mp3 has a hack lowering the probescore for format detection. This is
because detecting mp3s is hard due to their nature, and the fact that
ID3v2 tags are sometimes several megabytes big.
When playing mp3 from network, the mime-type is usually set, and that
matches the format hack entry meant for webradios, overriding the normal
mp3 entry. This can lead to network mp3s not being detected. Lower the
network case to the same probescore as on-disk mp3s. The difference is
that for network mp3s, we don't load the full probe-buffer, and we lower
the amount of audio the demuxer will read to collect data on opening
(0.5 seconds instead of typically 5 seconds).
Also remove MSGL_SMODE and friends.
Note: The indent in options.rst was added to work around a bug in
ReportLab that causes the PDF manual build to fail.
VP9 packets can contain 2 frames in some video packets (from which 1
frame is invisible). Due to a design mismatch between libvpx and the
libavcodec vp9 decoder, libvpx can take the "full" packets, but lavc vp9
can not. The consequence is that we have to split the packets if we want
to feed them to the lavc codec.
This is not entirely correct yet: timestamp handling is missing.
--demuxer=lavf and ffmpeg native utilities have the same problem. We can
fix this only once the ffmpeg VP9 parser is fixed.
For some reason, some files appear to have broken mp3 packets, or at
least in a form that libavcodec can't deal with. The audio in the sample
file in question could not be decoded using libavcodec.
The problematic file had variable packet sizes, and the libavcodec
decoder kept printing "mp3: Header missing" for each packet it was fed.
Remuxing with mkvmerge fixes the problem. The mp3 data is probably not
VBR, and remuxing resulted in fixed-size mp3 frames. So I don't know why
the sample file was muxed this way - it might just be incorrect.
The sample file had "libmkv 0.6.4" as MuxingApp (although I could not
get mkvinfo to print this element, maybe the file uses an incorrect
element ID), and "HandBrake 0.9.4" as WritingApp.
Note that the libmpg123 decoder does not have any issues with it. It's
probably more robust, because libmpg123 was made to decode whole mp3
files, not just single frames.
Fixes issue #742.
glob() is mandated by POSIX. For the only non-POSIX platform we support,
Windows, we have our own replacement. So the ifdeffery is not needed.
Still leave the checks in the configure scripts, because they have to
decide whether to compile the replacement or not. (Although this could
be special cased to mingw-only, the wscript seems to make this hard.)
glob-win.c wasn't big, so it was easier to rewrite it. The new version
supports Unicode, handles directories properly, sorts the output and
puts all its allocations in the same talloc context to simplify the
implementation of globfree.
Notably, the old glob had error checking code, but didn't do anything
with the errors since the error reporting code was commented out. The
new glob doesn't copy this behaviour. It just treats errors as if there
were no more matching files, which shouldn't matter for mpv, since it
ignores glob errors too.
To match the other Windows I/O helper functions, the definition is moved
to osdep/io.h.
I hate tabs.
This replaces all tabs in all source files with spaces. The only
exception is old-makefile. The replacement was made by running the
GNU coreutils "expand" command on every file. Since the replacement was
automatic, it's possible that some formatting was destroyed (but perhaps
only if it was assuming that the end of a tab does not correspond to
aligning the end to multiples of 8 spaces).
This was broken at some unknown point (even before the recent cache
changes). There are several problems:
- stream_dvd returning a random stream position, confusing the cache
layer (cached data and stream data lost their 1:1 corrospondence by
position)
- this also confused the mechanism added with commit a9671524, which
basically triggered random seeking (although this was not the only
problem)
- demux_lavf requesting seeks in the stream layer, which resulted in
seeks in the cache or the real stream
Fix this by completely removing byte-based seeking from stream_dvd. This
already works fine for stream_dvdnav and stream_bluray. Now all these
streams do time-based seeks, and pretend to be infinite streams of data,
and the rest of the player simply doesn't care about the stream byte
positions.
Instead, always use the mpctx->chapters array. Before this commit, this
array was used only for ordered chapters and such, but now it's always
populated if there are chapters.
Stream-level chapters (like DVD etc.) did potentially not have
timestamps for each chapter, so STREAM_CTRL_SEEK_TO_CHAPTER and
STREAM_CTRL_GET_CURRENT_CHAPTER were needed to navigate chapters. We've
switched everything to use timestamps and that seems to work, so we can
simplify the code and remove this old mechanism.
Instead of parsing the ASS file in demux_libass.c and trying to pass the
ASS_Track to the subtitle renderer, just read all file data in
demux_libass.c, and let the subtitle renderer pass the file contents to
ass_process_codec_private(). (This happens to parse full files too.)
Makes the code simpler, though it also relies harder on the (messy)
probe logic in demux_libass.c.
The mplayer decoder (spudec.c) actually handled this. There was explicit
code for binary palettes (16 32 bit values), and the subtitle resolution
was handled by video resolution coincidentally matching the subtitle
resolution.
Whoever puts vobsub into mp4 should be punished.
Fixes the sample gundam_sample.mp4, closes github issue #547.
This skipped all audio packets before the first video key frame was
found. I'm not really sure why this would be needed; most likely it
isn't. So get rid of it. Even if audio packets are returned to the
player too soon, the player will sync the audio start to the video
start by decoding and discarding audio data.
Note that although the removed code was just added in the previous
commit, it merely kept the old keeping semantics which demux_mkv
always followed. This commit removes these special semantics.
v_skip_to_keyframe is set to true while non-keyframe video packets are
skipped. Until now, audio packets were also skipped when doing this. I
can't see any good reason why this would be done, but for now I want to
keep the old logic when audio+video seeks are done.
However, for audio-only mode, do proper seeking, which also fixes
behavior when trying to seek past the end of the file: playback is
terminated properly, instead of starting playback on the start of the
last cluster.
Note that a_no_timecode_check is used only for audio+video seek. I'm
not sure what this is needed for, but it might influence A/V sync after
seeking.
