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.
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.
Instead of implicitly changing the window title on config(), do it as
part of the new VOCTRL.
At first I wanted to make all VOs use the VOCTRL argument directly, but
on a second thought it appears vo_get_window_title() is much more useful
for some (namely, if the window is created lazily on first config()).
Not all VOs are changed. Wayland and OSX have to follow.
Use the recently introduced screensaver VOCTRLs to control the
screensaver in the X11 backend. This means the behavior when paused
changes: the old code always kept the screensaver disabled, but now the
screensaver is reenabled on pausing.
Rename the --stop-xscreensaver option to --stop-screensaver and make it
more generic. Now it affects all backends that respond to the
screensaver VOCTRLs.
This is slightly better because VOCTRL_RESUME/VOCTRL_PAUSE are usually
needed by VOs to know whether video is actually being played (for
whatever reason), and they wouldn't be passed to the backend's VOCTRL
handler, like vo_x11_control().
Also try to make sure that these flags (both pause state and screensaver
state) are set consistently in some corner cases. For example, it seems
enabling video in the middle of playing a file while the player is
paused would not set the paused flag.
If codec initialization fails, destroy the VO instead of keeping it
around to make sure the state is consistent.
Framestepping is implemented by unpausing the player for the duration of
a frame. Remove the special handling of VOCTRL_PAUSE/RESUME in these
cases. It was most likely needed because these VOCTRLs used to be
important for screen redrawing (blatant guess), which is now handled
completely differently. The only potentially bad side-effect is that the
screensaver will be disabled/reenabled for the duration of one frame.
Raw MPEG streams can contain PTS discontinuities. While the playback
core has obvious code for handling PTS going backward, PTS going
forward was as far as I can see not handled.
This can be an issue with DVD playback. This wasn't caught earlier,
because DVD playback was just recently switched to demux_lavf, which
implies -no-correct-pts mode. This mode doesn't use PTS in the same way
as the normal playback mode, and as such was too primitive to be
affected by this issue.
Use the following heuristic to handle PTS forward jumps: if the PTS
difference between two frames is higher than 10 seconds, consider it a
reset. (Also, use MSGL_WARN for the PTS discontinuity warnings.)
In this particular case, the MPEG stream was going from pts=304510857
to pts=8589959849 according to ffprobe (raw timestamps), which seems a
bit strange.
Note that this heuristic breaks if the source video has unusually high
frame times. For example Rooster_Teeth_Podcast_191.m4a, an audio file
with a slide show encoded as MJPEG video track.
Some code in mplayer.c did stuff like accessing (dvd_priv_t *)st->priv.
Do this indirectly by introducing STREAM_CTRL_GET_DVD_INFO. This is
extremely specific to DVD, so it's not worth abstracting this further.
This is a preparation for turning the cache into an actual stream, which
simply wraps the cached stream. There are other streams which are
accessed in the way DVD was, at least TV/radio/DVB. We assume these
can't be used with the cache. The code doesn't look thread-safe or fork
aware.
This branch heavily refactors the subtitle code (both loading and
rendering), and adds support for a few new formats through FFmpeg.
We don't remove any of the old code yet. There are still some subtleties
related to subreader.c to be resolved: code page detection & conversion,
timing post-processing, UTF-16 subtitle support, support for the -subfps
option. Also, SRT reading and loading ASS via libass should be turned
into proper demuxers. (SRT is needed because Libav's is gravely broken,
and we want ASS loading via libass to cover full libass format support.
Both should be demuxers which are probed _before_ libavformat, so that
all subtitles can be loaded through the demuxer infrastructure, and
libavformat subtitles don't need to be treated in a special way.)
Until now, this happened only when the -no-ass option was used. This
difference in behavior doesn't make much sense, so change it so that
whether -no-ass is used or not doesn't matter. (-no-ass enables the OSD
subtitle renderer, which has the terminal fallback, while the normal
path is video only.)
the changes in set_osd_subtitle() and reinit_video_chain() are for
resetting the state correctly when switching between video/no-video.
