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.
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).
Increase robustness against out of bound chapter numbers. Normally
these functions expect that the callers sanitize the chapter number.
This went wrong at least in add_seek_osd_messages() (which displayed
a chapter "-1" when chapters were not available). Make these functions
a bit friendler and add some reasonable checks and fallbacks, which
fixes the mentioned chapter seeking case as well.
This affects the "show_progress" command, by defualt on the 'P' key.
If there are complaints, I'll probably remove it again. (It looks
relatively annoying, but it also valueable information... sort of.)
Seeks can be performed with OSD bar invisible (e.g. "osd-msg seek ..."
command), and then an already visible bar won't be updated. But the bar
will stick around until the OSD text is hidden. This is confusing, so
change it that the bar is updated. (Making the bar disappear on such
seeks would require much more changes, so we're lazy and go with this
commit.)
The seek bar appeared to be "stuck" to the start of the current chapter.
This is a regression from 630a2b1. This commit assumed that hrseek_pts
would always contain the hrseek target time (when hrseek_active==true).
But this is not always the case: when playing timeline stuff (e.g.
ordered chapters), hrseek framedropping is abused to handle an obscure
corner case, and then hrseek_pts contains something completely unrelated
to the current playback time. See the added comment in mplayer.c and
commit c1232c9.
Fix this by trying something else to get a correct time "during"
hr-seeks. mpctx->restart_playback looks ideal, because it's set while
audio is being synced / audio buffers being filled, so we know that the
audio time is probably bogus while it is set. Let's hope this is
correct.
Also move the lang field into the general stream header. (SH_COMMON is
an old hack to "share" code between audio/video/sub headers.)
There should be no functional changes, other than not printing stream
info in verbose mode or with slave mode. (The frontend already prints
stream info, and this is just a leftover when individual demuxers did
this, and slave mode remains broken.)
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.
Simplify --no-config and make it a normal flag option, and doesn't take
an argument anymore. You can get the same behavior by using --no-config
and then --include to explicitly load a certain config file.
Make --no-config work for input.conf as well. Make it so that
--input:conf=file still works in this case. As a technically unrelated
change, the file argument now works as one would expect, instead of
making it relatively to "~/.mpv/". This makes for simpler code and
easier to understand option semantics. We can also print better error
messages.
Doesn't have much of a purpose for normal playback. You can get
milliseconds display with --osd-fractions. It's also possible to build
a custom status line with --status-msg.
This gives more space on the status line and, in my opinion, is a bit
less annoying.
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}"
This fixes a problem that happened with syncplay.pl [1] when ad_mpg123
was in use, and get_current_time() returning a bogus time position.
This only happens during seeking; the reported time is correct after the
seek is done.
The audio PTS as returned by playing_audio_pts() is simply bogus during
hr-seek. With ad_ffmpeg, it was actually set to MP_NOPTS_VALUE during
seeking, so get_current_time() did a fallback to the video PTS. However,
ad_mpg123 is different and explicitly decodes some audio when resetting
on seek (reasons why it does this unknown and uninvestigated; apparently
it's to reinit libmpg123). As a result, the audio PTS was set to the
start position of the seek (or something similar), which could be very
different from the seek target time.
This confused syncplay. It got the bogus time because it spams the
player with read commands to the "time-pos" property, so this corner
case was hit.
Fix this by making get_current_time() return the seek target time if
hr-seek is active. This should make behavior the same as before commit
3f949cf "mplayer: prefer audio PTS over video PTS for status line".
[1] http://syncplay.pl
When doing a framestep while there is no more video, nothing happened,
and audio continued to play. When advancing to the next file, the player
was paused. Fix it so that it always pauses (except on very low frame
rate video, which is yet another corner case).
We also change the meaning of framestepping a bit: in audio only mode,
framstepping unpauses for a single playloop iteration. This is probably
not useful at all, but makes the code a bit more simpler/uniform.
Just like the previous commit, this matters most for audio files with
cover art, for which this special case is the normal case.
mpctx->delay is used to control audio/video sync. If more audio than
video has been played, it grows larger, meaning A/V desync is happening.
This logic is a bit broken when video has ended, and audio is still
playing. In that case, it tries to read additional video frames from the
video decoder (because even if you don't feed new packets to the
decoder, it could still return delayed frames). For that, the code to
determine whether frames should be dropped is invoked
(check_framedrop()). This function detects that video is behind audio (mpctx-
>delay growing big),
and attempts to issue a framedrop.
Reset mpctx->delay if there's no more video.
