demux_mf allocated the "type" suboption of "--mf" with strdup if it
was not explicitly set. This caused a crash after playing an mf://
entry. Fix to use talloc instead.
FFmpeg has increased FF_INPUT_BUFFER_PADDING_SIZE to 16 (unlike Libav
which still has it at 8). Raise MP_INPUT_BUFFER_PADDING_SIZE to 16 to
allow compilation against FFmpeg too (demuxer.c checks the padding
size for packets is at least as much as libavcodec wants for its
decoders, and this check failed with the previous value of 8).
Commit 6e8d420a41 ("demux: avoid a copy of demux packets with lavf,
reduce padding") was missing an av_dup_packet() line. As a result at
least formats that use parsing on the lavf side could fail (with
parsing the packet may contain pointers to temporary fields that
will be overwritten/freed when reading further packets, and
av_dup_packet() is required to allocate permanent storage).
After 0ece360eea ("demux_mkv: skip files faster in ordered chapter
file search") some Matroska files failed to open. The problem was that
demux_mkv_read_info() returned 0 on success, but the opening code
interpreted this as a value to stop parsing further headers. Fix this
and also modify some of the other return value handling.
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.
Export the codec private data field for WavPack and TrueHD audio
tracks. At least for WavPack this is necessary to make some samples
work.
Also change some other cases to use the same data-copying code.
When switching audio or video tracks, demux_mkv only checked that the
new index fell in the range corresponding to tracks existing in the
file being played. However, if the demuxer can not recognize the
format of a track or detects an error, some of those tracks in the
file may not be exported from the demuxer and are not visible to the
rest of the player. Selecting such a track would cause a crash. Add
checks skip such tracks when cycling to next track and switch to
nosound instead if given an explicit track number corresponding to
such a track.
demuxer.c calls demuxer->close() even if opening failed. Thus the
mkv_free() call added in 0ece360eea ("demux_mkv: skip files faster
in ordered chapter file search") was wrong, and could cause a crash
from a double free if some data structures were allocated before the
opening attempt was aborted.
Drop the unnecessary include and add a missing direct include in some
files. This also revealed that demux_rtp_internal.h was missing a
config.h include, fix that too.
Remove unnecessary demuxer.h include from aviheader.h. Through
stheader.h aviheader.h is included in a lot of files. Add missing
mp_msg.h includes to av_sub.c and sd_ass.c (previously hidden by
indirect inclusion through demuxer.h and stream.h).
When demux_lavf read a new packet it used to copy the data from
libavformat's struct AVPacket to struct demux_packet and then free the
lavf packet. Change it to instead keep the AVPacket allocated and
point demux_packet fields to the buffer in that.
Also change MP_INPUT_BUFFER_PADDING_SIZE to 8 which matches
FF_INPUT_BUFFER_PADDING SIZE; demux_lavf packets won't have more
padding now anyway (it was increased from 8 earlier when
FF_INPUT_BUFFER_PADDING_SIZE was increased in libavcodec, but that
change was reverted).
Don't interpret native MPEG codec tags using our generic
format-agnostic codec tag tables. MPEG may use tag 3 for MP3, whereas
the generic tables map 3 to uncompressed PCM. Make the code ignore the
codec_tag field for the "mpeg" and "mpegts" libavformat demuxers and
rely on the codec_id value provided by lavf only.
Ordered chapter code tries opening files to find those matching the
SegmentUID values specified in the timeline. Previously this scan did
a full initialization of the Matroska demuxer for each file, then
checked whether the UID value in the demuxer was a match. Make the
scan code instead provide a list of searched-for UIDs to the demuxer
open code, and make that do a comparison against the list as soon as
it sees the UID in the file, aborting if there is no match.
Also fix units used in "Merging timeline part" verbose message.
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.
If the argument given to demux_lavf audio/video switch code is not one
of -2, -1, or valid audio/video ID the code will treat it the same as
-2 (switch to no sound / no video). However the returned index was not
set to -2 in this case. Fix. Also change the returned index from -1 to
-2 when staying at no sound / video.
Fix bugs in the handling of stream index values in video stream
switching. This is similar to what commit 90bedd0b87
did for audio.
Also clean up the corresponding audio code a little bit.
Disable compilation of demux_ty_osd.c because of its GPL v2-only
license. This only affects TiVo files with -subcc. After this no
v2-only code should get compiled (yuv4mpeg_intern.h has a v2-only
license, but the contents of the header look like they're not
copyrightable).
Drop support for specifying demuxer types by numeric ID (options
-demuxer, -audio-demuxer and -sub-demuxer). Stop printing the numeric
values in "-demuxer help" output. Convert the list of DEMUXER_TYPE_XXX
defines to "enum demuxer_type".
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.
Change DVB SPU stream format in TS demuxer so it can be decoded by
libavcodec (as soon as lavc is fixed not to fail just because of an
extra padding byte).