Trying to set a non-existent flag (like +keepside on Libav) causes
libavutil print an incomprehensible warning (something about eval;
probably the overengineered libavutil option parser tripping over the
'+' normally used for flags, and trying to interpret it as formula).
There's apparently no easy way to check for the existence of a flag,
so add some more ifdeffery to shut it up.
There is some logic to discard packets from streams that are not
selected. Run the metadata update code before this, just to make 100%
sure that no metadata updates can be lost when streams are deselected.
(I'm not sure why this logic would be needed, since both libavformat and
the generic demuxer code do this already. But a quick test shows that
av_read_frame() can return a packet from a stream even if the stream has
AVStream.discard set to AVDISCARD_ALL. This happened after stream
switching. Maybe libavformat doesn't discard already queued packets.)
Instead of printing lines like:
Demuxer info GENRE changed to Alternative Rock
Just output all tags once they change. The assumption is that individual
tags rarely change, while all tags change in the common case.
This changes tag updates to use polling. This could be fixed later,
although the ICY stuff makes it a bit painful, so maybe it will remain
this way.
Also remove DEMUXER_CTRL_UPDATE_INFO. This was intended to check for tag
updates, but now we use a different approach.
Use an arbitrary constant instead, which is as good as PATH_MAX.
This helps us to avoid having to think about pull request #523.
Also fix a case where a potentially signed char was passed to isspace().
This used to work; I'm not sure when or why it regressed. When setting
AVProbeData.filename to NULL, libavformat will crash in rtp_probe() by
unconditionally accessing the string.
We used to set the filename to NULL to prevent probing by file extension
when we don't deem it as necessary. Using an empty string also works for
this purpose.
This generally affects mp3 files that don't have any (or many) mp3
frames in the first 2 MB. 2 MB is the maximum probe size, and
libavformat returns a low probescore even if we give it the full 2 MB.
Trying to probe a larger buffer (or even the full file) doesn't work for
mysterious reasons.
The workaround consists in accepting a very weak probescore if the
format is detected as mp3 and we probed already 2 MB.
Restructure it a bit, so we can use the format hack list even if no mime
type applies. Shouldn't change anything functionally yet. Preparation
for the next commit.
If there's more than one edition, print the list of editions, including
the edition name, whether the edition is selected, whether the edition
is default, and the command line option to select the edition. (Similar
to stream list.)
Move reading the tags to a separate function process_tags(), which is
called when all other state is parsed. Otherwise, that tags will be lost
if chapters are read after the tags.
Pretty worthless. This is called from the seek code, which will
reinitialize these anyway. Even if seeking somehow decides to fail, the
new values are still valid.
One could say a failed seek (if that happens) should jump back to the
original position, and thus it would be better to make sure the state
is restored. But then demux_mkv_seek needs to do this correctly,
including not setting up skipping to the target timestamp. But not
bothering with this.
Extremely obscure corner case with concatenated segments, in which EOF
wasn't recognized correctly, and it tried to demux clusters from the
next segment.
See [MKV]_Editions,_Linked_Segments,_&_Tracksets.mkv from the CCCP test
file collection.
This basically used to be part of the user interface, before mpv moved
printing the track list to the frontend, and this code was raised to
verbose output level.
For some reason, if an error happened when reading headers, it merely
stopped reading the headers, and then continued normally. (It looks like
the case to exit hard (-2) was mainly used for skipping unwanted ordered
chapter segments.)
I can't comprehend this. Always exit on error when reading headers.
(Maybe some more error tolerance would be good, but I have no test case,
and there's some danger of entering endless loops.)
This makes everything more robust, and also somewhat simpler (even if
the diffstat isn't very impressive).
Instead of recursively following SeekHeads while reading headers, just
read the headers until the first cluster, and then possibly use
SeekHeads to read the remaining missing headers.
As of this commit, stream_read_line() can't actually error (except in
the case the passed in buffer is 0, which never happens here). This
commit is preparation for the following commit, which checks harder
whether the read data is actually text. Before this commit, an error
was treated as end-of-file, but the data read so far was considered
valid.
Many ebml_read_* functions have a length int pointer parameter, which
returns the number of bytes skipped. Nothing actually needed this
(anymore), and code using it was rather hard to understand, so get rid
of them.
Matroska makes it pretty hard to resync correctly on broken files:
random data returns "valid" EBML IDs with a high probability, and when
trying to skip them it's likely that you skip a random amount of data
(instead of considering the element length invalid).
Improve upon this by skipping known level 1 elements only. Consider
everything else invalid and call the resync code. This might result in
annoying behavior when Matroska adds new level 1 elements, although it
won't be particularly harmful. Matroska doesn't really allow us to do
better (even mkvtoolnix explicitly checks for known level 1 elements).
Since we now don't always want to combine EBML element skipping and
resyncing, remove ebml_read_skip_or_resync_cluster(), and make
ebml_read_skip() more tolerant against skipping broken elements.
Also, don't resync when reading sub-elements, and instead do resyncing
when reading them results in an error.
Until now, corrupted files were detected if the size of an element (that
should be skipped) was larger than the remaining file. This still could
skip larger regions of the file itself if the broken size happened to be
within the file.
Change it so that it's never allowed to skip outside the parent's
element.
MicroDVD files _can_ contain real timestamps instead of frame timestamps
if they declare a FPS. But this seems to be rare, so ignore that if the
FPS happens to match with the libavformat microdvd parser's default FPS.
This might actually break files that declare 23.976 FPS, but the video
file is not 23.976 FPS, but the chance that this happens is probably
very low, and the commit fixes the more common breakage with 25 FPS
video.
demux_subreader.c contains the old MPlayer subtitle parser, and I have
absolutely no confidence in this (very crappy) code. There might be
one or two security risks associated with running that code on
arbitrary input.
Apparently, Matroska packs TrueHD packets in a way lavc doesn't expect.