Before this, subtitle packets were returned as data ptr/len pairs, and
mplayer.c got the rest (pts and duration) directly from the demuxer
data structures. Then mplayer.c reassembled the packet data structure
again.
Pass packets directly instead. The mplayer.c side stays a bit awkward,
because the (now by default unused) DVD path keeps getting in the way.
In demux.c there's lots of weird stuff (3 functions that read packets,
really?), but we want to keep the code equivalent for now to avoid
hitting weird issues and corner cases.
The -no-ass option used to disable all use of libass completely. This
doesn't work this way anymore, and the text subtitle path has an
inherent dependency on libass. Currently -no-ass does 3 things:
1. Strip tags and formatting on display, and use a separate renderer for
the result. (Which might be the terminal, or libass via OSD code.)
2. Not loading attached fonts from Matroska files.
3. Use subreader.c instead of libass for reading .ass files.
1. and 2. are ok and what the user (probably wants), but 3. doesn't
really make sense anymore. subreader.c reads .ass files just fine, but
then does some strange things to them (something about coalescing and
re-adding newlines?), leading to even more broken display with -no-ass.
Instead of fighting with subreader.c, just use libass as loader.
This means subassconvert.c is split in sd_srt.c and sd_microdvd.c. Now
this code is involved in the sub conversion chain like sd_movtext is.
The invocation of the converter in sd_ass.c is removed.
This requires some other changes to make the new sub converter code work
with loading external subtitles. Until now, subtitles loaded via
subreader.c was assumed to be in plaintext, or for some formats, in ASS
(except in -no-ass mode). Then these were added to an ASS_Track. Change
this so that subtitles are always in their original format (as far as
decoders/converters for them are available), and turn every sub event
read by subreader.c as packet to the dec_sub.c subtitle chain.
This removes differences between external/demuxed and -ass/-no-ass code
paths further.
After killing the non functional AR support in c8fd9e5 I got much complaints so
this adds AR support back in (and it works). I am using the HIDRemote class by
Felix Schwarz and that part of the code is under the BSD license. I slightly
modified it replacing [NSApplication sharedApplication] with NSApp. The code
of the class is quite complex (probably because it had to deal with all the
edge cases with IOKit) but it works nicely as a black box.
In a later commit I'll remove the deprecation warnings caused by HIDRemote's
usage of Gestalt.
Check out `etc/input.conf` for the default bindings.
Apple Remote functionality is automatically compiled in when cocoa is enabled.
It can be disabled at runtime with the `--no-ar` option.
On OSX with Cocoa enabled keyDown events are now handled with
addLocalMonitorForEventsMatchingMask:handler:. This allows to respond to
events even when there is no VO initialized but the GUI is focused.
Add a basic infrastructure for subtitle converters. These converters
work sort-of like decoders, except that they produce packets instead
of subtitle bitmaps. They are put in front of actual decoders.
Start with sd_movtext. 4 lines of code are blown up to a 55 lines file,
but fortunately this is not going to be that bad for the following
converters.
Make the sub decoder stuff independent from sh_sub (except for
initialization of course). Sub decoders now access a struct sd only,
instead of getting access to sh_sub. The glue code in dec_sub.c is
similarily independent from osd.
Some simplifications are made. For example, the switch_id stuff is
unneeded: the frontend code just has to make sure to call osd_changed()
any time subtitles are switched.
This is also preparation for introducing subtitle converters. It's much
cleaner to completely separate demuxer header/renderer glue/decoders
for this purpose, especially since sub converters might completely
change how demuxer headers have to be interpreted.
Also pass data as demux_packets. Currently, this doesn't help much, but
libavcodec converters might need scary stuff like packet side data, so
it's perhaps better to go with passing packets.
