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).
Recent commit 5d5ca22a6d ("options: commandline: accept --foo=xyz
style options") left some bad code under "#ifdef MP_DEBUG" in
playtree.c, which caused a compilation failure if configured with
"--enable-debug". Fix this. Having the "#ifdef MP_DEBUG" there was
completely unnecessary; it only increased the risk for this kind of
problems for no real benefit - executing the asserts under it would
have no noticeable performance or other penalty in default builds
either. Remove several cases of such harmful "#ifdef MP_DEBUG".
Rename the BSTR() function to bstr(). The former caused a conflict
with some Windows OS name, and it's no longer a macro so uppercase
naming is less appropriate.
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.
Some versions of lavf abuse codec_tag for passing Bink version
information to the decoder, which broke detection based on codec tag
(though this has already stopped again in latest Libav). Move bink
audio codec IDs from mp_wav_tags to mp_codecid_override_tags so that
codec tags are completely ignored for them.
Setting AVIOContext for AVFMT_NOFILE formats now triggers a warning
from libavformat (and triggered an error for a while), so add a check
to avoid setting AVIOContext when not necessary.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33695 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix printing of subtitle type, the wrong index was used to look up the
type.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33664 b3059339-0415-0410-9bf9-f77b7e298cf2
Acording to the ASF documentation, the play duration is zero
if the preroll value is greater than the play duration.
The new way of determining it (suggested by reimar) prevents
overflows as well.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33492 b3059339-0415-0410-9bf9-f77b7e298cf2
According to the ASF documentation,
MF_PD_ASF_FILEPROPERTIES_PREROLL (preroll) is UINT64. Fix type
mentioned in comment.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33484 b3059339-0415-0410-9bf9-f77b7e298cf2
If the played file has per-track titles for audio and subtitles show
those on the OSD when switching tracks. This changes the OSD message
from 'Audio: (2) eng' to 'Audio: (2) eng ("Director's commentary")'.
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.
Selecting the colorspace to output from a decoder is done in the
function mpcodecs_config_vo(). Add a new version of this function,
mpcodecs_config_vo2(), that allows the decoder to specify a list of
candidate colorspaces instead of always using a hardcoded list
specified in the codecs.conf entry. If the codecs.conf entry has any
"out" lines then those still take priority and the decoder-provided
list (if any) is ignored. Make vd_ffmpeg provide a list of the
colorspaces it's willing to output. Remove "out" lines from most
entries for libavcodec video decoders in codecs.conf, so that the
automatic values are now used instead.
sd_ass relies on there being a zero byte after packet data. However
the packet allocation routines special-cased data length 0 and left
the data pointer as NULL in that case. This could cause a crash in
sd_ass if there was an empty subtitle packet. Change the allocation
routines to stop special-casing empty data and always allocate
padding. Empty packets are not so common that special casing them
would be a worthwhile optimization.
Also fix resize_demux_packet() to use MP_INPUT_BUFFER_PADDING SIZE as
the padding size, instead of a hardcoded value of 8.
Update various code to use newer alternatives instead of deprecated
functions/fields that are being dropped at libav API bump. An
exception is avcodec_thread_init() which is being dropped even though
it's still _necessary_ with fairly recent libav versions, so there's
no good alternative which would work with both those recent versions
and latest libavcodec. I think there are grounds to consider the drop
premature and revert it for now; if that doesn't happen I'll add a
version-test #if check around it later.
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 the various decoders to codecs.conf and increase the maximum
number of buffered pts in stheader.h (apparently CrystalHD can have
very high decoder lag).
Patch by Philip Langdale, philipl overt org
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33095 b3059339-0415-0410-9bf9-f77b7e298cf2
Libavcodec has no parser that would work on byte-swapped AC3, but at
least don't run the normal AC-3 one which would only break things.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33026 b3059339-0415-0410-9bf9-f77b7e298cf2
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33027 b3059339-0415-0410-9bf9-f77b7e298cf2
Support audio in Leitch/Harris' VR native stream format (LXF).
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32990 b3059339-0415-0410-9bf9-f77b7e298cf2
Support dvvideo in Leitch/Harris' VR native stream format (LXF).
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32991 b3059339-0415-0410-9bf9-f77b7e298cf2
* 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
libavformat returns nonsense per-stream bitrate values for some MPEG
files (0 or many times higher than the overall bitrate of the file),
which triggered the heuristic to enable byte-based seeking in
demux_lavf and then made the byte-based seeks wildly inaccurate.
Disable the support for byte-based seeks. This will avoid problems
with files that have consistent timestamps, but on the other hand will
completely break seeking in MPEG files that have timestamp resets.
I'll probably add at least an option to manually enable byte-based
seeking 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
Remove code that tries to select audio track during demuxer
initialization from demux_mkv and demux_lavf. Just leave audio
disabled at that point; the higher-level select_audio() function will
call the demuxer to switch track later anyway.
Removing this unneeded code also fixes use of these demuxers as the
main demuxer with -audiofile. Before the automatic track selection
would have enabled an audio track (if the file had any); as the main
demuxer was not used for audio the unused packets from this enabled
track would accumulate until they reached queue size limits.
Commit de42015a97 ("demux_mkv: read tags") added code that
failed to initialize a loop variable. Fix. No visible problems caused
by the bug have been reported.
Duration may now be set for packet types other than subtitles; as far
as I can tell nothing should care. A check requiring valid duration
values for subtitles is removed, because duration may not be properly
set for all bitmap subtitle types; hopefully this doesn't make the
behavior with (already broken) subtitles without duration worse.
In 59058b54a7 (from svn r31129) Aurelien
changed demux_lavf -vid indexing, but failed to change the initial
video stream selection based on -vid to match. Fix.
