
the TV code could be updated to do proper runtime switching, but I
can't test this stuff.
Delete demux_avi, demux_asf, demux_mpg, demux_ts. libavformat does
better than them (except in rare corner cases), and the demuxers have
a bad influence on the rest of the code. Often they don't output
proper packets, and require additional audio and video parsing. Most
work only in --no-correct-pts mode.
Remove them to facilitate further cleanups.
This commit removes the "old" networking code in favor of libavformat's
code.
The code was still used for mp_http, udp, ftp, cddb. http has been
mapped to libavformat's http support since approximately 6 months ago.
udp and ftp have support in ffmpeg (though ftp was added only last
month). cddb support is removed with this commit - it's probably not
important and rarely used if at all, so we don't care about it.
The entries were always added after the insertion point - but that
means the entries are appended in reverse order. So update the
insertion point on each entry.
Regression introduced by commit 5f664d7.
Update Cocoa parts to remove usage of the mp_fifo internal API to send events
to the core and use the input context directly. This is to follow commits the
work in commits 70a8079c and d603e73c.
For some reason mp_fifo specifically handled double clicks, and other
than that was a pointless wrapper around input.c functionality.
Move the double click handling into input.c, and get rid of mp_fifo. Add
some compatibility wrappers, because so much VO code uses these
functions. Where struct mp_fifo is still used it's just a casted
struct input_ctx.
STREAM_CTRL_GET_METADATA will be used to poll for streamcast metadata.
Also add DEMUXER_CTRL_UPDATE_INFO, which could in theory be used by
demux_lavf.c. (Unfortunately, libavformat is too crappy to read metadata
mid-stream for mp3 or ogg, so we don't implement it.)
Making key up events implicit was sort-of a nice idea, but it's too
tricky and unreliable and makes the key lookup code (interpret_keys())
hard to reason about. See e.g. previous commit for subtle bugs and
issues this caused.
Make key-up events explicit instead. Add key up events to all VOs.
Any time MP_KEY_STATE_DOWN is used, the matching key up event must
use MP_KEY_STATE_UP.
Rewrite the key lookup code. It should be simpler and more robust now.
(Even though the LOC increases, because the new code is less "compact".)
Wayland is the only backend that actually sends per-key key up events
(the X11 one just sends MP_INPUT_RELEASE_ALL for simplicity). Handling
was broken with Wayland, and each key event was interpreted twice, once
on key down and once on key up.
This commit should fix it.
Before this commit, only mouse events with both down and up events
were processed. This caused a regression with ignoring mouse wheel
events in cocoa, because these don't distinguish between up and down.
Regression caused by 5b38a52.
This means the direct libass usage can be removed from command.c, and no
weird hacks for retrieving the ASS_Track are needed.
Also fix a bug when using this feature with ordered chapters.
This was changed as part of commit b44202b as an intended
simplification, but it's actually nicer to have the subtitles
update immediately even if paused.
Instead mark individual key bindings as builtin.
Not sure whether this is conceptually simpler or more complicated.
For one, it requires the annoying remove_binds() function to wipe
existing bindings instead of just killing the section, on the other
hand it gets rid of almost all special handling of builtin vs. normal
sections.
Also, implement mouse leave events for X11. But evne on other
platforms, these events will be generated if mouse crosses a section's
mouse area boundaries within the mpv window.
Before this commit, mouse movement events emitted a special command
("set_mouse_pos"), which was specially handled in command.c. This was
once special-cased to the dvdnav and menu code, and did nothing after
libmenu and dvdnav were removed.
Change it so that mouse movement triggers a pseudo-key ("MOUSE_MOVE"),
which then can be bound to an arbitrary command. The mouse position is
now managed in input.c. A command which actually needs the mouse
position can use either mp_input_get_mouse_pos() or mp_get_osd_mouse_pos()
to query it. The former returns raw window-space coordinates, while the
latter returns coordinates transformed to OSD- space. (Both are the same
for most VOs, except vo_xv and vo_x11, which can't render OSD in
window-space. These require extra code for mapping mouse position.)