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32866 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix r32587: the previous approach to return subtitles in time broke
DVB subtitles due to returning incomplete packets and even for
PGS subtitles resulted in incorrect pts values for the sub packets.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32864 b3059339-0415-0410-9bf9-f77b7e298cf2
Do not pointlessly cast the return value of memalign().
memalign() returns void*, which is compatible with any pointer in C.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32850 b3059339-0415-0410-9bf9-f77b7e298cf2
Check that rlen is valid before using it to increment a pointer.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32832 b3059339-0415-0410-9bf9-f77b7e298cf2
af_lavcac3enc: use old SampleFormat names without AV_ prefix, the
latter were only added in 2010-11
vd_ffmpeg: add ifdef around CODEC_ID_LAGARITH use
demux_real: use ffmpeg_files/intreadwrite.h
stream/http.c, stream/realrtsp/real.c: define AV_BASE64_SIZE macro for
old libavutil versions lacking it
Use limits.h to get the maximum length instead of hardcoding it.
Original patch by Sang-Uok Kum.
Signed-off-by: Tobias Diedrich <ranma@google.com>
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32766 b3059339-0415-0410-9bf9-f77b7e298cf2
* 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).
demux_ty relied on demuxer->filepos being initially set to 0, but
demuxer.c has been changed to initialize it to -1. This caused a
"Invalid seek to negative position!" error message when running the
demux_ty file format check (so it occurred for any file which had not
been recognized as another type before that). Fix by making demux_ty
initialize filepos to 0.
When trying to determine the format of an input stream, demux_lavf
retries the probe with a larger buffer size up to some limit if the
match score is low, but when reaching the size limit it accepted the
best match (if any) regardless of its score. Change it to require a
score of at least AVPROBE_SCORE_MAX/4 to accept a match at all.
demuxer.c new_sh_sub_sid() tried to immediately select the created sub
track for playback if its id matched the "-sid" option value. This was
buggy, as more initialization is needed to properly enable subtitles.
Normally the correct track to play is selected after the demuxer has
been created. It's possible that some DVD use case or such depended on
the removed code to make -sid work with a subtitle track that's not
found at start and only added later (vobsubs probably would start
playing without separate initialization); if so then that needs to be
fixed later in a different way.
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.
Change new_demux_packet() and resize_demux_packet() length parameter
type from int to size_t and add a check to abort() if the size is over
1 GB. This should make integer overflow problems leading to memory
corruption in demuxers less likely; and aborting should be no worse
than insane memory consumption. Also make the functions abort() if the
actual allocation fails instead of trying to continue with a
zero-sized buffer.
Make seeks backward from a time before the first index entry go to the
first entry instead of failing completely. This change doesn't affect
behavior for most files, because seeks are clamped to 0 from below and
normally files have the first index entry at 0.
len < 8 is also invalid for 64-bit codec chunk size.
Previous code could cause hang.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32708 b3059339-0415-0410-9bf9-f77b7e298cf2
Avoid using uninitialized data if index read does not return enough data.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32707 b3059339-0415-0410-9bf9-f77b7e298cf2
Always free before overwriting a pointer with a newly allocated one,
always use calloc instead of realloc when the previous data is not
needed.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32703 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace memcpy with memmove since at least src==dst is possible.
Fixes another issue that is part of bug #1280.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32697 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace malloc+memset with calloc and use sizeof(*variable).
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32694 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace hard-coded number for loop limits for array index by
the define used in the array declaration.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32695 b3059339-0415-0410-9bf9-f77b7e298cf2
Add memset to avoid using uninitialized data with sample in bug 1280.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32693 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix mp_check_mp3_header: it checked for a byte-swapped MP3-header
on little-endian, and on big-endian it would only accept a MP3-header
that would be valid when read in both directions.
The latter was the reason for bug 905, causing the PS demuxer to
claim files far too agressively (the MP3 check avoiding misdetection
as DV is not exactly a sane approach, but it mostly works).
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32692 b3059339-0415-0410-9bf9-f77b7e298cf2
Move setup of sh_audio->format to a more appropriate place (in asfheader.c).
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32684 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove pointless assignments that are already handled in new_sh_audio.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32685 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove useless assignment already done in new_sh_video.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32686 b3059339-0415-0410-9bf9-f77b7e298cf2
Use FFMAX for slightly better readability.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32687 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix some unaligned writes and avoid some (incorrect due to alignment) casts.
Might also fix bug #371.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32683 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix TS files with MP4 ES AAC descriptor to be correctly recognized
as AAC and not AAC in LATM.
This fixes playback of http://samples.mplayerhq.hu/A-codecs/AAC/freetv_aac_latm.ts,
actual LATM samples seem unaffected.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32667 b3059339-0415-0410-9bf9-f77b7e298cf2
Fixes:
ffmpeg://rtsp://stream.diffusion.ens.fr/2008_10_03_albarede.mov
and other X-SV3V-ES rtsp streams opened with ffmpeg://
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32660 b3059339-0415-0410-9bf9-f77b7e298cf2
Mark input-only buffers as const.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32652 b3059339-0415-0410-9bf9-f77b7e298cf2
Use uint8_t type instead of unsigned char.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32653 b3059339-0415-0410-9bf9-f77b7e298cf2
Mark input buffer that is never modified as const.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32654 b3059339-0415-0410-9bf9-f77b7e298cf2
Mark input-only buffer as const.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32655 b3059339-0415-0410-9bf9-f77b7e298cf2
Make it seek back to the stream->start_pos position instead of 0 in
that case.
