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.
Code in get_metadata() allocated too small a buffer for the text it
wrote (noticed by Clément Bœsch). Make the code cleaner and more
robust by changing it to use talloc_asprintf(). Also make it always
return non-NULL and remove checks on caller side.
Let higher-level code call update_video() again instead of looping
inside it until there's a frame ready to show. Change the conditions
for running user commands somewhat. Overall effect shouldn't be that
big. Now other commands can be executed after a seek before a video
frame is decoded; in this case the seek target time may be used as the
"current position".
build_afilter_chain is not safe to use directly, thus make it
static and instead use reinit_audio_chain which should have
better error handling.
Fixes a crash with -af hrtf and changing speed, audio will
still stop playing though.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32648 b3059339-0415-0410-9bf9-f77b7e298cf2
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32630 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix crash on path without directories.
Regression introduced in r32630. Patch by Yuriy Kaminskiy yumkam at mail ru.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32631 b3059339-0415-0410-9bf9-f77b7e298cf2
Handle correctly paths with mixed '/' and '\' in it.
Patch by Yuriy Kaminskiy (yumkam at mail ru)
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32632 b3059339-0415-0410-9bf9-f77b7e298cf2
Handle ':' on systems with DOS paths: it allows paths like C:foo.avi.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32642 b3059339-0415-0410-9bf9-f77b7e298cf2
Add a new field "video_pts" to mpctx. It records the time of the last
frame flipped visible on VO. Change various code which used
sh_video->pts to use either the new field or get_current_time(mpctx).
Add separate handling for the case where the time to flip the next
frame on the VO is more than 50 ms away. In that case don't update OSD
contents yet, but wait for possible changes until 50 ms before the
frame. Sleep until that time in mp_input_get_cmd(), so the sleep is
done in select() and input events can be responded to immediately.
Also raise the limit on audio out delay used to limit sleep.
Some Matroska files have inaccurate ordered chapter endpoints, and so
parts where one chapter should end and the next begin at the same
timestamp were not merged. This resulted in an unnecessary seek over a
minimal distance. Add a heuristic to merge parts with a minimal gap or
overlap between them.
Based on patch by Hector Martin <hector@marcansoft.com>.
ogg/ogm demuxers can give first audio packets without timestamp after
a seek. Due to some backwards compatibility code this results in the
sync code getting audio timestamp 0. In this case a lot of audio was
dropped unnecessarily when seeking to a position later in the file, as
the code saw audio starting from 0, video from something larger.
Make the code more robust in two ways. First, add a special case to
not try syncing if we get audio timestamp <= 0 (hopefully there aren't
many files where we'd really get audio starting from 0 and video from
a later timestamp). Second, when throwing audio away, make the code
recalculate from scratch the amount of bytes that still need to be
thrown away after every decode call. This limits the amount of damage
initial too-small timestamps can do, as the code will see the better
timestamps after a while.
Playing AVI files containing B-frames with demux_lavf printed two
"decreasing pts" info messages at the start of the file. We know the
timestamps from AVI won't be valid pts, so add a demuxer field to
convey that information to the timing code and make that not even try
to use the timestamps as valid pts.
Add code to enforce matching pts with video when (re)starting the
audio stream, by either cutting away the first samples or inserting
silence at the beginning. New option -noinitial-audio-sync can be used
to disable this and return to old behavior.
demuxer_get_current_chapter() accessed sh_video/sh_audio pts fields to
determine playback position. demux layer shouldn't access those and
the values used weren't quite correct anyway. Give the playback
position as a parameter to the demux layer function instead. Also
change the top-level get_current_chapter() to use get_current_time()
in the timeline case where it didn't refer to demux layer.
If the option is enabled and all audio has been buffered to the AO,
then the player will move to the next file without waiting for the
buffered audio to drain, while leaving the AO initialized. If the
playback of the next file starts quickly enough (before the AO buffer
empties) then it should continue writing audio to the same AO with no
gap in between.