This broke decoding of some files [1] completely. A short look at the
libavcodec parser shows that parsing this ourselves would probably be
too much work, so make use of the libavcodec parser API.
[1] http://www.cccp-project.net/beta/test_files/mzero_truehd_sample.mkv
The TV code pretends to be part of stream/, but it's actually demuxer
code too. The audio_in code is shared between the TV code and
stream_radio.c, so stream_radio.c needs a small hack until stream.c is
converted.
Since m_option.h and options.h are extremely often included, a lot of
files have to be changed.
Moving path.c/h to options/ is a bit questionable, but since this is
mainly about access to config files (which are also handled in
options/), it's probably ok.
The tmsg stuff was for the internal gettext() based translation system,
which nobody ever attempted to use and thus was removed. mp_gtext() and
set_osd_tmsg() were also for this.
mp_dbg was once enabled in debug mode only, but since we have log level
for enabling debug messages, it seems utterly useless.
The end of the current segment will be the end of the file if there is
no next segment. Normally, this didn't matter much, since UNIX files
allow seeking past the end of the file. But when opening files from
HTTP, this would print confusing error messages. So explicitly check for
EOF before trying to read a segment.
This readds a more or less completely new dvdnav implementation, though
it's based on the code from before commit 41fbcee. Note that this is
rather basic, and might be broken or not quite usable in many cases.
Most importantly, navigation highlights are not correctly implemented.
This would require changes in the FFmpeg dvdsub decoder (to apply a
different internal CLUT), so supporting it is not really possible right
now. And in fact, I don't think I ever want to support it, because it's
a very small gain for a lot of work. Instead, mpv will display fake
highlights, which are an approximate bounding box around the real
highlights.
Some things like mouse input or switching audio/subtitles stream using
the dvdnav VM are not supported.
Might be quite fragile on transitions: if dvdnav initiates a transition,
and doesn't give us enough mpeg data to initialize video playback, the
player will just quit.
This is added only because some users seem to want it. I don't intend to
make mpv a good DVD player, so the very basic minimum will have to do.
How about you just convert your DVD to proper video files?
There are 3 users of the image format option type: demux_raw,
vf_format, vf_noformat. Allow the hwaccel formats (like vdpau etc.)
in general, so that the filters can use it. This won't work for
demux_raw, so explicitly reject these formats there.
So, FFmpeg/Libav requires us to figure out video timestamps ourselves
(see last 10 commits or so), but the methods it provides for this aren't
even sufficient. In particular, everything that uses AVI-style DTS (avi,
vfw-muxed mkv, possibly mpeg4-in-ogm) with a codec that has an internal
frame delay is broken. In this case, libavcodec will shift the packet-
to-image correspondence by the codec delay, meaning that with a delay=1,
the first AVFrame.pkt_dts is not 0, but that of the second packet. All
timestamps will appear shifted. The start time (e.g. the time displayed
when doing "mpv file.avi --pause") will not be exactly 0.
(According to Libav developers, this is how it's supposed to work; just
that the first DTS values are normally negative with formats that use
DTS "properly". Who cares if it doesn't work at all with very common
video formats? There's no indication that they'll fix this soon,
either. An elegant workaround is missing too.)
Add a hack to re-enable the old PTS code for AVI and vfw-muxed MKV.
Since these timestamps are not reorderd, we wouldn't need to sort them,
but it's less code this way (and possibly more robust, should a demuxer
unexpectedly output PTS).
The original intention of all the timestamp changes recently was
actually to get rid of demuxer-specific hacks and the old timestamp
sorting code, but it looks like this didn't work out. Yet another case
where trying to replace native MPlayer functionality with FFmpeg/Libav
led to disadvantages and bugs. (Note that the old PTS sorting code
doesn't and can't handle frame dropping correctly, though.)
Bug reports:
https://trac.ffmpeg.org/ticket/3178https://bugzilla.libav.org/show_bug.cgi?id=600
This was broken by the recent commits. Apparently realvideo timestamps
are severely mangled, and Matroska _of course_ doesn't have the sane,
umangled timestamps, but something unusable. The existing unmangling
code in demux_mkv.c didn't output proper timestamps either. Instead,
it was something weird that triggered sorting. Without sorting (it was
disabled by default recently), you'd get decreasing PTS warnings
In order to fix this, steal some code from libavcodec. Basically copy
the contents of rv34_parser.c (with some changes), which makes
everything magically work. (Maybe it would be better to use the
libavcodec parser API, but I don't want to do that just for this. An
alternative idea would be refusing to read files that have realvideo
tracks, and delegate this to demux_lavf.c, but maybe that's too redical
too.)
I wish I hadn't notice this...
These packets have to be explicitly dropped, because usually libavcodec
uses 0-sized packets to flush delayed frames, meaning just passing
through these packets would have bad consequences.
Normally, libavformat doesn't output 0-sized packets anyway. But I don't
want to take any chances, so don't delete it, and just move it out of
the way to demux.c.
It appears PTS sorting was useful only for avi files (and VfW-muxed
mkv). Maybe it was historically also important for decoders with broken
or non-existent PTS reordering (win32 codecs?). But now that we handle
demuxers which outputs DTS only correctly, it just seems dead weight.
Disable it by default. The --pts-association-mode option is now forced
to always use the decoder's PTS value. You can still enable the old
default (auto) or force sorting. But we will probably remove this option
entirely at some point.
Make demux_mkv export timestamps at DTS when it's in VfW mode. This is
needed to get correct timestamps with the new default mode. demux_lavf
already does that.
This was needed to determine PTS from DTS, but the previous commits
make it unnecessary.
The builtin genpts hack was used for DVD, because libavformat's genpts
essentially went amok on DVD timestamp resets. See commit 65d87091 for
details.
Having the DTS directly can be useful for restoring PTS values.