Fixes bug 1790.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32635 b3059339-0415-0410-9bf9-f77b7e298cf2
Use the value of the OutputSamplingFrequency element instead of the
SamplingFrequency element as the "container samplerate". In most cases
this only removes a warning, as those typically differ for SBR AAC
files and there was already a special case detecting this in
ad_ffmpeg.
The implementation adds a new "container_out_samplerate" field to the
sh_audio struct. Reusing the existing "samplerate" field and the
equivalent inside the 'wf' struct and just setting those to the new
value instead would probably work (at least I'm not aware of any codec
that would need the original SamplingFrequency for initialization).
However using a separate field also avoids some ugliness: the 'wf'
struct may not exist (though most demuxers create it), and the
'samplerate' field is overwritten to reflect the final value decided
by codec when decoding is first initialized.
Add definitions for DisplayUnit, OutputSamplingFrequency and
FileDescription in matroska.py. Regenerate the C template files to
allow using all current definitions in code.
Commit 91ea30c585 ("demux_lavf: use lavf for all formats except those
listed") broke handling of files whose type libavformat couldn't
recognize at all. Fix the demux_lavf probe function to correctly
return failure in that case.
Commit 3c2cfee488 ("demux: improve -alang / -slang track choosing
logic") had a copy/paste error which left the subtitle code using
audio variables and broke initial subtitle track selection. Fix.
I actually realized soon after creating the original commit that I'd
forgotten to change the variables and went back to fix it, but then
somehow thought that it was already OK and didn't change it...
Due to a bug created back in 2006 when SimpleBlock support was added,
demux_mkv demuxed one audio packet from the initial file position
after a seek, then skipped the following ones until a video keyframe
was found. This wasn't very noticeable earlier, but it had bad effects
after the recently added -initial-audio-sync code as the extra packet
with an earlier timestamp confused timing calculations and resulted in
desync after seeking. Fix.
Commit fc66c94360 ("demux_mkv: seek: with no track-specific index
entries use any") used uint64_t for a variable that should have been
int64_t. Fix. The practical effects of this error were minor; mainly
it made the player unnecessarily read the file contents between the
previous index entry and the correct one when seeking.
If audio_block_size is 0 that is a bug (and will result in a division by 0
in one case that does not check this), thus remove all checks for it.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32623 b3059339-0415-0410-9bf9-f77b7e298cf2
[ Note: the questionable changes in svn that triggered this problem
were never included in git, and so this commit is not strictly
necessary here. It's included to reduce the differences between git
and svn demux_avi versions. ]
Fix possible division by 0 if -aid is used for AVI files.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32622 b3059339-0415-0410-9bf9-f77b7e298cf2
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.
lavf demuxers are mostly better and receive more maintenance,
therefore it makes sense to prefer them in most cases. Change the
"preferred" logic from listing all formats for which lavf is preferred
to listing exceptions for which it isn't. Currently there are 3
exceptions: Matroska, FLAC and RealMedia (.rm).
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.
"libavformat file format detected" wasn't a very useful message due to
the many file formats supported to libavformat. Change the message so
that for demux_lavf it says something like
"Detected file format: QuickTime/MPEG-4/Motion JPEG 2000 format (libavformat)"
(using long name from FFmpeg), and for non-lavf something like
"Detected file format: Matroska".
The code choosing the demuxer to use only printed an error if given an
unknown demuxer name, then continued with default demuxer selection.
Change it to abort instead. This feels like more sensible behavior.
Also there's no fallback to autodetection in the case where the
demuxer name is recognized but the demuxer fails to open the file
either.
Seeking in MPEG files with pts resets could fail completely, as it was
always done by timestamps and those of course don't unambiguously
specify a file position in such files. Add basic functionality for
byte-based seeking and playback position reporting, and decide whether
to use that functionality based on a simple heuristic (could be
improved).
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.
The Cue entries in typical Matroska files have information for the
video track only. This caused seeks to fail when playing with
-novideo, as demux_mkv tried to use audio track index entries then.
Add a fallback case that uses any index entries without caring what
track they're for if there are no entries specific to the track we're
interested in.
Relative seeks didn't add the current position as they should. Fix.
Note that this had no effect in normal playback case even if the file
had no index, because the "accurate_seek" logic at higher level would
convert all commands to absolute seeks before calling demuxer level.
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.
demux_demuxers doesn't run the normal demuxer.c initialization for new
demuxers. Initialize stream_pts separately (it won't ever be changed
with the current implementation). This at least avoids other code
assuming it was set properly.
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).
Ensure we queue all subtitle packets before demuxing the next video
packet.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32587 b3059339-0415-0410-9bf9-f77b7e298cf2
Uoti Urpala
Clément Bœsch
Anton Khirnov
reimar
diego
cehoyos
ib
Uoti Urpala
ranma
compn
cboesch