As of this commit, there is still nothing that uses mouse movement, so
MOUSE_MOVE is mapped to "ignore" to silence warnings when moving the
mouse (much like MOUSE_BTN0).
Extend the concept of input sections. Allow multiple sections to be
active at once, and organize them as stack. Bindings from the top of
the stack are preferred to lower ones.
Each section has a mouse input section associated, inside which mouse
events are associated with the bindings. If the mouse pointer is
outside of a section's mouse area, mouse events will be dispatched to
an input section lower on the stack of active sections. This is intended
for scripting, which is to be added later. Two scripts could occupy
different areas of the screen without conflicting with each other. (If
it turns out that this mechanism is useless, we'll just remove it
again.)
percent-pos was an integer (0-100). Sometimes higher precision is
wanted, but the property is this way because fractional parts would
look silly with normal OSD usage. As a compromise, make percent-pos
double (i.e. includes fractional parts), but print it as integer.
So ${percent-pos} is like an integer, but not ${=percent-pos}.
This adds support for libquvi 0.9.x, and these features:
- start time (part of youtube URL)
- youtube subtitles
- alternative source switching ('l' and 'L' keys)
- youtube playlists
Note that libquvi 0.9 is still in development. Although this seems to
be API stable now, it looks like there will be a 1.0 release, which is
supposed to be the next stable release and the actual successor of
libquvi 0.4.x.
Should we actually get into trouble for unproper handling of
frame-based subtitle formats, this might be the simplest way to
work this around. Also is a bit more intuitive than -subfps, which
might use an unknown, misdetected, or non-sense video FPS.
Still pretty silly, though.
This code was once part of subreader.c, then traveled to libass, and now
made its way back to the fork of the fork of the original code, MPlayer.
It works pretty much the same as subreader.c, except that we have to
concatenate some packets to do auto-detection. This is rather annoying,
but for all we know the actual source file could be a binary format.
Unlike subreader.c, the iconv context is reopened on each packet. This
is simpler, and with respect to multibyte encodings, more robust.
Reopening is probably not a very fast, but I suspect subtitle charset
conversion is not an operation that happens often or has to be fast.
Also, this auto-detection is disabled for microdvd - this is the only
format we know that has binary data in its packets, but is actually
decoded to text. FFmpeg doesn't really allow us to solve this properly,
because a) the input packets can be binary, and b) the output will be
checked whether it's UTF-8, and if it's not, the output is thrown away
and an error message is printed. We could just recode the decoded
subtitles before sd_ass if it weren't for that.
demux_libass.c allows us to make subtitle format detection part of the
normal file loading process. libass has no probe function, but trying to
load the start of a file (the first 4 KB) is good enough. Hope that
libass can even handle random binary input gracefully without printing
stupid log messages, and that the libass parser doesn't accept too many
non-ASS files as input.
This doesn't handle the -subcp option correctly yet. This will be fixed
later.
The default correct-pts mode depended on which demuxer was opened last.
Often this is the subtitle demuxer. The correct-pts mode should be
decided on the demuxer for video instead.
subreader.c (before this commit renamed to demux_subreader.c) was
special cased to the -sub option. The plan is using the normal demuxer
codepath for all subtitle formats (so we can prefer libavformat demuxers
for most formats).
There are some subtle changes. The probe size is restricted to 32 KB
(instead of unlimitted + giving up after 100 lines of input). For
formats like MicroDVD, the video FPS isn't used anymore, because it's
not available on the subtitle demuxer level. Instead, hardcode it to
23.976 FPS (libavformat seems to do the same). The user can probably
still use -sub-fps to fix the timing. Checking the file extension for
".utf"/".utf8"/".utf-8" is simply removed (seems worthless, was in the
way, and I've never seen this anywhere).
This fixes the -subfps option (which unfortunately is still useful),
and fixes minor annoying timing errors (which unfortunately still
happen).
Note that none of these affect ASS or image subtitles. ASS is specially
handled: libass loads subtitles as ASS_Track. There are no actual
packets passed around, and sd_ass just uses the ASS_Track.
