MSWindows does not have properly working support for detecting events
on file descriptors. As a result the current mplayer2 code does not
support waking up when new input events occur. Make the central
playloop wake up more often to poll for events; otherwise response
would be a lot laggier than on better operating systems during pause
or other cases where the process would not otherwise wake up.
Make A/V sync at the start of playback with nonzero --delay behave the
same way as it does when seeking to the beginning later, meaning video
plays from the start and audio is truncated or padded with silence to
match timing. This was already the default behavior in case the
streams in the file started at different times, but not if the
mismatch was due to --delay. Trigger similar audio synchronization
when switching to a new video stream. Previously, switching a video
stream on after playing for some time in audio-only mode was buggy and
caused initial desync equal to the duration of prior audio-only
playback.
Uninitialize video and audio outputs when switching to a file without
a corresponding track (audio-only file / file with no sound), or when
entering --idle mode. Switching track choice to "off" during playback
already did this.
It could be useful to have a mode where the video window stays open
even when no video plays, but implementing that properly would require
more than just leaving the window on screen like the code did before
this commit.
The player tried to disable mute before exiting, so that if mute is
emulated by setting volume to 0 and the volume setting is a
system-global one, we don't leave it at 0. However, the logic doing
this at process exit was flawed, as volume settings are handled by
audio output instances and the audio output that set the mute state
may have been closed earlier. Trying to write reliably working logic
that restores volume at exit only would be tricky, so change the code
to always unmute an audio driver before closing it and restore mute
status if one is opened again later.
MPlayer volume control was originally implemented with the assumption
that it controls a system-wide volume setting which keeps its value
even if a process closes and reopens the audio device. However, this
is not actually true for --softvol mode or some audio output APIs that
only consider volume as a per-client setting for software mixing. This
could have annoying results, as the volume would be reset to a default
value if the AO was closed and reopened, for example whem moving to a
new file or crossing ordered chapter boundaries. Add code to set the
previous volume again after audio reinitialization if the current
audio chain is known to behave this way (softvol active or the AO
driver is known to not keep persistent volume externally).
This also avoids an inconsistency with the mute flag. The frontend
assumed the mute status is persistent across file changes, but it
could be similarly lost.
The audio drivers that are assumed to not keep persistent volume are:
coreaudio, dsound, esd, nas, openal, sdl. None of these changes have
been tested. I'm guessing that ESD and NAS do per-connection
non-persistent volume settings.
Partially based on code by wm4.
Current volume was always queried from the the audio output driver (or
filter in case of --softvol). The only case where it was stored on
mixer level was that when turning off mute, volume was set to the
value it had before mute was activated. Change the mixer code to
always store the current target volume internally. It still checks for
significant changes from external sources and resets the internal
value in that case.
The main functionality changes are:
Volume will now be kept separately from mute status. Increasing or
decreasing volume will now change it relative to the original value
before mute, even if mute is implemented by setting AO level volume to
0. Volume changes no longer automatically disable mute. The exception
is relative changes up (like the volume increase key in default
keybindings); that's the only case which still disables mute.
Keeping the value internally avoids problems with granularity of
possible volume values supported by AO. Increase/decrease keys could
work unsymmetrically, or when specifying a smaller than default
--volstep, even fail completely. In one case occurring in practice, if
the AO only supports changing volume in steps of about 2 and rounds
down the requested volume, then volume down key would decrease by 4
but volume up would increase by 2 (previous volume plus or minus the
default change of 3, rounded down to a multiple of 2). Now, the
internal value will keep full precision.
Stop trying to read terminal input if a read attempt returns EOF. The
most important case where this matters is when someone runs the player
with stdin redirected from /dev/null and without specifying
--no-consolecontrols. This used to cause 100% CPU load while paused,
as select() would continuously trigger on stdin (the need for
--no-consolecontrols was not apparent to people with older mplayer
versions, as input reading was less efficient and latencies like
hardcoded sleeps kept CPU use well below 100%). Now this will only
cause a "Dead key input" error message.
Make the code read current real time again after drawing OSD. This
ensures time taken in OSD drawing is properly deducted from the
duration of the following sleep. The main practical effect is to avoid
the A-V field on the status line staying at a value a couple of
milliseconds above 0 (depending on VO).
Fix a missing check that could sometimes result in video frames being
shown after specified end pts (end of timeline segment or --endpos).
Fix mistaken video EOF detection after aspect change in video stream,
when there is no current valid visible frame but the next frame is
already buffered in VO.
For ao_pulse, the current latency is not a good indicator of how soon
the AO requires new data to avoid underflow. Add an internal pipe that
can be used to wake up the input loop from select(), and make the
pulseaudio main loop (which runs in a separate thread) use this
mechanism when pulse requests more data. The wakeup signal currently
contains no information about the reason for the wakup, but audio
buffers are always filled when the event loop wakes up.
Also, request a latency of 1 second from the Pulseaudio server. The
default is normally significantly higher. We don't need low latency,
while higher latency helps prevent underflows reduces need for
wakeups.
Timeline handling converted the pts values from demuxed subtitles to
timeline scale. Change the code to do most subtitle handling in
original subtitle source pts, and instead convert current playback
timeline pts to those units when deciding which subtitle to show.
The main functionality changes are that now demuxed subtitles which
overlap chapter boundaries are handled correctly (at least for libass
subtitles), and external subtitles are assumed to use same pts scale
as current source (this needs improvements later).
Before, a video subtitle that had a duration continuing past the end
of the chapter would continue to be shown for the original duration,
even if the chapter ended and playback switched to a position in the
source where the subtitle shouldn't exist. Now, the subtitle will
correctly end.