Subtitle files are opened in mplayer.c, not using the demuxer
infrastructure in general. Pretend that this is not the case (outside of
the loading code) by opening a pseudo demuxer that does nothing. One
advantage is that the initialization code is now the same, and there's
no confusion about what the difference between track->stream,
track->sh_sub and mpctx->sh_sub is supposed to be.
This is a bit stupid, and it would be much better if there were proper
subtitle demuxers (there are many in recent FFmpeg, but not Libav). So
for now this is just a transition to a more proper architecture. Look
at demux_sub like an artifical limb: it's ugly, but don't hate it - it
helps you to get on with your life.
This unifies the subtitle rendering path. Now all subtitle rendering
goes through sd_ass.c/sd_lavc.c/sd_spu.c.
Before that commit, the spudec.h functions were used directly in
mplayer.c, which introduced many special cases. Add sd_spu.c, which is
just a small wrapper connecting the new subtitle render API with the
dusty old vobsub decoder in spudec.c.
One detail that changes is that we always pass the palette as extra
data, instead of passing the libdvdread palette as pointer to spudec
directly. This is a bit roundabout, but actually makes the code simpler
and more elegant: the difference between DVD and non-DVD dvdsubs is
reduced.
Ideally, we would just delete spudec.c and use libavcodec's DVD sub
decoder. However, DVD playback with demux_mpg produces packets
incompatible to lavc. There are incompatibilities the other way around
as well: packets from libavformat's vobsub demuxer are incompatible to
spudec.c. So we define a new subtitle codec name for demux_mpg subs,
"dvd_subtitle_mpg", which only sd_spu can decode.
There is actually code in spudec.c to "assemble" fragments into complete
packets, but using the whole spudec.c is easier than trying to move this
code into demux_mpg to fix subtitle packets.
As additional complication, Libav 9.x can't decode DVD subs correctly,
so use sd_spu in that case as well.
It appears demux_mpg doesn't output timestamps for subtitles. The vobsub
code handled this by doing its own PTS calculations. This code is absent
from the normal subtitle decoder path. Copy this code into the normal
path, so that we can unify the subtitle decoder paths in a later commit.
Decoding subtitles with sd_lavc when playing DVD with demux_mpg still
doesn't work.
The -no-ass switch used to disable any use of libass for text subtitles.
This is not really the case anymore, because libass is now always
involved when rendering text. The only remaining use of -no-ass is
disabling styling or showing subtitles on the terminal. On the other
hand, the old subtitle rendering path is a big reason why the subtitle
code is still a big mess with an awful number of obscure special cases.
In order to simplify it, remove the old subtitle rendering code, and
always go through sd_ass.c. Basically, we use ASS_Track as central data
structure for storing text subtitles instead of struct sub_data. This
also makes libass mandatory for all text subs, even if they are printed
to the terminal in -no-video mode. (We could add something like sd_text
to avoid this, but it's not worth the trouble.)
struct sub_data and subreader.c are still around, even its ASS/SSA
reader. But struct sub_data is freed right after converting it to
ASS_Track. The internal ASS reader actually can handle some obscure
cases libass can't, like files encoded in UTF-16.
The core deselected all streams on initialization, and then selected the
streams it actually wanted. This was no problem for
demux_mkv/demux_lavf, but old demuxers (like demux_asf) could lose some
packets. The problem is that these demuxers can buffer some data on
initialization, which then is flushed on track switching. Fix this by
explicitly avoiding deselecting a wanted stream.
Playing something with "mpv f1.mkv f2.mkv --gapless-audio --volume=20"
caused the volume to be reset when playing a new file. Normally, the
volume should not be reset (unless explicitly requested with per-file
options), and without either --gapless-audio or --volume it works as
expected.
The underlying problem is that volume was saved only when the AO was
uninitialized, and also the volume was always set when starting a file.
Fix this by saving the volume when playback ends, and when the audio
is reinitialized. To make sure the volume is never restored twice or
saved in the wrong situation, introduce INITIALIZED_VOL.