This fixes the the frame drop display "counting up" on each playloop
iteration when playing audio files with cover art. These files are
basically audio+video files with a single video frame. When playing
these files the the corner case of having run out of video while audio
is still playing is the normal case.
Also reset mpctx->last_av_difference. This is not updated anymore if
video ends (since update_avsync() sets it, but it's not called if
video_left is false). This removes the "stuck" A/V sync value when video
ends. With audio files containing cover art we would display a
meaningless value over the duration of the whole file otherwise.
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.
Should be dead code. Stream selection is handled either during
demuxer initialization, or via DEMUXER_CTRL_SWITCH_*.
(If there were actually situations where this code did something, it
was probably broken anyway.)
Move things that are used by vo_xv only into vo_xv, same for vo_x11.
Rename some functions exported by x11_common, like vo_init to
vo_x11_common. Make functions not used outsode of x11_common.c private
to that file. Eliminate all global variables defined by x11_common
(except error handler and colormap stuff).
There shouldn't be any functional changes, and only code is moved
around. There are some minor simplifications in the X11 init code, as
we completely remove the ability to initialize X11 and X11+VO
separately (see commit b4d9647 "mplayer: do not create X11 state in player frontend"),
and the respective functions are conflated into vo_x11_init() and
vo_x11_uninit().
If we detect Libav, always use the old builtin vobsub decoder (in
spudec.c). Note that we do not want to use it for newer ffmpeg, as
spudec.c can't handle the vobsub packets as generated by the .idx
demuxer, and we want to get rid of spudec.c in general anyway.
Drop queued frames on seek. Reset the internal state of some filters
that seem to need it as well: at least vf_divtc still produced some
frames using the previous PTS.
This fixes weird behavior with some filters on seeking. In particular,
this could lead to A/V desync or apparent lockups due to the PTS of
filtered frames being too far away from audio PTS.
This commit does only the minimally required work to fix these PTS
related issues. Some filters have state dependent on previously filtered
frames, and these are not automatically reset with this commit (even
vf_divtc and vf_softpulldown reset the PTS info only). Filters that
actually require a full reset can implement VFCTRL_SEEK_RESET.
Format changes within a file can e.g. happen in MPEG-TS streams. This
fix also fixes encoding of such files, because ao_lavc is not capable of
reconfiguring the audio stream.
VFCAP_TIMER disables any additional waiting done by mpv in the
playloop. Remove VFCAP_TIMER, but re-use the idea for vo_image and
vo_lavc.
This means --untimed doesn't have to be passed when using --vo=image.
Change the entire filter API to use reference counted images instead
of vf_get_image().
Remove filter "direct rendering". This was useful for vf_expand and (in
rare cases) vf_sub: DR allowed these filters to pass a cropped image to
the filters before them. Then, on filtering, the image was "uncropped",
so that black bars could be added around the image without copying. This
means that in some cases, vf_expand will be slower (-vf gradfun,expand
for example).
Note that another form of DR used for in-place filters has been replaced
by simpler logic. Instead of trying to do DR, filters can check if the
image is writeable (with mp_image_is_writeable()), and do true in-place
if that's the case. This affects filters like vf_gradfun and vf_sub.
Everything has to support strides now. If something doesn't, making a
copy of the image data is required.
Ensure that even if a seek is inaccurate it will not show video from
outside the defined timeline. Previously, seeking to the beginning of
a segment could show frames from before the start of the segment if
the seek was done in inaccurate mode and the demuxer seeked to an
earlier position. Now hr-seek machinery is used to skip at least the
frames that should not be part of playback timeline at all.
Now external subtitles essentially use the playback time, instead of
the segment time.
This is more useful when using external subtitles with mkv ordered
chapters. The previous behavior is not necessarily incorrect, and e.g.
makes it easier to use subtitles directly extracted from ordered
chapters segments. But we consider the new behavior more useful.
Also see commit 06e3dc8.
This is simpler and more useful. We could add a new switch for the old
functionality, but that would probably be more confusing than helpful.
When passing only a single file to the command line, this commit
shouldn't change behavior.
(Classic mplayer provided both features by duplicating the loop
functionality in the "playtree".)
When the last frame is displayed, and a frame step command is issued,
playback ands and advances to the next file. But before this commit,
the next file was played unpause. Fix this, and make sure pause is
kept.
This is better than having just the operating system type decide the
wakeup period, as e.g. when compiling for Win32/cygwin, a wakeup period
of 0.5 would work perfectly fine.