Before, external subtitle files were interpreted as specifying pts
values in timeline scale. Now, they're interpreted as specifying pts
values in source file time scale, for _every_ source file. This is
probably more likely to be what the user wants for the "main" source
file in case there is one, but almost certainly not quite right for
multiple source files where the same subs could be shown over
different scenes. If the user wants them to match some main source
file, it's probably still better to have incorrect extra subs for
video from some files than to have every subtitle appearing at the
wrong time. The new code makes it easier to change the interpretation
of the subtitle times, and some configurability should be added in
the future.
When switching to a timeline part from another file, decoders were
reinitialized after doing the demuxer-level seek. This is necessary
for audio because some decoders read from the demuxer stream during
initialization and the previous stream position before seek could have
been at EOF. However, this initialization sequence could lose first
subtitles or first part of audio.
The problem for subtitles was that the seek itself or audio
initialization could already have buffered subtitle packets from the
new position, and the way subtitles are reinitialized flushes packet
buffers. Thus early subtitles could be lost (even if they were demuxed
- unfortunately demuxers may not know about still active subtitles
earlier in the file, but that's another issue). Fix this by moving
subtitle and video reinitialization before the demuxer seek; they
don't have the problems which prevent that for audio.
Audio initialization can already decode and buffer some output.
However, the seek_reset() call done last would then throw away this
buffered output. Work around this by adding an extra flag to
seek_reset().
Restructure parts of the code in the main play loop. The main
functionality difference is that if a video track ends first, now
audio will continue to be played until it ends too.
Now the process also wakes up less often if there's no need to update
video or audio. This will reduce unnecessary wakeups especially when
paused, but may make handling of input events laggier when fd-based
notifications are not supported (like most input on Windows).
Change the terminal status line to show "???" instead of a huge
negative number if audio or video pts is missing (there was a partial
workaround for audio before, but not video or A-V difference).
The name "MPlayer2" isn't used anywhere. It's either "MPlayer" or
"mplayer2". Make it more consistent by using "mplayer2" instead.
Note that the version string passed as network user-agent changes from
"MPlayer" to "mplayer2" as well.
The current code tried to print -1000 as unsigned integer if the
chapter time was unknown. Print -1 instead. This affects only the
-identify output used for slave mode, such as ID_CHAPTER_0_START.
Remove the old EDL implementation that was activated with the --edl
option. It is mostly redundant and inferior compared to the newer
demux_edl support, though currently there's no support for using the
same EDL files with the new implementation and the mute functionality
of the old implementation is not supported. The main reason to remove
the old implementation at this point is that the mute functionality
would conflict with following audio volume handling changes, and
working on the old code would be a wasted effort in the long run as at
some point it would be removed anyway.
The --edlout functionality is kept for now, even though after this
commit there is no code that could directly read its output.
Windows uses a legacy codepage for char* / runtime functions accepting
char *. Using UTF-8 as the codepage with setlocale() is explicitly
forbidden.
Work this around by overriding the MSVCRT functions with wrapper
macros, that assume UTF-8 and use "proper" API calls like _wopen etc.
to deal with unicode filenames. All code that uses standard functions
that take or return filenames must now include osdep/io.h. stat()
can't be overridden, because MinGW-w64 itself defines "stat" as a
macro. Change code to use use mp_stat() instead.
This is not perfectly clean, but still somewhat sane, and much better
than littering the rest of the mplayer code with MinGW specific hacks.
It's also a bit fragile, but that's actually little different from the
previous situation. Also, MinGW is unlikely to ever include a nice way
of dealing with this.
The _UWIN define causes the mingw headers not to declare deprecated (on
Windows) function names such as open and mkdir. But the code uses these. I
have no idea why this used to work (if it even did), but the original
reason why it was defined seems to have vanished.
This adds the --screenshot-template option, which specifies a template
for the filename used for a screenshot. The '%' character is parsed as
format specifier. These format specifiers insert metadata into the
filename. For example, '%f' is replaced with the filename of the
currently played file.
The following format specifiers are available:
%n Insert sequence number (padded with 4 zeros), e.g. "0002".
%0Nn Like %n, but pad to N zeros (N = 0 to 9).
%n behaves like %04n.
%#n Like %n, but reset the sequence counter on every screenshot.
(Useful if other parts in the template make the resulting
filename already mostly unique.)
%#0Nn Use %0Nn and %#n at the same time.
%f Insert filename of the currently played video.
%F Like %f, but with stripped file extension ("." and rest).
%p Insert current playback time, in HH:MM:SS format.
%P Like %p, but adds milliseconds: HH:MM:SS.mmmm
%tX Insert the current local date/time, using the date format X.
X is a single letter and is passed to strftime() as "%X".
E.g. "%td" inserts the number of the current day.
%{prop} Insert the value of the slave property 'prop'.
E.g. %{filename} is the same as %f. If the property doesn't
exist or is not available, nothing is inserted, unless a
fallback is specified as in %{prop:fallback text}.
%% Insert the character '%'.
The strings inserted by format specifiers will be checked for
characters not allowed in filenames (including '/' and '\'), and
replaced with the placeholder '_'. (This doesn't happen for text that
was passed with the --screenshot-template option, and allows specifying
a screenshot target directory by prefixing the template with a relative
or absolute path.)
Callign add_step_frame is not necessary, because mplayer always decodes
at least one frame when starting a new file. Calling pause_player is
sufficient, and unlike add_step_frame doesn't play any audio.
The terminal OSD line was written with mp_msg(MSGT_CPLAYER, ...) but
erased with printf(). This meant that disabling MSGT_CPLAYER messages
would prevent the terminal line from being printed, but a line
(probably unrelated) would still be cleared. Change the clearing code
to use mp_msg(MSGT_CPLAYER, ...) too.
If mplayer is started with -msglevel cplayer=-1, there can't be any
terminal OSD output, but the terminal line was still cleared
unconditionally. Fix this by using mp_msg(), which will throw away the
output to clear the terminal if disabled.
Fixes#154.