The avi file format doesn't actually store PTS values, just DTS. An
older hack explicitly exported the DTS as PTS (ignoring the [I assume]
genpts generated non-sense PTS), which is not necessary anymore due to
this change.
This used to be needed to access the generic stream header from the
specific headers, which in turn was needed because the decoders had
access only to the specific headers. This is not the case anymore, so
this can finally be removed again.
Also move the "format" field from the specific headers to sh_stream.
This is similar to the sh_audio commit.
This is mostly cosmetic in nature, except that it also adds automatical
freeing of the decoder driver's state struct (which was in
sh_video->context, now in dec_video->priv).
Also remove all the stheader.h fields that are not needed anymore.
sh_audio is supposed to contain file headers, not whatever was decoded.
Fix this, and write the decoded format to separate fields in the decoder
context, the dec_audio.decoded field. (Note that this field is really
only needed to communicate the audio format from decoder driver to the
generic code, so no other code accesses it.)
Move all state that basically changes during decoding or is needed in
order to manage decoding itself into a new struct (dec_audio).
sh_audio (defined in stheader.h) is supposed to be the audio stream
header. This should reflect the file headers for the stream. Putting the
decoder context there is strange design, to say the least.
This was forgotten when the parser for mplayer2 EDL files was removed.
Change the header of the mpv EDL format to include a '#', so a naive
parser could skip the header as comment. (Maybe this is questionable;
on the other hand, if it can be simpler, why not.)
Also, strip the header in demux_edl.c before passing on the data, so the
header check doesn't need to be duplicated in tl_mpv_edl.c.
Edit Decision Lists (EDL) allow combining parts from multiple source
files into one virtual file. MPlayer had an EDL format (which sucked),
which mplayer2 tried to improve with its own format (which sucked). As
logic demands, mpv introduces its very own format (which sucks).
The new format should actually be much simpler and easier to use, and
its implementation is simpler and smaller too.
demuxer->filepos contains the byte offset of the last read packet. This
is so that the player can estimate the current playback position, if no
proper timestamps are available. Simplify it to use demux_packet->pos in
the generic demuxer code, instead of bothering every demuxer
implementation about it.
(Note that this is still a bit incorrect: it relfects the position of
the last packet read by the demuxer, not that returned to the user. But
that was already broken, and is not that trivial to fix.)
This was originally added for better seeking where libavformat's seek
function won't work well: files with timestamp resets. In these cases,
the code tried to calculate an average bitrate, and then do byte based
seeks by multiplying the seek target time with the bitrate.
Apparently this was unreliable enough that the code was just commented
(and other parts became inactive). Get rid of it.
Note that the player still does byte based seeks in these cases when
doing percent-seeks.
Slightly simplifies memory management. This might make adding a demuxer
cache wrapper easier at a later point, because you can just copy the
complete stream header, without worrying that the wrapper will free the
individual stream header fields.
Most libavcodec decoders output non-interleaved audio. Add direct
support for this, and remove the hack that repacked non-interleaved
audio back to packed audio.
Remove the minlen argument from the decoder callback. Instead of
forcing every decoder to have its own decode loop to fill the buffer
until minlen is reached, leave this to the caller. So if a decoder
doesn't return enough data, it's simply called again. (In future, I
even want to change it so that decoders don't read packets directly,
but instead the caller has to pass packets to the decoders. This fits
well with this change, because now the decoder callback typically
decodes at most one packet.)
ad_mpg123.c receives some heavy refactoring. The main problem is that
it wanted to handle format changes when there was no data in the decode
output buffer yet. This sounds reasonable, but actually it would write
data into a buffer prepared for old data, since the caller doesn't know
about the format change yet. (I.e. the best place for a format change
would be _after_ writing the last sample to the output buffer.) It's
possible that this code was not perfectly sane before this commit,
and perhaps lost one frame of data after a format change, but I didn't
confirm this. Trying to fix this, I ended up rewriting the decoding
and also the probing.
This affects 64 bit floats and big endian integer PCM variants
(basically crap nobody uses). Possibly not all MS-muxed files work, but
I couldn't get or produce any samples.
Remove a bunch of format tags that are not needed anymore. Most of these
were used by demux_mov, which is long gone. Repurpose/abuse 'twos' as
mpv-internal tag for dealing with the PCM variants mentioned above.
This member was redundant. sh_audio->sample_format indicates the sample
size already.
The TV code is a bit strange: the redundant sample size was part of the
internal TV interface. Assume it's really redundant and not something
else. The PCM decoder ignores the sample size anyway.
Somehow the new parser ends up much smaller. Much of it is because we
don't parse some additional information. We just skip it, instead of
parsing it and then throwing it away.
More importantly, we use the physical order of entries, instead of
trying to sort them by entry number. Each "File" entry is followed by a
number that is supposed to be the entry number, and "File1" is first.
(Should it turn out that this is really needed, an additional field
should be added to playlist_entry, and then qsort().)
Make TOOLS/matroska.pl output structs with fields sorted by name in
ebml_types.h to make the order of fields deterministic. Fix warnings in
demux_mkv.c caused by the first struct fields switching between scalar
and struct types due to non-deterministic ebml_types.h field order.
Since it's deterministic now, this shouldn't change anymore.
The warnings produced by the compilers are bogus, but we want to silence
them anyway, since this could make developers overlook legitimate
warnings.
What commits 7b52ba8, 6dd97cc, 4aae1ff were supposed to fix. An earlier
attempt sorted fields in the generated C source file, not the header
file. Hopefully this is the last commit concerning this issue...
The configure followed 5 different convetions of defines because the next guy
always wanted to introduce a new better way to uniform it[1]. For an
hypothetic feature 'hurr' you could have had:
* #define HAVE_HURR 1 / #undef HAVE_DURR
* #define HAVE_HURR / #undef HAVE_DURR
* #define CONFIG_HURR 1 / #undef CONFIG_DURR
* #define HAVE_HURR 1 / #define HAVE_DURR 0
* #define CONFIG_HURR 1 / #define CONFIG_DURR 0
All is now uniform and uses:
* #define HAVE_HURR 1
* #define HAVE_DURR 0
We like definining to 0 as opposed to `undef` bcause it can help spot typos
and is very helpful when doing big reorganizations in the code.
