| Commit message (Collapse) | Author | Age | Files | Lines |
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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.
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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.
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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.
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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).
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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.
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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.
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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.
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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.
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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.
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* 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
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Sleep 100 ms between filling audio output buffers. Also do the
sleeping in input read functions to enable immediate wakeups on new
input.
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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.
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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
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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.
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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.
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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.
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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.
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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.
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git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32865 b3059339-0415-0410-9bf9-f77b7e298cf2
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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
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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.
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git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32791 b3059339-0415-0410-9bf9-f77b7e298cf2
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git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32751 b3059339-0415-0410-9bf9-f77b7e298cf2
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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.
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* 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
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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).
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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.
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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.
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