The --paused option will start the player in paused state. That means it
will start out with a still image of the first frame.
This can be useful in combination with --ss to inspect a certain frame.
Caveat: this plays a small bit of audio at the start, which might be
perceived as an annoying artifact. This is because this is implemented
by frame stepping after initialization in order to decode and display
the first video frame.
The vd_ffmpeg decode() function returned without doing anything if the
input packet had size 0. This meant that flushing buffered frames at
EOF did not work. Remove this test. Have the core code skip such
packets coming from the file being played instead (Libav treats
0-sized packets as flush signals anyway, so better assume such packets
do not represent real frames with any codec).
Remove the private bswap and intreadwrite.h implementations and use
libavutil headers instead.
Originally these headers weren't publicly installed by libavutil at
all. That already changed in 2010, but the pure C bswap version in
installed headers was very inefficient. That was recently (2011-12)
improved and now using the public bswap version probably shouldn't
cause noticeable performance problems, at least if using a new enough
compiler.
Now the option --term-osd=force will cause mplayer to display all OSD
messages on the terminal, even if there is video.
Possible values for --term-osd:
- auto: use video OSD, or of there's no video, the terminal (default)
- off: always use video for OSD
- force: always use terminal for OSD
-term-osd and --term-osd are equivalent to --term-osd=force. This
changes the meaning of the option, since -term-osd used to enable the
OSD default behavior, i.e. --term-osd=auto.
-noterm-osd has the same effect as --term-osd=off, and is kept for
compatibility.
Implementation note:
The location for the OSD text was shared between the two code paths (it
was in osd_state.osd_text). We can't rely on the fact that the video-OSD
update code normally isn't run when --term-osd is called. When e.g.
panscan is updated, the video OSD code will draw the OSD anyway. This
would sometimes show unwanted OSD text on the video.
Deal with this by putting the current terminal-OSD text in a different
place (in MPContext.terminal_osd_text) to deal with this.
Playing a .cue file directly will now parse the .cue file, and load and
play the file(s) referenced in the cue. If multiple files are referenced,
a timeline including all files will be created to create the impression
of a single, flat audio file containing all the tracks.
For each track, a chapter is created. The chapter navigation commands can
be used to jump between tracks. The chapter titles will use the string
provided by the track's TITLE cue command. (The -identify command can be
used to print all chapters in a not so user friendly way.)
Other than the chapter names, there is no attempt at displaying or exposing
any other meta data contained in the cue files yet.
The handling (or lack of thereof) of gaps (track pregaps and postgaps) is
probably not correct yet. In general, mplayer's mapping of tracks to the
source audio files can be verified by examining the timeline, which will
be printed when passing the -v switch.
Note that this has nothing to do with the old cue:// support. The old code
isn't touched, and is still only able to play .cue/.bin pairs. Prefixing a
.cue file with cue:// will always invoke the old code, while playing a .cue
file directly (i.e. "mplayer file.cue") will always use the new code.
Playing audio images (.cue/.bin pairs of files) doesn't work yet.
I'm not sure what's the point of this feature. Aside from that, the EDL
code is relatively buggy anyway, and I see no reason why such an obscure
feature should be left in, if it possibly causes bugs.
When you mute audio, mplayer is supposed to restore the volume controls
on exit. This affects when --softvol isn't used and the audio output
driver volume controls directly affect the system wide volume controls.
This wasn't done in some cases.
At least in the case when switching to no audio track and then switching
back, the volume settings were not restored with --softvol. Fix this by
moving the call restoring the settings to a better place.
Since the recent OSD redraw changes, every GUI expose event causes the
message "===== PAUSE =====" to be printed on console. This was a bit
annoying, so change it so that it is only printed once when going into
paused mode. It's also printed again if the cache status changes (when
playing URLs), or when the status line is printed during pause mode (when
you seek while paused).
This also removes some minor code duplication.
When the OSD was enabled and the player was paused by executing the
frame_step command, the OSD still displayed the icon indicating
playback. Fix this and always set the proper icon when the pause
state is changed.
When --softvol is enabled, the volume set by the "volume" property is
reset when changing to a new file or crossing ordered chapter boundaries.
Fix this by explicitly restoring the volume on audio reinitialization.
Now the behavior with --softvol should be the same as if a system mixer
is used, and the volume should be persistent across file changes.
This also works around an inconsistency with the mute flag. The frontend
assumed the mute flag is persistent across file changes, which was not
true with --softvol.
If not resetting the volume on playing new files is undesired, it can
be avoided by putting volume=100 in the mplayer config file.
Remove no longer necessary tests from hrseek code. As a result each
field of vo_vdpau framerate-doubling deinterlace modes is now
considered as a possible seek target.
Remove code refreshing window contents after events such as resize
from vo_vdpau, vo_gl and vo_xv. Instead have them simply set a flag
indicating that a refresh is needed, and have the player core perform
that refresh by doing an OSD redraw. Also add support for updating the
OSD contents over existing frames during slow-but-not-paused playback.
The VOs now also request a refresh if parameters affecting the picture
change (equalizer settings, colormatrix, VDPAU deinterlacing setting).
Even previously the picture was typically redrawn with the new
settings while paused because new OSD messages associated with setting
changes triggered a redraw, but this did not happen if OSD was turned
off.
A minor imperfection is that now window system events can trigger a
single one-frame step forward when using vo_xv after pausing so that
vo_xv does not yet have a copy of the current image. This could be
fixed but I think it's not important enough to bother.
Previously the core sent VFCTRL_REDRAW_OSD to change OSD contents over
the current frame. Change this to VFCTRL_REDRAW_FRAME followed by
normal EOSD and OSD drawing calls, then vo_flip_page(). The new
version supports changing EOSD contents for libass-rendered subtitles
and simplifies the redraw support code needed per VO. vo_xv doesn't
support EOSD changes because it relies on vf_ass to render EOSD
contents earlier in the filter chain.
vo_xv logic is additionally simplified because the previous commit
removed the need to track the status of current and next images
separately (now each frame is guaranteed to become "visible" soon
after we receive it as "next", with no VO code running in the interval
between).