Also note that this volume saving and restoring only happens if the
--volume option is used. mixer.c does its own bookkeeping of volume.
The main reason for this is that the volume option could be reset by
per-file options (see manpage), and mixer.c doesn't know anything
about this stuff. This is probably dumb, and maybe some things could
be simplified. But for now this will work.
In commit 0e07189, I made the status line always print a newline,
instead of cutting the output at 80 columns (or if stderr is a terminal,
whatever width the terminal reports). This is better in the case the
output goes into a log file or a pipe.
This caused problems for people who want to pipe raw video to mpv, so
change it again. (Not sure why they won't use FIFOs instead.)
Now output untrimmed lines if the slave mode flag is set, which makes
sense to do, too. The current slave mode is still on life support,
though.
GetTimer() is generally replaced with mp_time_us(). Both calls return
microseconds, but the latter uses int64_t, us defined to never wrap,
and never returns 0 or negative values.
GetTimerMS() has no direct replacement. Instead the other functions are
used.
For some code, switch to mp_time_sec(), which returns the time as double
float value in seconds. The returned time is offset to program start
time, so there is enough precision left to deliver microsecond
resolution for at least 100 years. Unless it's casted to a float
(or the CPU reduces precision), which is why we still use mp_time_us()
out of paranoia in places where precision is clearly needed.
Always switch to the correct time. The whole point of the new timer
calls is that they don't wrap, and storing microseconds in unsigned int
variables would negate this.
In some cases, remove wrap-around handling for time values.
This was used by some VOs to do timing of cursor autohiding, but we
recently moved that out of the VOs. Even though this mechanism might
be a good idea and could be needed again in future (but for what?),
it's unused now. So better just get rid of it.
Make OS specific timer code export a mp_raw_time_us() function, and
add generic implementations of GetTimer()/GetTimerMS() using this
function. New mpv code is supposed to call mp_time_us() in situations
where precision is absolutely needed, or mp_time_s() otherwise.
Make it so that mp_time_us() will return a value near program start.
We don't set it to 0 though to avoid confusion with relative vs.
absolute time. Instead, pick an arbitrary offset.
Move the test program in timer-darwin.c to timer.c, and modify it to
work with the generic timer functions.
In the long run this should be done differently. ID_... output sucks.
This commit will be reverted as soon as I have a good idea how this
should be done properly.
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().
The main() function is special, and omitting the return statement would
make it always return 0. And also, mpv_main() actually never returns, it
calls exit() through exit_player() instead. But change it anyway,
because it looks misleading.
Apparently useful for dumping DVD. Could also be used to rip streams
with libquvi and such, but for that there are better tools. Actually
I doubt there aren't better tools to dump DVDs, but whatever, this was
a feature request, so I don't need a good reason.
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.
This actually breaks audio for 5/6/8 channels. There's no reordering
done yet. The actual reordering will be done inside of af_lavrresample
and has to be made part of the format negotiation.
This commit is a followup on the previous one and uses a solution I like more
since it totally decouples the Cocoa code from mpv's core and tries to emulate
a generic Cocoa application's lifecycle as much as possible without fighting
the framework.
mpv's main is executed in a pthread while the main thread runs the native cocoa
event loop.
All of the thread safety is mainly accomplished with additional logic in
cocoa_common as to not increase complexity on the crossplatform parts of the
code.
Schedule mpv's playloop as a high frequency timer inside the main Cocoa event
loop. This has the benefit to allow accessing menus as well as resizing the
window without the playback being blocked and allows to remove countless hacks
from the code that involved manually pumping the event loop as well simulating
manually some of the Cocoa default behaviours.
A huge improvement consists in removing NSApplicationLoad. This is a C function
defined in the Cocoa header and implements a minimal OSX application under ther
hood so that you can use the Cocoa GUI toolkit from C/C++ without having to
respect the Cocoa standards in terms of application initialization. This was
bad because the behaviour implemented by NSApplicationLoad was hard to customize
and had several gotchas especially in the menu department.
mpv was changed to be just a nib-less application. All the Cocoa part is still
generated in code but the event handling is now not dissimilar to what is
present in a stock Mac application.