[1]: http://xkcd.com/927/ related
Instead of having each demuxer do it (only demux_mkv actually did...),
let generic code determine whether the file is seekable. This requires
adding exceptions to demuxers where the stream is not seekable, but the
demuxer is.
Sort-of try to improve handling of unseekable files in the player. Exit
early if the file is determined to be unseekable, instead of resetting
all decoders and then performing a pointless seek.
Add an exception to allow seeking if the file is not seekable, but the
stream cache is enabled. Print a warning in this case, because seeking
outside the cache (which we can't prevent since the demuxer is not aware
of this problem) still messes everything up.
Pointless, using stream->start_pos/end_pos instead.
demux_mf was the only place where this was used specially, but we can
rely on timestamps instead for this case.
There are some Microsoft Windows symbols which are traditionally used by
the mplayer core, because it used to be convenient (avi was the big
format, using binary windows decoders made sense...). So these symbols
have the exact same definition as the Windows one, and if mplayer is
compiled on Windows, the symbols from windows.h are used.
This broke recently just because some files were shuffled around, and
the symbols defined in ms_hdr.h collided with windows.h ones. Since we
don't have windows binary decoders anymore, there's not the slightest
reason our symbols should have the same names. Rename them to reduce the
risk for collision, and to fix the recent regression.
Drop WAVEFORMATEXTENSIBLE, because it's mostly unused. ao_dsound defines
its own version if the windows headers don't define it, and ao_wasapi is
not available on systems where this symbol is missing.
Also reindent ms_hdr.h.
Now that matroska.pl generates struct fields in deterministic order,
this should be the last time I change this.
(gcc and clang shouldn't warn about this line of code, but since they
do, we want to workaround and silence the warning anyway.)
Unfortunately, we can't avoid this warning 100%, because ebml_info is
written by a Perl script. I think the script writes the struct fields in
random order (thanks Perl), so there's no way to know whether the first
struct field is a scalar or a struct.
At least {0} is always valid here, even if it shows a warning. (The
compilers are wrong, see e.g. [1].)
[1] http://gcc.gnu.org/bugzilla/show_bug.cgi?id=53119
This one really did bite me hard (see previous commit), so enable it by
default.
Fix some cases of shadowing throughout the codebase. None of these
change behavior, and all of these were correct code, and just tripped up
the warning.
gcc and clang happen to allow {} to default-initialize a struct, but
strictly speaking, C99 requires at least {0}. In one case, we use {{0}},
but that's only because gcc as well as clang are too damn stupid not
to warn about {0}, which is a perfectly valid construct in this case.
(Sure is funny, don't warn about the non-standard case, but warn about
another standard conform case.)
Leaving these braces away just because the syntax allows them is really
obnoxious. It removes the visual cues which help understanding the code
at the first look.
For the record,
if (cond)
something();
is ok, as long as there's no else branch, and the if body is one
physical line. But everything else should have braces.
This was probably not a real problem. But it's not entirely clear
whether this could actually happen or not, so it's better to be
defensive. The code is now also somewhat easier to understand.
This adds support for ChapterSegmentEditionUID (pull request #258),
and also fixes issue #278 (pull request #292).
In fact, this is a straight merge of pr/292, which also contains pr/258.
Note that you still need --vd-lavc-o='strict=-2' to enable the decoder.
Also, there's no guarantee that all required features for HEVC demuxing
are actually implemented, nor that the current muxing schema is the
final one.
Change talloc destructor so that they can never signal failure, and
don't return a status code. This makes our talloc copy even more
incompatible to upstream talloc, but on the other hand this is
preparation for getting rid of talloc entirely.
(The talloc replacement in the next commit won't allow the talloc_free
equivalent to fail, and the destructor return value would be useless.
But I don't want to change any mpv code either; the idea is that the
talloc replacement commit can be reverted for some time in order to
test whether the talloc replacement introduced a regression.)
To support edition references in matroska chapters, editions need to be
remembered for each chapter and source. To facilitate easier management
of these now-paired uids, a single structure is used.
There is uninitialized memory access if the actual size isn't passed
along. In the worst case, this can cause a source to be loaded against
the uninitialized memory, causing a false count of found versus required
sources, preventing the "Failed to find ordered chapter part" message.
By default, libavformat uses UDP for rtsp playback. This doesn't work
very well. Apparently the reason is that the buffer sizes libavformat
chooses for UDP are way too small, and switching to TCP gets rid of this
issue entirely (thanks go to Reimar Döffinger for figuring this out).
In theory, you can set buffer sizes as libavformat options, but that
doesn't seem to help.
Add an option to select the rtsp transport, and make TCP the default.
Also remove an outdated comment from stream.c.
In insane files with a very huge number of subtitle events, and if the
--demuxer-mkv-subtitle-preroll option is given, seeking can still
overflow the packet queue. Normally, the subtitle_preroll variable
specifies the maximum number of packets that can be added. But once this
number is reached, the normal seeking behavior is enabled, which will
add all subtitle packets with the right timestamps to the packet queue.
At this point the next video keyframe can still be quite far away, with
enough subtitle packets on the way to overflow the packet queue.
Fix this by always setting an upper limit of subtitle packets read
during seeking. This should provide additional robustness even if the
preroll option is not used.
This means that even with normal seeking, at most 500 subtitle packets
are demuxed. Packets after that are discarded.
One slightly questionable aspect of this commit is that subtitle_preroll
is never reset in audio-only mode, but that is probably ok.
The quicktime html scripting guide suggests to wrap urls not
necesarly associated with quicktime in a .mov file.