Separate passing a new frame to VOs using the new API into two steps.
The first, vo_draw_image(), happens after a new frame is available
from the filter chain. In constrast to old behavior, now the frame is
not actually rendered yet at this point (though possible slice draw
calls can already reach the VO before). The second step,
vo_new_frame_imminent(), happens when we're close enough to the
display time of the new frame that we'll commit to flipping it as the
next action and will not change the OSD over the previous frame any
more.
This new behavior fixes a previous problem with vo_vdpau and vo_gl in
the situation where the player is paused after decoding a new frame
but before flipping it; previously changing OSD in that state would
switch to the new frame as a side effect. It would also allow an easy
way to fix extra output files produced with something like "--vo=png
--frames=1" with precise seeking, but this is not done yet.
The code now relies on a new mp_image from the filter chain staying
valid even after the vf_vo put_image() call providing it returns. In
other words decoders/filters must not deallocate or otherwise
invalidate their output frame between passing it forward and returning
from the decode/filter call.
Add a VO command (VOCTRL_SCREENSHOT) which requests a screenshot
directly from the VO. If VO support is available, screenshots will be
taken instantly (no more 1 or 2 frames delay). Taking screenshots when
hardware decoding is in use will also work (vdpau). Additionally, the
screenshots will now use the same colorspace as the video display.
Change the central MPContext to be allocated with talloc so that it
can be used as a talloc parent context.
This commit does not yet implement the functionality for any VO (added
in subsequent commits).
The old screenshot video filter is not needed anymore if VO support is
present, and in that case will not be used even if it is present in
the filter chain. If VO support is not available then the filter is
used like before. Note that the filter still has some of the old
problems, such as delaying the screenshot by at least 1 frame.
Some demuxers do not accurately seek to a keyframe before a given
time but instead start too late. This means that precise seeks cannot
work either. Most notably the libavformat mpeg demuxer exhibits this
behavior depending on the file being played (with the internal mpeg
demuxer precise seeks don't work at all). Add new option
--hr-seek-demuxer-offset which can be used as a workaround with such
demuxers. The value of the option is subtracted from the seek target
position given to the demuxer when doing a precise seek.
Before, precise seeking only worked if there was a video stream; in
the audio-only case playback always started from the demuxer seek
position. Add code to cut away samples from the demuxer seek position
to the seek target position.
Information about individual chapters was printed during demuxer
opening phase, and total chapter count (ID_CHAPTERS) was printed
according to mpctx->demuxer->num_chapters. When playing a file with
ordered chapters, this meant that chapter information about every
source file was printed individually (even though only the chapters
from the first file would be used for playback) and the total chapter
count could be wrong. Remove the printing of chapter information from
the demuxer layer and print the chapter information and count actually
used for playback in core print_file_properties().
Also somewhat simplify the internal chapters API and remove possible
inconsistencies.
Something like the OSD menu functionality could be useful. However the
current implementation has several problems and would require a
relatively large amount of work to get into good shape. As far as I
know there are few users of the existing functionality. Nobody is
working on the existing code and keeping it compiling at all while
changing other code would require extra work. So delete the menu code
and some related code elsewhere that's used by nothing else.
Commit dde8b753e4 merged an mplayer1 change (r31328) that set
correct_pts to true if FPS was not set (on the assumption that
correct-pts mode could provide proper timing without FPS). As the
merge commit noted this change was somewhat questionable, as the
option shouldn't really change after things have already been
initialized. After recent changes it can cause an outright crash
(assert in ds_get_packet2() from 9c7c4e5b7d fails). Remove the hack.
Also only print a warning about not having FPS if correct_pts is not
set (in correct_pts mode not having FPS shouldn't be a real problem,
as everything is based on timestamps anyway).
Pass the libavformat packet side_data field from demux_lavf to
vd_ffmpeg. Libavcodec/libavformat use this field for palette data, and
passing it is required for the playback of some paletted video codecs.
The implementation works by giving vd_ffmpeg a copy of the struct
demux_packet used to store the video packet (from which it can access
the avpacket field). The definition of struct demux_packet is moved to
new file demux_packet.h so that vd_ffmpeg.c can use it without
including all of demuxer.h.
The committed version of 58834653c0 ("dvdnav: make
mp_dvdnav_save_smpi() more robust") was somehow missing one line which
caused a crash with dvdnav. Add it back.
The OSD text buffers (mp_osd_msg_t.text and osd_state.text) used to be
static arrays, with the buffer sizes spread all over the code as magic
constants. Make the buffers dynamically allocated and remove the
arbitrary length limits.
Remove outdated "!mpctx->sh_video" checks in chapter seeking and
naming functions left over from when timeline functionality did not
support audio-only case.
I had delayed reformatting mplayer.c as I wanted to split it, but
since I didn't come up with a good way to do that I'll clean up the
messy formatting now.
Initialize mpctx->last_chapter_seek to -2 instead of -1. This changes
get_current_chapter() return value to -2 for files which have no
chapters. -2 is used by some commands related to chapters to recognize
files without chapters and return failure without any effect in that
case.
Seeking while paused could result in the current audio pts being
reported incorrectly due to relevant variables not being reinitialized
after the seek until more audio was played. When playing audio-only
files, this meant that current overall playback position could be
reported incorrectly which in turn could break further seeks. Improve
things on two levels: First, store the seek target position and use
that as the current playback position for audio-only files until
things can be reinitialized. Second, try to reinitialize audio
decoding enough to know its current pts even while paused. Also avoid
printing the actual huge negative value of MP_NOPTS_VALUE on the
status line when pts could not be determined.