As a part of reviewing the initialization process, I also removed all of
`osdep/macosx_finder_args`. The useful parts of the code were moved to
`osdep/macosx_appication` which has the broaded responsibility of managing the
full lifecycle of the Cocoa application. By consequence the
`--enable-macosx-finder` configure switch was killed as well, as this feature
is always enabled.
Another change the users will notice is that when using a bundle the `--quiet`
option will be inserted much earlier in the initializaion process. This results
in mpv not spamming mpv.log anymore with all the initialization outputs.
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.
The frontend doesn't use this.
Also use double for returning the chapter times. Everything uses double
for times, and there's no reason to use float here.
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.
Also, mark demuxer as not capable if DVD playback is done. The problem
with DVD is that playback time (stream_pts) is not reported frame-exact,
and the time is a "guess" at best.
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.
There were two problems.
First, frames past the end of the current segment were added to the
index, which messed up backstepping. Check for the endpts before
added a frame to the index.
Second, it wasn't possible to step over segments which change the file.
Changing a file causes decoder reinitialization, which (rightfully)
is treated as discontinuity (and vo_pts_history_seek_ts was changed).
Add some extra code to pretend that a segment-switching seek/reinit
does not introduce discontinuities.
There's still a weird corner case: sometimes, you can frame step forward
on the last frame of a segment without reaching the next segment
immediately. This is because the playloop switches into audio-only mode.
The segment is switched when both audio and video have ended, so the
frame stepping will play random sized chunks of audio until the segment
will be switched. This gives the impression that backstepping doesn't
work perfectly, even though it's the other way around and frame stepping
behaves weird. This is a consequence of wanting to make frame stepping
work with audio, and is not really a bug.
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.
It's not sure if there's anything that could trigger this accidentally.
Normally this can't happen, because hrseek ends always if the PTS is
large enough, the same condition which disables framedrop. Seeking
resets hrseek framedrop anyway.
On the other hand, this change makes the code easier to understand,
and might be more robust against weird corner cases.
Before this commit, it was more or less random which subtitle was
preferred if there was both an auto-loaded external subtitle, and a
subtitle loaded via -sub or -subfile. -sub subtitles happened to be
preferred over auto-loaded subs, while -subfile didn't. Fix the -subfile
case, and make the behavior consistent by making the selection behavior
explicit.
Get rid of the 1-char subtitle type field. Use sh_stream->codec instead
just like audio and video do. Use codec names as defined by libavcodec
for simplicity, even if they're somewhat verbose and annoying.
Note that ffmpeg might switch to "ass" as codec name for ASS, so we
don't bother with the current silly "ssa" name.
With Matroska ordered chapters, the main file (i.e. the file you're
playing) can be empty, while all video/audio data is in linked files.
Some files don't even contain the track list, only chapter information.
mpv refused to play these, because normally, the main file dictates the
track layout.
Fix this by using the first segment for track data if no part of the
timeline is sourced from the main file.
Switch the internal channel order to libavcodec's. If the channel number
mismatches at some point, use libavresample for up- or downmixing.
Remove the old af_pan automatic downmixing.
The libavcodec channel order should be equivalent to WAVEFORMATEX order,
at least nowadays. reorder_ch.h assumes that WAVEFORMATEX and libavcodec
might be different, but all defined channels have the same mappings.
Remove the downmixing with af_pan as well as the channel conversion with
af_channels from af.c, and prefer af_lavrresample for this. The
automatic downmixing behavior should be the same as before (if the
--channels option is set to 2, which is the default, the audio output
is forced to 2 channels, and libavresample does all downmixing).