Instead, the default wakeup period is now only decided by availability
of a working select() system call (which is the case on cygwin but not
mingw and MSVC) AND a vo that can provide an event file descriptor or a
similar hack (vo_corevideo). vos that cannot do either need polling for
event handling and now can set the wakeup period to 0.02 in the vo code.
Add `mp_find_config_file` to search different known paths and use that in
ass_mp to look for the fontconfig configuration file.
Some incidental changes spawned by this feature where:
* Buffer allocation for the strings containing the paths is now performed
with talloc. All of the allocations are done on a NULL context, but it still
improves readability of the code.
* Move the OSX function for lookup inside of a bundle: this code path was
currently not used by the bundle generated with `make osxbundle`. The plan
is to use it again in a future commit to get a fontconfig config file.
Keep the currently displayed subtitles even when the user cycles through
subtitle tracks, and the subtitle is decoded by libavcodec (such as
vobsubs). Do this by not clearing the subtitles on reset(). reset() is
also called on seek, so check the start PTS whether the subtitle should
really be displayed (there's already an end PTS). Note that sd_ass does
essentially something similar.
The existing code has checks for whether the PTS reported by the demuxer
is invalid (MP_NOPTS_VALUE). I don't know under what circumstances this
can happens, so fall back to the old behavior if the PTS is invalid.
This slightly improves display of the current playback time in files
with sparse video packets (like video tracks containing a slow MJPG
slideshows as in [1]), or audio files with cover art image attachments.
While the video PTS is always "stuck" at the last frame displayed or
the last seek, audio is usually continuous. Given sane samplerates and
working audio drivers (to query how much of the current audio buffer has
been played), the audio PTS should always be more reliable.
[1] http://www.podtrac.com/pts/redirect.mp3/traffic.libsyn.com/rtpodcast/Rooster_Teeth_Podcast_191.m4a
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.)
This caused errors like:
core/mplayer.c:4308:5: error: implicit declaration of function 'pthread_win32_thread_detach_np' [-Werror=implicit-function-declaration]
It turns out a pthread.h include was missing. It's not clear why this
used to work (or rather, why it happens only sometimes). Possibly some
libraries or system headers recursively include pthread.h under certain
circumstances or configurations.
Fix missing quoting in configure, which led to broken terminal output.
Closes#6.
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).
This affects streams loaded with -subfile and -audiofile. They could get
out of sync when they were deselected, and the main file was seeked. Add
code to seek external files when they are selected (see
init_demux_stream()).
Use avformat_seek_file() under certain circumstances. Both av_seek_frame()
("old" API) and avformat_seek_file() ("new" API) seem to be broken with
some formats. At least the vobsub demuxer doesn't implement the old API
(and the old API doesn't fallback to the new API), while the fallback
from new API to old API gives bad results. For example, seeking forward
with small step sizes seems to fail with the new API (tested with
Matroska by trying to seek 1 second forward relative to priv->last_pts).
Since only subtitle demuxers implement the new API anyway, checking
whether iformat->read_seek2 is set to test whether the old API is not
supported gives best results. This is a hack at best, but makes things
work.
Remove backwards seeking on seek failure. This was annoying, and only
was there to compensate for obscure corner cases (see 1ad332). In
particular, files with completely broken seeking that used to skip back
to the start on every seek request may now terminate playback.
Do this only if demux_lavf is used. Using demux_mpg and the ffmpeg DVD
subtitle decoder doesn't work. The problem is probably that demux_mpg
doesn't join split sub packets, while demux_lavf does. The internal
DVD sub decoder (spudec.c) can, while ffmpeg's dvdsub can't. I do not
know whether this is the actual problem.
If DVD playback is used, create "fake" vobsub-style text extradata
(like .idx files) to pass resolution and palette information to the
ffmpeg decoder. We could use the "palette" AVOpt and avcodec_set_dimensions()
instead, but it's actually simpler this way. Note that the decoder
doesn't parse any other fields. Also note that DVD playback still uses
demux_mpg by default, so this code is inactive unless -demuxer lavf is
specified. This is mainly preparation for the case when we manage to get
rid of demux_mpg for DVD playback.
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.
This commit is separate from the previous one to separate our own
changes from changes merged from mplayer2 (as far as that was possible).
Make it easier for stream implementations to request being cached. Set
a default cache size in stream.c, and remove them from various stream
implementations. Only MS streaming support sets a meaningful cache size.
Make querying cache size saner. This reduces the amount of #ifdefs
needed.
Code enabling the cache by default for network streams did that by
modifying the value of the "cache" option. This wasn't sane, as
multiple streams may be created and all share the same options. Change
the code to not modify options but store data in the stream instance
instead.