Change written_audio_pts() and playing_audio_pts() to return
MP_NOPTS_VALUE if no reasonable pts estimate is available. Before they
returned some incorrect value typically around zero (but not
necessarily exactly that).
Allow writing commandline options with two leading dashes. In this
mode a parameter for the option, if any, follows after a '=';
following separate commandline arguments are never consumed as a
parameter to a previous double-dash option.
Flag options may omit parameter and behave like old single-dash
syntax. "--fs=yes", "--fs=no" and "--fs" are all valid; the first two
behave like configuration file "fs=yes" and "fs=no", and last is the
same as old "-fs" (same effect as "--fs=yes").
Make per-file loop option start from --ss position, not always 0.
Do looping in more cases; before looping was only done when
encountering real end of file, now it also happens for example at
--endpos or --frames limits. Also move the --ss option to the option
struct.
The --ass-force-style option was only applied when the main libass
library handle was created. Thus any per-file option changes later had
no effect. Do the ass_set_style_overrides() call in per-file
initialization instead so that possible changes will be applied. Also
move the option variable to the option struct.
Current libass will crash (usually) if you set style overrides to a
nonempty value, then an empty one. It'll be easier to trigger this bug
after this commit, but the problem is not on mplayer2 side. The fix is
trivial so hopefully there will be a fixed libass soon.
Do the global initialization of libavcodec and libavformat
(avcodec_register_all(), av_register_all()) immediately on program
startup and remove the initialization calls from various individual
modules that use libavcodec/libavformat functionality.
Setting O_NONBLOCK on a file descriptor also affects all other fds
that share the same underlying open file description, and in case of
stdin such sharing is likely. Making stdin nonblocking can also make
stdout nonblocking (they may be the same connection to a terminal),
and it can also affect other processes (in "program1 | program2", the
shell may give the same terminal connection to program1 as stdin and
to program2 as stdout, thus program1 making its stdin nonblocking also
turns program2's stdout nonblocking).
To avoid these problems stop making fd 0 nonblocking. After the
previous commit this should no longer cause problems as long as
select() does not spuriously report the fd as readable.
Rework much of the logic related to reading from event sources and
queuing commands. The two biggest architecture changes are:
- The code buffering keycodes in mp_fifo.c is gone. Instead key input
is now immediately fed to input.c and interpreted as commands, and
then the commands are buffered instead.
- mp_input_get_cmd() now always tries to read every available event
from every event source and convert them to (buffered) commands.
Before it would only process new events until one new command became
available.
Some relevant behavior changes:
- Before commands could be lost when stream code called
mp_input_check_interrupt() which read commands (to see if they were
of types that triggered aborts during slow IO tasks) and then threw
them away. This was especially an issue if cache was enabled and slow
to read. Fixed - now it's possible to check whether there are queued
commands which will abort playback of the current file without
throwing other commands away.
- mp_input_check_interrupt() now prints a message if it returns
true. This is especially useful because the failures caused by
aborted stream reads can trigger error messages from other code that
was doing the read; the new message makes it more obvious what the
cause of the subsequent error messages is.
- It's now possible to again avoid making stdin non-blocking (which
caused some issues) without reintroducing extra latency. The change
will be done in a subsequent commit.
- Event sources that do not support select() should now have somewhat
lower latency in certain situations as they will be checked both
before and after select()/sleep in input reading; before the sleep
always happened first even if such sources already had queued
input. Before the key fifo was also handled in this manner (first
key triggered select, but if multiple were read then rest could be
delayed; however in most cases this didn't add latency in practice
as after central code started doing command handling it queried for
further commands with a max sleep time of 0).
- Key fifo limiting is more accurate now: it now counts actual
commands intead of keycodes, and all queued keys are read
immediately from input devices so they can be counted correctly.
- Since keypresses are now interpreted immediately, commands which
change keybindings will no longer affect following keypresses that
have already been read before the command is executed. This should
not be an issue in practice with current keybinding behavior.
After commit 39e373aa8d ("options: allocate dynamic options with
talloc") dynamically allocated options must be allocated with talloc.
Code implementing -use-filename-title still set opts->vo_wintitle to a
value from strdup(), triggering an abort when the option was freed.
Fix.
Make mp_dvdnav_save_smpi more robust and ensure consistency of nav
buffer.
It seems that in_size could be negative sometimes, this would cause
crashes if the malloc somehow succeeded.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33599 b3059339-0415-0410-9bf9-f77b7e298cf2
This function was left over from older logic to manipulate the
"codec_path" global variable. Now that variable is fully handled by
the general option system, so that the only effect of the function
was to introduce memory leaks in some circumstances. Delete the
useless function.
Move the buffer storing audio data ready to be fed to the audio output
driver from the audio decoder object to the AO object. This will help
encoding code deal with end of input, and may also be useful to
improve other general gapless audio behavior (as AOs which do not
accept chunks smaller than a certain size may keep them in the buffer
while the decoder changes).
Less data may be dropped now when changing audio filters or switching
timeline parts.
Move the call to m_config_free() to be the last thing done before
exiting, otherwise mp_msg() might stop working if options it uses are
freed/reset.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33380 b3059339-0415-0410-9bf9-f77b7e298cf2
Add some asserts to check that decoders/filters produce complete
samples (byte amounts must be multiples of channels*datatype_size) and
that audio output drivers also accept input in complete units. Fix
ad_pcm which was known to violate this if its last input packet didn't
stop at a sample boundary.
Change ao_pcm to use the new audio output driver API and clean up some
of the code. Rewrite the logic controlling how playback timing works
when using -ao pcm. Deprecate the "fast" suboption; its only effect
now is to print a warning, but it's still accepted so that specifying
it is not an error.