(so that when <embed>ing videos quicktime would be forced.)
These mov files may contain several "Text Hacks".
One of these is RTSPtext.
The suggested/allowed format (as regex) is like:
RTSPtext[ \r]RTSP://url
See also p.51 of:
https://developer.apple.com/library/mac/documentation/QuickTime/Conceptual/QTScripting_HTML/QTScripting_HTML.pdf
In reality there are also files like (e.g. zdfmediathek.de):
RTSPtext\nrtsp://url\n\n
Lets handle these files as a playlist with one element.
The --deinterlace option does on playback start what the "deinterlace"
property normally does at runtime. You could do this before by using the
--vf option or by messing with the vo_vdpau default options, but this
new option is supposed to be a "foolproof" way.
The main motivation for adding this is so that the deinterlace property
can be restored when using the video resume functionality
(quit_watch_later command).
Implementation-wise, this is a bit messy. The video chain is rebuilt in
mpcodecs_reconfig_vo(), where we don't have access to MPContext, so the
usual mechanism for enabling deinterlacing can't be used. Further,
mpcodecs_reconfig_vo() is called by the video decoder, which doesn't
have access to MPContext either. Moving this call to mplayer.c isn't
currently possible either (see below). So we just do this before frames
are filtered, which potentially means setting the deinterlacing every
frame. Fortunately, setting deinterlacing is stable and idempotent, so
this is hopefully not a problem. We also add a counter that is
incremented on each reconfig to reduce the amount of additional work per
frame to nearly zero.
The reason we can't move mpcodecs_reconfig_vo() to mplayer.c is because
of hardware decoding: we need to check whether the video chain works
before we decide that we can use hardware decoding. Changing it so that
this can be decided in advance without building a filter chain sounds
like a good idea and should be done, but we aren't there yet.
Retrieve per-chapter metadata, but don't do much with it. We just make
the metadata of the _current_ chapter available as chapter-metadata
property. Returning the full chapter list with metadata would be no
problem, except that the property interface isn't really good with
structured data, so it's not available for now.
Not sure if it's worth it, but it was requested via github issue #201.
Consider the cluster used for prerolling contains an insane amount of
subtitle packets. Then the demuxer packet queue would be full of
subtitle packets, and demux.c would refuse to read any further packets -
including video and audio packets, resulting in EOF. Since everything
involving Matroska and subtitles is 100% insane, this can actually
happen.
Fix this by putting a limit on the number of subtitle packets read by
preroll, and throw away any further packets if the limit is exceeded. If
this happens, the preroll mechanism will stop working, but the player's
operation is unaffected otherwise.
The really funny thing about this commit is that this code is added on
top of another work around. Basically, subtitle seeking in libavformat
is completely broken. To make it useful, we have to add yet another
workaround.
The basic problem is that libavformat's subtitle seeking code always
uses the stream time base, instead of AV_TIME_BASE if stream index -1 is
passed to the avformat_seek_file() function.
Fixes github issue #216. Hopefully this will be fixed in ffmpeg too at
some point.
Port it from playlist_parser.c to demux_playlist.c. Also, change the m3u
parser to drop whitespace from the trailing part of the line (will make
it work properly with windows line endings).
(I hoped that this would make MMS URIs with http instead of mmsh
prefixes work, but it doesn't. Instead, it leads to a playlist loop. So
solving this issue would require a change in ffmpeg, probably.)
Apparently, it is popular to store large files in uncompressed rar
archives. Extracting files is not practical, and some media players
suport playing directly from uncompressed rar (at least VLC and some
DirectShow components).
Storing or accessing files this way is completely idiotic, but it is
a common practice, and the ones subjected to this practice can't do
much to change this (at least that's what I assume/hope). Also, it's
a feature request, so we say yes.
This code is mostly taken from VLC (commit f6e7240 from their git tree).
We also copy the way this is done: opening a rar file by itself yields
a playlist, which contains URLs to the actual entries in the rar file.
Compressed entries are simply skipped.
Modeled after the old playlist_parser.c, but actually new code, and it
works a bit differently.
Demuxers (and sometimes streams) are the component that should be used
to open files and to determine the file format. This was already done
for subtitles, but playlists still use a separate code path.
The way this was added to FFmpeg is less than ideal, because it requires
text parsing in the Matroska demuxer. But in order to use the FFmpeg
webvtt-to-ass converter, we still have to mimic this in some way. We do
this by putting the parsing into sd_lavc_conv.c, before the subtitle
packet is passed to libavcodec. At least this keeps the ugliness out of
unrelated code.
There is some change that FFmpeg will fix their design eventually.
Instead of rewriting the parsing code, we simply borrow it from FFmpeg's
Matroska demuxer.
Otherwise, this would just try to demux a good chunk of the file, even
though the operation can't succeed anyway.
This caused some pretty strange issues, where perfectly valid use cases
would print a "Too many packets in the demuxer packet queue..." message.
The rawaudio demuxer read one frame per packet, basically a few bytes,
which caused insane overhead. (I found this when I couldn't play raw
audio without dropouts when using -v, which printed a line per packet
read.)
Fix this and read 1 second of audio per packet. This is a regression
since cfa5712 (merging of demux_rawaudio and demux_rawvideo).
Originally, the objective of this commit was changing --edition to be
1-based, but this was cancelled. I'm still leaving the change to
demux_mkv.c though, which is now only of cosmetic nature.
This is completely useless, and in this particular case, it broke the
fallback for MLP2 subtitles (stored as .txt files) to demux_subreader.
(Yes, libavformat should be fixed to handle this, but for now this will
_always_ break playback of subtitle files stored in .txt.)
You can still force this demuxer, but by default we will just pretend
that the "tty" demuxer does not exist.
Perhaps not very useful, but reserved for situations when a user reports
awful latency and experimentation/debugging might be required to find
out why or to fix it (happens often).
avio_alloc_context() is documented to require an av_malloc'ed buffer. It
appears libavformat can even reallocate the buffer while it is probing,
so passing a static buffer can in theory lead to crashes.