Conflicts:
core/mplayer.c
demux/demux.c
stream/cache2.c
stream/network.c
stream/network.h
stream/pnm.c
stream/stream.c
stream/stream_rtp.c
Merged from mplayer2 commit e26070. Note that this doesn't solve any
actual bug, as the playlist crashing bug has been fixed before.
Since the global cache size option value is not overwritten anymore, the
option doesn't need to be restored on end of playback (M_OPT_LOCAL).
Based on a patch by qyot27. Add export LC_ALL=C on top of version.sh to
make the output locale independent.
Note that the build time will not be updated on every "make" invocation,
but only when the git revision is updated. This is a good thing, as
repeated make invocations should not rebuild the binary. (This would
break "sudo make install" too.)
This caused e.g. "--alang=" (without anything following) to be printed
in the terminal output when the file specified no language for the
track. Introduced by commit 9085b8.
Enable printf format warnings for set_osd_[t]msg.
Remove the pointless assertion in mplayer.c (the assertion proved that
the following NULL check is probably pointless, but leave that check
anyway for robustness - it's not really clear whether it's needed).
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.
Do not load codecs.conf files located in $PREFIX/etc/mpv/ or ~/.mpv/.
There really is no use for this, other than possibly breaking things.
It's still possible to use --codecs-file explicitly to load an external
config file, and this option can be used in ~/.mpv/config.
While we're at it, remove the global codecs_file variable, and another
unused variable.
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
When a video filter returned inf as PTS, the player crashed. One
reason for this was that decode_audio() was called with a negative
minlen parameter, which at some point caused it to call a memory
allocation function with a ridiculous value, triggering an out of
memory code path in talloc.c. (talloc.c has been modified to abort()
on out of memory situations.)
Fix this by sanity checking minlen in decode_audio(). (The check
against outbuf->len always succeeded, because it's an unsigned
comparison.)
Make an existing sanity check in mplayer.c more robust: check for NaN
too, which happens if the video PTS is inf.
This happened with "-vf pullup,softpulldown" (but is not triggered when
the following commit is applied).
ao_play() can fail; in that case a negative error code is returned.
This error code is returned by write_to_ao() in turn. The function
fill_audio_out_buffers(), which calls write_to_ao(), doesn't check for
any error codes, and will likely trigger the assertion following the
function call. Change write_to_ao() to return 0 on failure to hopefully
prevent crashes when AOs fail.
The language string was dynamically allocated, which completely fails
if the cache is forked (which it usually is). Change it back to a fixed
length string, like the original code had it.
The --start and --end switch now accept a chapter number. The chapter
number is prefixed with '#', e.g. "--start=#2" jumps to chapter 2.
The chapter support might be able to replace --chapter completely, but
for now I am not sure how well this works out with e.g. DVDs and BDs,
and a separate --chapter option is useful interface-wise.
(This was supposed to be added in 51503a, but apparently the fixup
commit adding it was lost in a rebase. This might also be the reason
for the mess-up fixed in 394285.)
Make demux_lavf not error out if no video or audio track is present.
This allows opening subtitle files with the demuxer.
Improve the test whether subtitles read from demuxers must do explicit
packet reads. (I'm not sure whether always doing these reads could have
bad effects, such as reading too many audio and video packets at once,
so be conservative.)
The computation for the A/V sync value was inside print_status(). Move
it into its own function; this makes things simpler and gets rid of some
minor dead code.
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.
Rename the -ss option to -start, and -endpos to -length. Add a -end
option. The -end option always specifies an absolute end time, as
opposed to -endpos/-length.
All these options (--start, --end, --length) now accept relative times.
Percent positions (e.g. "--start=30%") are interpreted as fractions of
the file duration. Negative times (e.g. "--start=-1:00) are interpreted
relative to the end of the file. Chapters (e.g. "--start=#3") yield the
chapter's time position.
The chapter support might be able to replace --chapter completely, but
for now I am not sure how well this works out with e.g. DVDs and BDs,
and a separate --chapter option is useful interface-wise.
Using --loop=inf on an unseekable file would put mpv (and all other
mplayers as well) into an endless loop, trying to seek to the start of
the file on each playback loop iteration. When the seek fails, playback
simply remains in the at-end-of-file state, and tries to issue a new
seek command for looping.
Fix by checking if the seek command fails, and abort looping in this
case. For that, queue_seek() is replaced with seek(). Due to the
circumstances, these two calls happen to be equal in this case: the
seek is absolute (i.e. no seek coalescing done), and the execution of
queued seeks is right after the loop code anyway.
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.