Before, timing with -ao pcm and video enabled had two possible
modes. In the default mode playback speed was rather arbitrary - not
realtime, but not particularly fast. -ao pcm:fast tried to play back
at maximum video playback speed - mostly succeeding, but not quite
guaranteed to work in all cases. Now the default is to play at
realtime speed. The -benchmark option can now be used to get faster
playback (same as the video-only case). In the audio-only case
playback is always maximum speed.
Neither fd 0 slave input (-slave) nor additional opened fds (-input
file=X) were set to nonblocking mode as they should have been. Fix.
Also rename the horribly generic USE_SELECT #define used for a
specific slave input detail.
Commit cbeed30ae8 ("core: wake up a bit less often for audio-only
files") increased the sleep time between audio buffer fills. This
turned out to cause problems on some machines where available audio
buffer sizes are extremely limited (example cases included 85 ms for
stereo and less for multichannel audio). Change the code to check
the amount of buffered audio and shorten sleep times accordingly if
needed.
Such short buffers violate some assumptions made by video timing code,
so they may still cause visible problems in some cases. At least on
some machines using ALSA the problem seems to be caused by bad
configuration defaults (small buffer memory limit which can be
increased).
Analogously to the previous commit, move path handling logic for
loading external vobsub files from mplayer.c to find_subfiles.c.
Based on a commit from Clément Bœsch but fixed and simplified.
Move path handling for loading external subtitle files from mplayer.c
to find_subfiles.c. Now the remaining code in mplayer.c only gets a
list of potential filenames and tries opening those.
Move sub_filenames() and related code from subreader.c to new file
find_subfiles.c. This function is used to find subtitle files that
should be loaded for the current video; this functionality is not
specific to the particular kind of text subtitle handling implemented
in subreader.c.
Also reindent and prettify the moved code a bit.
There is no reason to use manual language list splitting when an
automatic split function is already available.
Some types change from "unsigned char" to "char", but this shouldn't
cause issues since [as]lang settings are unlikely to have characters
above 127.
Add option -osd-fractions which enables display of fractional seconds
when showing the current playback time on OSD.
Based on a patch from Christian <herr.mitterlehner@gsmpaaiml.com> but
with several modifications.
Make the outside interface of audio output handling similar to the
video output one. An AO object is first created, and then methods
called with ao_[methodname](ao, args...). However internally libao2/
still holds all data in globals, and trying to create multiple
simultaneous AO instances won't work.
* edl:
core: support timeline with audio-only files
core: wake up a bit less often for audio-only files
core: audio: cut audio writes at end of timeline part
EDL: add support for new EDL file format
stream.[ch], ass_mp: new stream function for whole-file reads
tl_matroska.c: move the find_files() function here
bstr.[ch], path.[ch]: add string and path handling functions
core: ordered chapters: move timeline creation to timeline/
options: drop support for numeric -demuxer values
cleanup: demuxer.[ch]: remove unused code, make functions static
cleanup: reindent demuxer.h, use struct names for types
Cut audio data written to AO at the point where current timeline part
ends (before, AO buffers were always completely filled, but playback
of the "extra" audio was then cut short by resetting the AO when
switching timeline parts). This doesn't make much difference for
current playback behavior, but will be used by timeline support for
audio-only files and is necessary for future encoding support where
"playback" of written audio cannot be aborted later.
The timeline code previously added to support Matroska ordered
chapters allows constructing a playback timeline from segments picked
from multiple source files. Add support for a new EDL format to make
this machinery available for use with file formats other than Matroska
and in a manner easier to use than creating files with ordered
chapters.
Unlike the old -edl option which specifies an additional file with
edits to apply to the video file given as the main argument, the new
EDL format is used by giving only the EDL file as the file to play;
that file then contains the filename(s) to use as source files where
actual video segments come from. Filename paths in the EDL file are
ignored. Currently the source files are only searched for in the
directory of the EDL file; support for a search path option will
likely be added in the future.
Format of the EDL files
The first line in the file must be "mplayer EDL file, version 2".
The rest of the lines belong to one of these classes:
1) lines specifying source files
2) empty lines
3) lines specifying timeline segments.
Lines beginning with '<' specify source files. These lines first
contain an identifier used to refer to the source file later, then the
filename separated by whitespace. The identifier must start with a
letter. Filenames that start or end with whitespace or contain
newlines are not supported.
On other lines '#' characters delimit comments. Lines that contain
only whitespace after comments have been removed are ignored.
Timeline segments must appear in the file in chronological order. Each
segment has the following information associated with it:
- duration
- output start time
- output end time (= output start time + duration)
- source id (specifies the file the content of the segment comes from)
- source start time (timestamp in the source file)
- source end time (= source start time + duration)
The output timestamps must form a continuous timeline from 0 to the
end of the last segment, such that each new segment starts from the
time the previous one ends at. Source files and times may change
arbitrarily between segments.
The general format for lines specifying timeline segments is
[output time info] source_id [source time info]
source_id must be an identifier defined on a '<' line. Both the time
info parts consists of zero or more of the following elements:
1) timestamp
2) -timestamp
3) +duration
4) *
5) -*
, where "timestamp" and "duration" are decimal numbers (computations
are done with nanosecond precision). Whitespace around "+" and "-" is
optional. 1) and 2) specify start and end time of the segment on
output or source side. 3) specifies duration; the semantics are the
same whether this appears on output or source side. 4) and 5) are
ignored on the output side (they're always implicitly assumed). On the
source side 4) specifies that the segment starts where the previous
segment _using this source_ ended; if there was no previous segment
time 0 is used. 5) specifies that the segment ends where the next
segment using this source starts.
Redundant information may be omitted. It will be filled in using the
following rules:
- output start for first segment is 0
- two of [output start, output end, duration] imply third
- two of [source start, source end, duration] imply third
- output start = output end of previous segment
- output end = output start of next segment
- if "*", source start = source end of earlier segment
- if "-*", source end = source start of a later segment
As a special rule, a last zero-duration segment without a source
specification may appear. This will produce no corresponding segment
in the resulting timeline, but can be used as syntax to specify the
end time of the timeline (with effect equal to adding -time on the
previous line).