Note that mpv still can't do channel layouts. It will pick the default
channel layout according to the channel count. This will be fixed later
by passing down the channel layout as well.
af_hrtf depends on the order of the input channels, so reorder to ALSA
(for which this code was written). This is better than changing the
filter code, which is more risky.
ao_pulse can accept waveext order directly, so set that as channel
mapping.
This option can be used to selectively reset settings when playing the
next file in the playlist (i.e. restore mplayer and mplayer2 behavior).
Might remove this option again should it turn out that nobody uses it.
Consider:
mpv --volume 10 file1.mkv file2.mkv
Before this commit, the volume was reset to 10 when playing file2.mkv.
This was inconsistent to most other options. E.g. --brightness is a
rather similar case.
In general, settings should never be reset when playing the next file,
unless the option was explicitly marked file-local. This commit
corrects the behavior of the --volume and --mute options.
File local --volume still works as expected:
mpv --{ --volume 10 file1.mkv file2.mkv --}
This sets the volume always to 10 on playback start.
Move the m_config_leave_file_local() call down so that the mixer code
in uninit_player() can set the option volume and mute variables without
overwriting the global option values.
Another subtle issue is that we don't want to set volume if there's no
need to, which is why the user_set_volume/mute fields are introduced.
This is important because setting the volume might change the system
volume depending on other options.
Makes sure that seeking to a given time position shows the subtitle at
that position. This can fail if the subtitle packet is not close enough
to the seek target. Always enabled for hr-seeks, and can be manually
enabled for normal seeks with --mkv-subtitle-preroll.
This helps displaying subtitles correctly with ordered chapters. When
switching ordered chapter segments, a seek is performed. If the subtitle
is timed slightly before the start of the segment, it normally won't be
demuxed. This is a problem with all seeks, but in this case normal
playback is affected. Since switching segments always uses hr-seeks,
the code added by this commit is always active in this situation.
If no subtitles are selected or the subtitles come from an external
file, the demuxer should behave exactly as before this commit.
The main problem with video PTS was that it wasn't very useful when
playing audio files with cover art. Using the audio time instead was an
obvious solution. Unfortunately, this leads to "inexact" reporting of
the playback time in paused mode, and audio is always ahead by small,
essentially random amounts of time ahead. This is possibly because the
times reported by AOs are not entirely accurate when paused (see commit
9b3bf76).
Switch back to video PTS, and use a simpler way to deal with the cover
art case: if the video has ended, use the audio PTS.
Also see commit f9a259e (and the commits referenced from there).
Trying to step over a segment boundary didn't work, and the video was
stuck at the end of the current chapter. At this point, both video and
audio of the segment has ended, and the segment switching code is going
to call seek() to go to the next segment (the part of the code in
run_playloop that uses end_is_chapter). However, this seek() is not
called if playback is paused, and the framestepping code always paused
before this code is run.
Move the framestepping code below the chapter switching code. The added
restart_playback condition makes sure the code is called only after at
least one video frame has been shown. Also don't reset the framestep
counter after seek. It's not needed, and removing it prevents full
unpausing when stepping over a segment boundary.
This also terminates playback when frame stepping at the end of the
file. The --keep-open option can be used to get the old behavior.
Drawing the bar with vector drawings (instead with characters from the
OSD font) offers more flexibility and looks better. This also adds
chapter marks to the OSD bar, which are visible as small triangles on
the top and bottom inner border of the bar.
Change the default position of the OSD bar below the center of the
screen. This is less annoying than putting the bar directly into the
center of the view, where it obscures the video. The new position is
not quite on the bottom of the screen to avoid collisions with
subtitles.
The old centered position can be forced with ``--osd-bar-align-y=0``.
Also make it possible to change the OSD bar width/height with the new
--osd-bar-w and --osd-bar-h options.
It's possible that the new OSD bar renders much slower than the old
one. There are two reasons for this: 1. the character based bar
allowed libass to cache each character, while the vector drawing forces
it to redraw every time the bar position changes. 2., the bar position
is updated at a much higher granularity (the bar position is passed
along as float instead of as integer in the range 0-100, so the bar
will be updated on every single video frame).