Disable the --sub-no-text-pp option. It's misleading now and always was
completely useless.
If a subtitle is external, read it completely and add all subtitle
events in advance when the subtitle track is selected. This is done
for text subtitles only. (Note that subreader.c and subtitles loaded
with libass are different and don't have anything to do with this
commit.)
Seems like a completely unnecessary complication. Instead, always add a
1 byte padding (could be extended if a caller needs it), and clear it.
Also add some documentation. There was some, but it was outdated and
incomplete.
Don't print the ffmpeg context pointer as %p. This is usually useless
and confusing. Prefix all messages with "ffmpeg" to make clear that
ffmpeg is printing these messages, and not us.
If libavcodec is from Libav, use "libav" as prefix instead. (In theory,
FFmpeg/Libav libraries could be mixed, but I don't think that is
actually possible in practice.)
Basically rewrite all the code supporting the cache (i.e. anything other
than the ringbuffer logic). The underlying design is untouched.
Note that the old cache2.c (on which this code is based) already had a
threading implementation. This was mostly unused on Linux, and had some
problems, such as using shared volatile variables for communication and
uninterruptible timeouts, instead of using locks for synchronization.
This commit does use proper locking, while still retaining the way the
old cache worked. It's basically a big refactor.
Simplify the code too. Since we don't need to copy stream ctrl args
anymore (we're always guaranteed a shared address space now), lots of
annoying code just goes away. Likewise, we don't need to care about
sector sizes. The cache uses the high-level stream API to read from
other streams, and sector sizes are handled transparently.
Before this commit, the cache was franken-hacked on top of the stream
API. You had to use special functions (like cache_stream_fill_buffer()
instead of stream_fill_buffer()), which would access the stream in a
cached manner.
The whole idea about the previous design was that the cache runs in a
thread or in a forked process, while the cache awa functions made sure
the stream instance looked consistent to the user. If you used the
normal functions instead of the special ones while the cache was
running, you were out of luck.
Make it a bit more reasonable by turning the cache into a stream on its
own. This makes it behave exactly like a normal stream. The stream
callbacks call into the original (uncached) stream to do work. No
special cache functions or redirections are needed. The only different
thing about cache streams is that they are created by special functions,
instead of being part of the auto_open_streams[] array.
To make things simpler, remove the threading implementation, which was
messed into the code. The threading code could perhaps be kept, but I
don't really want to have to worry about this special case. A proper
threaded implementation will be added later.
Remove the cache enabling code from stream_radio.c. Since enabling the
cache involves replacing the old stream with a new one, the code as-is
can't be kept. It would be easily possible to enable the cache by
requesting a cache size (which is also much simpler). But nobody uses
stream_radio.c and I can't even test this thing, and the cache is
probably not really important for it either.
The core didn't use these fields, and use of them was inconsistent
accross AOs. Some didn't use them at all. Some only set them; the values
were completely unused by the core. Some made full use of them.
Remove these fields. In places where they are still needed, make them
private AO state.
Remove the --abs option. It set the buffer size for ao_oss and ao_dsound
(being ignored by all other AOs), and was already marked as obsolete. If
it turns out that it's still needed for ao_oss or ao_dsound, their
default buffer sizes could be adjusted, and if even that doesn't help,
AO suboptions could be added in these cases.
Currently every single AO was implementing it's own ringbuffer, many times
with slightly different semantics. This is an attempt to fix the problem.
I stole some good ideas from ao_portaudio's ringbuffer and went from there.
The main difference is this one stores wpos and rpos which are absolute
positions in an "infinite" buffer. To find the actual position for writing /
reading just apply modulo size.
The producer only modifies wpos while the consumer only modifies rpos. This
makes it pretty easy to reason about and make the operations thread safe by
using barriers (thread safety is guaranteed only in the Single-Producer/Single-
Consumer case).
Also adapted ao_coreaudio to use this ringbuffer.
This allows having properties like time-pos in the window title update
properly. There is a danger of this causing significant CPU usage,
depending on the properties used and the window manager.