Examples:
----- begin -----
mplayer EDL file, version 2
< id1 filename
0 id1 123
100 id1 456
200 id1 789
300
----- end -----
All segments come from the source file "filename". First segment
(output time 0-100) comes from time 123-223, second 456-556, third
789-889.
----- begin -----
mplayer EDL file, version 2
< f filename
f 60-120
f 600-660
f 30- 90
----- end -----
Play first seconds 60-120 from the file, then 600-660, then 30-90.
----- begin -----
mplayer EDL file, version 2
< id1 filename1
< id2 filename2
+10 id1 *
+10 id2 *
+10 id1 *
+10 id2 *
+10 id1 *
+10 id2 *
----- end -----
This plays time 0-10 from filename1, then 0-10 from filename1, then
10-20 from filename1, then 10-20 from filename2, then 20-30 from
filename1, then 20-30 from filename2.
----- begin -----
mplayer EDL file, version 2
< t1 filename1
< t2 filename2
t1 * +2 # segment 1
+2 t2 100 # segment 2
t1 * # segment 3
t2 *-* # segment 4
t1 3 -* # segment 5
+0.111111 t2 102.5 # segment 6
7.37 t1 5 +1 # segment 7
----- end -----
This rather pathological example illustrates the rules for filling in
implied data. All the values can be determined by recursively applying
the rules given above, and the full end result is this:
+2 0-2 t1 0-2 # segment 1
+2 2-4 t2 100-102 # segment 2
+0.758889 4-4.758889 t1 2-2.758889 # segment 3
+0.5 4.4758889-5.258889 t2 102-102.5 # segment 4
+2 5.258889-7.258889 t1 3-5 # segment 5
+0.111111 7.258889-7.37 t2 102.5-102.611111 # segment 6
+1 7.37-8.37 t1 5-6 # segment 7
The select_audio() call was done on the main demuxer, not -audiofile
one (the "if (mpctx->num_sources)" test in the previous code was
always true). Call it on the -audiofile demuxer instead. The
-audiofile stuff still needs a proper cleanup later though.
Windows pthreads requires certain functions to be called to initialize
itself. It can do that through DllMain but no such luck when linked
statically; mplayer needs to call the initialization explicitly.
When doing a precise seek video_out->frame_loaded was left to true
while frames were being skipped. However vo_get_buffered_frame()
always returns success if a frame is already loaded; due to this the
EOF detection in update_video() never triggered, and a hr-seek past
EOF could cause a soft hang (commands were still processed and it was
possible to seek again to exit the loop). This could also happen with
Matroska files using ordered chapters if an underlying file was
actually shorter than the chapter that was supposed to come from it.
Then seeking to a timestamp after the end of the file but before the
end of the chapter would trigger the bug.
Fix the problem by setting frame_loaded to false when we decide to
skip the frame in question.
Add new file timeline/tl_matroska.c. Move the code that parses
ordered chapter information from Matroska files and creates the
timeline structure based on that to the new file.
Initialize the format parameter given to open_stream() in the moved
code. The previous uninitialized value shouldn't have caused any
visible effects.
Remove some unused lines from demuxer.h. Make some demuxer.c functions
static. Move new_ds_stream() declaration from demuxer.h to stream.h
(the function is defined in stream.c). Clean up some code in mplayer.c
that had commented-out free_demuxer_stream() calls.
Convert cache_fill_status into a function so we always get the latest
state, not whatever it was after the last read.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32818 b3059339-0415-0410-9bf9-f77b7e298cf2
Update PAUSED status line with cache fill status if it changed.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32819 b3059339-0415-0410-9bf9-f77b7e298cf2
The "Core dumped ;)" message printed after finishing a stream dump is
known to confuse users but was kept as "humor". Change it to say
"Stream dump complete." instead.
With extreme playback speed changes it was possible to trigger an
overflow in code calculating frame timing. This could break the VDPAU
frame scheduling mechanism and lead to the shown picture not changing
until reset by events such as seeking. Add an extra check to prevent
the overflow.
* sub:
sub/OSD: move some related files to sub/
subtitles: options: enable -ass by default
subtitles: change default libass rendering style
demux_mkv, chapters: change millisecond arithmetic to ns
cleanup: rename ass_* functions to mp_ass_*
subs: use correct font aspect ratio for libass + converted subs
cleanup: some random minor code simplification and cleanup
vf_vo: fix EOSD change detection bug
sd_ass: remove subreader use, support plaintext markup
subtitles: style support for common SubRip tags and MicroDVD
core: ordered chapters: fix bad subtitle parameter
subs/demux: don't try to enable sub track when creating it
subtitles/demux: store duration instead of endpts in demux packets
subtitles: add framework for subtitle decoders
options: add special -leak-report option
subtitles: remove code trying to handle text subs with libavcodec
cleanup: move MP_NOPTS_VALUE definition to mpcommon.h
subtitles: move global ass_track to struct osd_state
core: move most mpcommon.c contents to mplayer.c
core: move global "subdata" and "vo_sub_last" to mpctx
subtitles: remove sub_last_pts hack
options: move -noconfig to option struct, simplify
demux_mkv kept various integer timestamps in millisecond units.
Matroska timestamp arithmetic is however specified in nanoseconds
(even though files typically use 1 ms precision), and using ms units
instead of that only made things more complex. Based on the demux_mkv
example the general demuxer-level chapter structure also used ms
units. Change the demux_mkv arithmetic and demuxer chapter structures
to use nanoseconds instead. This also fixes a seeking problem in
demux_mkv with files using a TimecodeScale other than the usual
1000000 (confusion between ms and TimecodeScale*ns units).