I couldn't reproduce such a crash, but apparently it happened to
mplayer-svn. This commit follows the mplayer fix in svn commit r36397.
Move the decoder parts from vo_vdpau.c to a new file vdpau_old.c. This
file is named so because because it's written against the "old"
libavcodec vdpau pseudo-decoder (e.g. "h264_vdpau").
Add support for the "new" libavcodec vdpau support. This was recently
added and replaces the "old" vdpau parts. (In fact, Libav is about to
deprecate and remove the "old" API without deprecation grace period,
so we have to support it now. Moreover, there will probably be no Libav
release which supports both, so the transition is even less smooth than
we could hope, and we have to support both the old and new API.)
Whether the old or new API is used is checked by a configure test: if
the new API is found, it is used, otherwise the old API is assumed.
Some details might be handled differently. Especially display preemption
is a bit problematic with the "new" libavcodec vdpau support: it wants
to keep a pointer to a specific vdpau API function (which can be driver
specific, because preemption might switch drivers). Also, surface IDs
are now directly stored in AVFrames (and mp_images), so they can't be
forced to VDP_INVALID_HANDLE on preemption. (This changes even with
older libavcodec versions, because mp_image always uses the newer
representation to make vo_vdpau.c simpler.)
Decoder initialization in the new code tries to deal with codec
profiles, while the old code always uses the highest profile per codec.
Surface allocation changes. Since the decoder won't call config() in
vo_vdpau.c on video size change anymore, we allow allocating surfaces
of arbitrary size instead of locking it to what the VO was configured.
The non-hwdec code also has slightly different allocation behavior now.
Enabling the old vdpau special decoders via e.g. --vd=lavc:h264_vdpau
doesn't work anymore (a warning suggesting the --hwdec option is
printed instead).
Remove the (now unused) code for determining correct-pts mode based on
the demuxer in use. Change its description in the manpage to reflect
what this option does now.
Gives really funky results with PNG attachments otherwise. The main
problem is that avcodec_flush_buffers() does not fully reset the
decoder, so passing multiple PNG packets without keyframe flags will
attempt to combine the new picture with the previously decoded
contents. (Makes no sense with proper PNG - maybe this codepath is
intended for MNG or APNG.)
In general, this warning can hint to actual bugs. We don't enable it
yet, because it would conflict with some unmerged code, and we should
check with clang too (this commit was done by testing with gcc).
This also affects --audiofile. The previous behavior wasn't really
useful. There are even separate switches for that: --audio-demuxer and
--sub-demuxer.
This fixes the sample RA_missing_timestamps.mkv. Pretty funny how this
code got it almost right, but not quite, so it was broken all these
years. And then, after everyone stopped caring, someone comes and fixes
it. (By the way, I know absolutely nothing about realaudio.)
This fixes playback of the sample linked by FFmpeg ticket 2508. The fix
follows ffmpeg commit 6158a3b (although it's not exactly the same).
The problem here is that the file contains an apparently non-sense
DefaultDuration value. DefaultDuration for audio tracks is used to
derive PTS values for packets with no timestamps, like they can happen
with frames inside a laced block. So the first packet of a SimpleBlock
will have a correct PTS, while the PTS values of the following packets
are calculated using DefaultDuration, and thus are broken.
This leads to seemingly ok playback, but broken A/V sync. Not using the
DefaultDuration value will leave the PTS values of these packets unset,
and the audio decoder can derive them from the output instead.
The fix more or less uses a heuristic to detect the broken case: if the
sample rate is 8 KHz (Matroska default, can assume unset), and the codec
is AC3 (as the broken file did), don't use it. I'm not sure why this
should be done only for AC3, maybe the muxing application (mkvmerge
v4.9.1) has known issues with AC3. AC3 also doesn't support 8 KHz as
sample rate natively.
(By the way, I'm not sure why we should honor the DefaultDuration at all
for audio. It doesn't seem to be needed. You can't seek to these frames,
and decoders should always be able to produce perfect PTS values by
adding the duration of the decoded audio to the first PTS.)
Matroska has an output sample rate (OutputSamplingFrequency), which in
theory should be forced instead of whatever the decoder outputs. But it
appears no software (other than mplayer2 and mpv until now) actually
respects this. Even worse, there were broken files around, which played
correctly with (in theory) broken software, but not mplayer2/mpv. Hacks
were added to our code to play these files correctly, but they didn't
catch all cases.
Simplify this by doing what everyone else does, and always use the
decoder's sample rate instead. In particular, we try to handle all
sample rate issues like libavformat's Matroska demuxer does.
Guess the colorspace directly in mpcodecs_reconfig_vo(), instead of in
set_video_colorspace(). The difference is that the latter function just
makes the video filter chain (and VOs) force the detected colorspace,
and then throws it away, while the former is a bit more general and
central. Not really a big difference and it doesn't matter much in
practice, but it guarantees that there is no internal disagreement about
the colorspace.
DVD playback had some trouble with PTS resets: libavformat's genpts
feature would try reading until EOF (worst case) to find a new usable
PTS in case a packet's PTS is not set correctly. Especially with slow
DVD access, this would make the player to appear frozen.
Reimplement it partially in demux_lavf.c, and use that code in the DVD
case. This is heavily "inspired" by the code in av_read_frame from
libavformat/utils.c. The difference is that we stop reading if no PTS
has been found after 50 packets (consider this a heuristic). Also, we
don't bother with the PTS wrapping and last-frame-before-EOF handling.
Even with normal PTS wraps, the player frontend will go to hell for the
duration of a frame anyway, and should recover quickly after that.
The terribleness of this commit is mostly that we duplicate libavformat
functionality, and that we suddenly need a packet queue.
All demuxers make a reasonable effort to set packet timestamps, and thus
support correct-pts mode. This commit also implicitly switches
demux_rawvideo to correct-pts mode.