At least with PCM it's possible to get an audio stream that doesn't
end at a multiple of whole sample per channel. At least ao_alsa
refuses to accept that part of input, and so EOF detection in
fill_audio_out_buffers didn't trigger until the 0.04 second sanity
check (as there "was still audio not sent to AO left"). Change the
logic to detect EOF if there's less than one sample per channel of
unsent data left.
When file format detection failed the output only said
"Exiting... (End of file)" after "Playing <file>." (or possibly error
messages triggered by format-specific check functions in between). Add
an explicit "Failed to recognize file format." error message.
When -alang / -slang was specified the numerically first matching
track (if any) was always chosen. This meant that specifying "-alang
eng" could change the track choice even if all tracks were in English,
because now the default flag of tracks was ignored. Change the logic
to take the default flag into account as a secondary sorting key.
The code also accepted prefix matches, so that "-slang g" would match
track language "ger". I think that was not intentional. Change it to
require exact matches.
Various code referred to "mpctx->demuxer" where it should really have
referred to the one used for audio/subtitles in case those differ. Fix
by using "mpctx->d_audio->demuxer" etc instead. Disable the copying of
streams in demux_demuxers; that was a partial workaround for things
referring to the main demuxer (and it wasn't enough anyway).
This fixes, among other things, switching audio tracks within the file
specified by -audiofile.
Move functions to query current playback position, percentage position
and total video length from from the demuxer layer to top level. The
functions need access to playback state that doesn't belong on the
demuxing level. Make the new functions more capable and simplify some
code that can now rely on them. This fixes some errors in displayed in
OSD and slave mode information when using timeline (ordered chapters).
Remove cache_fill_status #define.
The #define is only set if CONFIG_STREAM_CACHE is not defined,
but in this case the variable is not used.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32131 b3059339-0415-0410-9bf9-f77b7e298cf2
While convenient, it is too risky.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32025 b3059339-0415-0410-9bf9-f77b7e298cf2
100l, we need to search for SetDllDirectoryA instead of SetDllDirectory
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32026 b3059339-0415-0410-9bf9-f77b7e298cf2
Add support for parameter changes (e.g. channel count) during playback.
This makes decoding AC3 files that switch between 2 and 6 channels
work reasonably well even with -channels 6 and ffac3 decoder.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31737 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix typo in error message: ACC -> AAC
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32473 b3059339-0415-0410-9bf9-f77b7e298cf2
Avoid printing AAC with SBR warning on every decode call, instead print
it only after every decoder reconfiguration.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32476 b3059339-0415-0410-9bf9-f77b7e298cf2
Useful in combination with -fixed-vo and a vo that can do rescaling
during "pause" like gl.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31736 b3059339-0415-0410-9bf9-f77b7e298cf2
Change numbering so that demuxers can "asynchronously" add subtitles
during playback.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31678 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix calculation of global_sub_size.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31711 b3059339-0415-0410-9bf9-f77b7e298cf2
Update global_sub_pos if we auto-selected a subtitle.
This avoid strange behaviour with subtitle selection in that
case, because the subtitle selection code thinks no subtitle
was displayed while we actually did display one.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31863 b3059339-0415-0410-9bf9-f77b7e298cf2
Move function to avoid a forward declaration.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31864 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix subtitle selection: make selecting -sid 0 work again by replacing the
M_PROPERTY_STEP_UP hack by M_PROPERTY_SET and set the global sub pos also
when we do not yet have a subtitle stream but do know the number of subs
from out-of-band as e.g. for DVD.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31882 b3059339-0415-0410-9bf9-f77b7e298cf2
100l, re-add vobsub_get_id_by_index call accidentally removed in
r31678.
Patch by ubitux [ubitux gmail com].
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31985 b3059339-0415-0410-9bf9-f77b7e298cf2
This solution works just as well, but avoids a separate template file
that needs to be #included from multiple places. The end result is
simpler since common options and those of MPlayer and MEncoder are now
in only one file each.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31434 b3059339-0415-0410-9bf9-f77b7e298cf2
SIGPIPE is not really a crash, try to exit cleanly in case we receive one.
Particularly in the network code it would be better to ignore it, but
some window managers use it to indicate a close request.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31432 b3059339-0415-0410-9bf9-f77b7e298cf2
Move the code that prints main per-file -identify info to a separate
function.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31343 b3059339-0415-0410-9bf9-f77b7e298cf2
reasonable.