The various ass_* functions were created when libass was part of the
MPlayer tree and the distinction between MPlayer-specific and other
functions was less clear. Now that libass is a clearly separate
library, using the same ass_* namespace for player functions is ugly.
Rename the functions to use mp_ass_ prefix instead.
Rendering of ASS subtitles tries to be bug compatible with VSFilter
and stretches fonts when the video is anamorphic (some scripts try to
compensate for this VSFilter behavior, so trying to render them
"correctly" would give the wrong result). However this behavior is not
appropriate for subtitles we converted to ASS format ourselves for
libass rendering, as they certainly don't have VSFilter bug
workarounds. Change the code to use different behavior for "native"
ASS tracks and converted ones. It's questionable whether the
VSFilter-compatible behavior is appropriate for external .ass files
either, as there could be anamorphic and non-anamorphic versions of
the same video and the bug-compatible behavior can only be correct for
one alternative at most. However it's probably better to keep it as a
default at least, so that extracting a muxed subtitle track and using
that does not give behavior different from the original muxed one.
The aspect ratio setting is per ASS_Renderer, and changing it resets
libass caches. For that reason this commit adds separate renderer
instances to use for the "correct" and "VSFilter bug compatible"
cases.
SubRip subtitles have no "official" spec for any styling support, but
various tags are in common use; previous code filtered out text
between <> to remove HTML-style tags. Add support for those tags and
for MicroDVD subtitle styling. The style display is implemented by
converting the subtitles to the ASS subtitle format and displaying
them with libass, so libass needs to be enabled.
Original patch by Clément Bœsch <ubitux@gmail.com>.
mp_property_do() takes the value to set a property to through a
pointer. The calling code used '&mpctx->global_sub_pos' as the
pointer; however that variable could be changed during the
mp_property_do() call. Use a pointer to a copy of the original value
instead.
I think this only caused problems if you switched subtitle tracks from
a real one to "disabled" and then switched to a timeline part from
another source.
Add a framework for subtitle decoder modules that work more like
audio/video decoders do, and change libass rendering of demuxed
subtitles to use the new framework.
The old subtitle code is messy, with details specific to handling
particular subtitle types spread over high-level code. This should
make it easier to clean things up and fix some bugs/limitations.
Add a special option "-leak-report" that enables talloc leak
reporting. It only works if it's given as the first argument.
The code abuses the CONF_TYPE_PRINT option type to make main option
parsing ignore the option. The parser incorrectly consumed the
following commandline argument as a "parameter" for options of this
type when they had the flag to not exit after printing the message.
Fix this. It makes no difference for any previously existing option I
think.
The avsub implementation tries to fall back to MPlayer's other text
subtitle decoding if libavcodec returns text as the 'decoded'
subtitle. The code implementing this is buggy, and as far as I can see
it should not be triggered normally (libavcodec decoding is only
used for xvid, pgs and dvb subtitles, and for those libavcodec should
return bitmaps). Remove the buggy code (don't try to support
non-bitmap results) and simplify things a bit.
The contents of mpcommon.c were quite arbitrary; the most common
reason to place some functions in this file had been "MEncoder happens
to need similar code as MPlayer and we want to share some parts, but
we have no clue whatsoever how to organize things in a sensible way,
so we'll just dump those parts we want to share in mpcommon.c". As a
result of containing an essentially random subset of top-level player
functionality the mpcommon.h header required access to central structs
and was unsuitable for inclusion in lower-level code, but was
nonetheless included there for the mplayer_version symbol.
Move almost all contents from mpcommon.c to mplayer.c. mplayer.c is
already big and should perhaps be split further, but keeping a few
random functions in mpcommon.c would not be an improvement.
PulseAudio could keep reporting high delay values after a reset of
playing audio. This broke playback after seeking in some cases. Add a
workaround that should make things more robust against such
misbehavior.
Trying to do a framestep while playing an audio-only file would play
the file until the end, then start the next file in paused state. Make
framestep state enter pause again immediately if there is no video.
Also reset framestep state when switching files.
* hr-seek:
input: add default keybindings Shift+[arrow] for small exact seeks
input: support bindings with modifier keys for X input
core: audio: make ogg missing audio timing workaround more complex
core: add support for precise non-keyframe-limited seeks
core: add struct for queued seek info
commands: add generic option -> property wrapper
options: add "choice" option type, use for -pts-association-mode
core: remove looping in update_video(), modify command handling a bit
core: seek: use accurate seek mode with audio-only files
core: avoid using sh_video->pts as "current pts"
libvo: register X11 connection fd in input event system
core: timing: add special handling of long frame intervals
core: move central play loop to a separate function
Conflicts:
DOCS/tech/slave.txt
After the addition of exact seeking the code to work around missing
audio timestamps with ogg/ogm needs improvement. Now it's normal to
need adjustment at stream start time 0 (seeking to a position after
start of video but before second keyframe) with any video format, and
for exact seeks with ogg it's now more important not to skip the
sync. Make the check to detect the problem case more precise to avoid
affecting most other formats, and try to decode a second of audio
(hoping to get timestamps for those packets) before giving up.
Add support for seeking to an arbitrary non-keyframe position by
decoding video starting from the previous keyframe. Whether to use
this functionality when seeking is controlled by the new option
-hr-seek and a new third argument to the "seek" command. The default
is to use it for absolute seeks (like chapter seeks) but not for
relative ones. Because there's currently no support for cutting
encoded audio some desync is expected if encoded audio passthrough is
used. Currently precise seeks always go to the first frame with
timestamp equal to or greater than the target position; there's no
support for "matching or earlier" backwards seeks at frame level.
To prepare for the addition of exact seek support, add a struct for
queued seek state and a helper function to update its state. It would
have been cumbersome to update additional state (showing whether the
seek is forced to be exact or non-exact) manually at every point that
handles seeks.
