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.
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.)
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.
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.
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.
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.
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.
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().
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.
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.
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.
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).
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.
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.
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).
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.
This was supposed to be fixed in f897138, but there's another corner
case. Basically, set_osd_function() reset the OSD time, which is not
nice at all and breaks the logic of letting OSD elements disappear when
they're not wanted anymore. Fix this by adding a separate timer for
this.
Additionally, make sure the OSD bar is _really_ always updated when
visible. Also, redraw the OSD only if the OSD bar actually changes to
prevent redrawing too often (every vo_osd_changed() will flag that the
OSD should be redrawn, even if nothing changes).
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.
ffmpeg pretends that image attachments (such as contained in ID3v2
metadata) are video streams. It injects the attached pictures as packets
into the packet stream received with av_read_frame().
Add the --audio-display option to allow configuring whether attached
pictures should be displayed. The default behavior doesn't change
(images are displayed).
Identify video streams, that are actually image attachments, with "[P]"
in the terminal output.
Modify the default stream selection such that real video streams are
preferred over attached pictures. (This is just for robustness; I do not
know of any samples where images are added before actual video streams
and could lead to bad default stream selection with the old code.)
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).
When the cache fill status goes below a certain threshold, automatically
pause the player. When the cache is filled again, unpause again.
This is intended to help with streaming from http. It's better to pause
a while, rather than exposing extremely crappy behavior when packet
reads during decoding block the entire player.
In theory, we should try to increase the cache if underruns happen too
often. Unfortunately, changing the cache implementation would be very
hard, because it's insane code (forks, uses shared memory and "volatile"
etc.). So for now, this just reduces the frequency of the stuttering if
the network is absolutely too slow to play the stream in realtime.
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.
The --keep-open option causes mpv not to close the current file.
Instead, it will pause, and allow the user to seek around. When
seeking beyond the end of the file, mpv does a precise seek back to
the previous last known position that produced video output.
In some corner cases, mpv might not be able to produce video output at
all, despite having created a VO. (Possibly when only 1 frame could be
decoded, but the video filter chain queues frames. Then a VO would be
created, without sending an actual video frame to the VO.) In these
cases, the VO window will not redraw, not even OSD.
Based on a patch by coax [1].
[1] http://devel.mplayer2.org/ticket/210#comment:4
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.