This avoids completely losing A-V sync e.g. when pts was not reordered correctly.
This was tested with http://samples.mplayerhq.hu/V-codecs/h264/PAFF/tv_cut.mkv
for this sample proper pts reordering is the correct solution, but more resilient
handling of the error case is still useful.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31311 b3059339-0415-0410-9bf9-f77b7e298cf2
Instead of only relying on the MIME type, use the file extension as a
fallback for deciding which attachments are fonts and should be fed to
libass.
This also refactors the check into a separate function in mpcommon.
This fixes subtitles duplication when seeking back in ass stream formated
with the "standard" format FFmpeg uses.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31293 b3059339-0415-0410-9bf9-f77b7e298cf2
insanely large value.
Avoids hang if a decoder handles the data/size == 0 condition incorrectly.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31205 b3059339-0415-0410-9bf9-f77b7e298cf2
Options selecting audio track to play (-nosound, -alang, -aid) only
affected the first source file. When you moved to a timeline part that
came from another file the default audio track for that file was used.
Add code to apply the options to all files.
The option field corresponding to -slang is now called "sub_lang"
instead of the old misleading global name "dvdsub_lang". The code
handling -slang in subreader.c looks rather broken; disable it instead
of converting it to use the option field.
Make the number of output surfaces allocated by vo_vdpau configurable
and allow queuing multiple future frames with the VDPAU frame timing
functionality. There are two known issues that could be polished in
the future for particularly long queue time cases. First, toplevel
code should possibly wait for the queued changes to finish displaying
in some situations like switching files. Second, the VO can block
waiting for a surface to become available, and in the worst case this
wait can be almost as long as the maximum queuing time (potentially
causing audio underruns). However those issues are not high priority.
Due to inexact seeks, chapter seek commands may result in a playback
position that's inside the previous chapter. This causes problems when
the user does repeated next-chapter/previous-chapter seeks, because
the code will use the wrong base for calculating 'next' or
'previous'. Improve the behavior by adding a heuristic that keeps
track of what chapter the user last wanted to seek to and adjusts the
"current chapter" value based on that.
The code processing seek commands only sets/alters variables
specifying the current seek target. Before all queued commands were
processed first, and any needed seeks were executed after that. This
was somewhat unreliable, as commands queued later were not guaranteed
to see all the effects of earlier seek commands if they happened to be
processed in the same batch. Change the behavior so that processing
commands is interrupted and the real seek executed if the next command
is anything other than a basic seek. This guarantees that other
commands see a consistent state, while still allowing the combining of
consecutive seeks (which is useful for example when the user keeps the
seek-forward key pressed down, and key repeat is faster than seeks can
be executed).
Before "-chapter 1" did nothing even if the first chapter didn't start
at the beginning of file. Fix it.
Before all chapter property commands (including chapter seek keys)
failed if the current playback position was before the start of the
first chapter. Now they'll work. Relative chapter seeks will go to the
first chapter (even if that's in the wrong direction for backward
seeks).
-chapter can optionally take a range with a start and an end. Add a
new option type which supports such values and use that instead of a
custom per-option function.
This commit also fixes a build configuration bug: before the
availability of the -chapter option depended on DVD functionality
being enabled in the binary, even though the option works with other
sources too.
Move code resetting various things after a seek into a separate
function and use that for chapter seeks too. In most cases this won't
change behavior because chapter seeks were already falling back to the
same time-based seek code.
Move code for resetting decoders after seeks, chapter seeks and angle
changes out of demuxer.c. This functionality belongs on a higher
level; the demux layer can't always know what kind of reinitialization
is required.
This keeps the naming of all path-related functions consistent.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@30944 b3059339-0415-0410-9bf9-f77b7e298cf2
These files now contain different functions related to path handling.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@30943 b3059339-0415-0410-9bf9-f77b7e298cf2
This fixes compilation with the Win32 loader disabled but other binary
codec loaders enabled.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@30942 b3059339-0415-0410-9bf9-f77b7e298cf2