Makes the code a bit simpler to follow, at least in the "modern"
decoding path (update_video_nocorrect_pts() is used with old demuxers,
which don't return proper packets and need further parsing, so this code
looks less simple now).
This way it's possible to retrieve correct information about video, like
actual width/height, which in general are available only after at least
one frame has been sent to the video output, such as dwidth/dheight.
mpv_identify.sh becomes a bit slower, because we let it decode enough
audio and video to fill the audio buffers and to send one frame to the
video output. Also, --playing-msg isn't shown anymore with --frames=0
(could be fixed by special-casing it, should this break any use cases).
Note that in some corner cases, like when the demuxer for some reason
returns lots of audio packets but no video packets at the start, but
video actually starts later, the --playing-msg will still be output
before video starts.
This has the same (useless) definition as frame stepping in audio-only
mode: one frame means one playloop iteration. (It's relatively useless,
because one playloop iteration has a random duration. But it makes
--frames=1 work, which is useful again.)
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.
When paused, --cursor-autohide worked with a precision of 500ms, which
is the main loop's default sleep time when paused. Cursor hiding is
polled in x11_common, and the main loop never called the X11 code at
the right time. Fix this by allowing the VO to set a time when it
should be called next.
Emulate percentage-seeks (SEEK_FACTOR) as normal time-seeks if possible.
This fixes some issues with (let's call it) low quality implementations
of SEEK_FACTOR (e.g. demux_mkv basically interprets this as byte-seek,
and also seeking to 99.9% makes it seek back to the start).
For weird MPEG formats the demuxer level SEEK_FACTOR is still used.
These formats, which can have timestamp resets, are identified by
setting demuxer->ts_resets_possible to true.
Also, have get_current_pos_ratio() follow the same rules, and calculate
the percentage position with the file position if timestamp resets are
possible.
This actually fixes percentage-seeks in .ts files with demux_lavf.c.
This kind of seek is not really used now, but it will be more important
when we add a progress bar.
Note: seeking in chained ogg files is still completely broken. The main
issue is that ffmpeg doesn't provide a sane API for dealing with
timestamp resets, and trying to do byte seeks with ogg confuses demuxer
and decoder (or something like this) and just does random things.
(Tested with two concatenated flac-in-ogg files).
VFCAP_OSD was used to determine at runtime whether the VO supports OSD
rendering. This was mostly unused. vo_direct3d had an option to disable
OSD (was supposed to allow to force auto-insertion of vf_ass, but we
removed that anyway). vo_opengl_old could disable OSD rendering when a
very old OpenGL version was detected, and had an option to explicitly
disable it as well.
Remove VFCAP_OSD from everything (and some associated logic). Now the
vo_driver.draw_osd callback can be set to NULL to indicate missing OSD
support (important so that vo_null etc. don't single-step on OSD
redraw), and if OSD support depends on runtime support, the VO's
draw_osd should just do nothing if OSD is not available.
Also, do not access vo->want_redraw directly. Change the want_redraw
reset logic for this purpose, too. (Probably unneeded, vo_flip_page
resets it already.)
Use floats instead of integers in the range 0-100. Currently, the OSD
is currently made up of 46 elements so no change should be visible, but
rendering of the bar will be changed later to use vector drawings (using
pixel coordinates) instead of glyphs. This commit is for preparation.
The percent position is used for the OSD, the status line, and for the
OSD bar (shown on seeks). By default, the PTS of the last demuxed packet
was used to calculate it. This led to a "jumpy" display when the
percentage value (casted to int) was changing. The reasons for this were
the presence of video frame reordering (packet PTS is not monotonic), or
getting PTS values from different streams (like audio/subs).
Since these rely on PTS values and correct file durations anyway,
simplify it by calculating it with the current playback position in
mplayer.c instead.
"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.
wm4
John Hawthorn
Stefano Pigozzi
Alexander Preisinger