We still allow demuxers to disable correct-pts mode in theory.
Get rid of the strange and messy reliance on DEMUXER_TYPE_ constants.
Instead of having two open functions for the demuxer callbacks (which
somehow are both optional, but you can also decide to implement both...),
just have one function. This function takes a parameter that tells the
demuxer how strictly it should check for the file headers. This is a
nice simplification and allows more flexibility.
Remove the file extension code. This literally did nothing (anymore).
Change demux_lavf so that we check our other builtin demuxers first
before libavformat tries to guess by file extension.
This removes the dependency on DEMUXER_TYPE_* and the file_format
parameter from the stream open functions.
Remove some of the playlist handling code. It looks like this was
needed only for loading linked mov files with demux_mov (which was
removed long ago).
Delete a minor bit of dead network-related code from stream.c as well.
Move codec_tags.h include to demux_mkv.c, because this is the only file
which still uses it.
Move new_sh_stream() to demux.h, because this is more proper.
Before this commit, we tried to play along with libavformat and tried
to pretend that attached pictures are video streams with a single
frame, and that the frame magically appeared at the seek position when
seeking. The playback core would then switch to a mode where the video
has ended, and the "remaining" audio is played.
This didn't work very well:
- we needed a hack in demux.c, because we tried to read more packets in
order to find the "next" video frame (libavformat doesn't tell us if
a stream has ended)
- switching the video stream didn't work, because we can't tell
libavformat to send the packet again
- seeking and resuming after was hacky (for some reason libavformat sets
the returned packet's PTS to that of the previously returned audio
packet in generic code not related to attached pictures, and this
happened to work)
- if the user did something stupid and e.g. inserted a deinterlacer by
default, a picture was never displayed, only an inactive VO window)
- same when using a command that reconfigured the VO (like switching
aspect or video filters)
- hr-seek didn't work
For this reason, handle attached pictures as separate case with a
separate video decoding function, which doesn't read packets. Also,
do not synchronize audio to video start in this case.
The code touched by this commit makes sure that DVD subtitle tracks
known by libdvdread but not known by demux_lavf can be selected and
displayed properly. These subtitle tracks have the first packet
some time late in the packet stream, so that libavformat won't
immediately recognize them, and will add the track as soon as the
first packet is seen during normal demuxing.
demux_mpg used to handle this elegantly: you just set the MPEG ID of
the stream you wanted. demux_lavf couldn't do this, so it was emulated
with a DEMUXER_CTRL. This commit changes it so that new streams are
selected by default (if autoselect is enabled), and the playloop
simply can take appropriate action before the lower layer throws away
the first packet.
This also changes the demux_lavf behavior that subtitle packets are
always demuxed, even if not needed. (They were immediately thrown away,
so there was no advantage to this.)
Further, this adds the ability to demux.c to deal with demuxing more
than one stream of a kind at once. (Though currently it's not useful.)
AVDISCARD_DEFAULT is probably a bit better for normal decoding.
AVDISCARD_NONE would (as by documentation) include "useless" packets
too, while DEFAULT filters these.
Generally remove all accesses to demux_stream from all the code, except
inside of demux.c. Make it completely private to demux.c.
This simplifies the code because it removes an extra concept. In demux.c
it is reduced to a simple packet queue. There were other uses of
demux_stream, but they were removed or are removed with this commit.
Remove the extra "ds" argument to demux fill_buffer callback. It was
used by demux_avi and the TV pseudo-demuxer only.
Remove usage of d_video->last_pts from the no-correct-pts code. This
field contains the last PTS retrieved after a packet that is not NOPTS.
We can easily get this value manually because we read the packets
ourselves. Reuse sh_video->last_pts to store the packet PTS values. It
was used only by the correct-pts code before, and like d_video->last_pts,
it is reset on seek. The behavior should be exactly the same.
Currently, all demuxer fill_buffer functions have a demux_stream
parameter. We want to remove that, but the TV code still depends on
it. Add a hack to remove that dependency.
The problem with the TV code is that reading video and audio frames
blocks, so in order to avoid a deadlock, you should read either of
them only if the decoder actually requests new data.
For now, we want to get rid of the demux->sub access, because this
field will become private to demux.c in a later commit. So replace the
current hack with another hack.
The need for the hack will be removed sooner or later. (Instead of
autoselecting a specific stream, all new streams will be enabled by
default, so that no packets can get lost. The frontend will then be
responsible to deselect unwanted streams.)
This is not directly related to the handling of format changes itself,
but playing audio normally after the change. This was broken: the output
byte rate was not recalculated, so audio-video sync was simply broken.
Fix this by calculating the byte rate on the fly, instead of storing it
in sh_audio.
Format changes are relatively common (switches between stereo and 5.1
in TV recordings), so this fixes a somewhat critical bug.
Partial packet reads were needed because the video/audio parsers were
working on top of them. So it could happen that a parser read a part of
a packet, and returned that to the decoder. With libavformat/libavcodec,
packets are already parsed, and everything is much simpler.
Most of the simplifications in ad_spdif could have been done earlier.
Remove some other stuff as well, like the questionable slave mode start
time reporting (could be replaced by proper code, but we don't bother).
Remove the unused skip_audio_frame() functionality as well (it was used
by old demuxers). Some functions become private to demux.c, like
demux_fill_buffer(). Introduce new packet read functions, which have
simpler semantics. Packets returned from them are owned by the caller,
and all packets in the demux.c packet queue are considered unread.
Remove special code that dropped subtitle packets with size 0. This
used to be needed because it caused special cases in the old code.
The demux_open as well as demux_open_withparams calls don't use the
stream selection parameters anymore, so remove them everywhere.
Completes the previous commit.
These separate arrays were used by the old demuxers and are not needed
anymore. We can simplify track switching as well.
One interesting thing is that stream/tv.c (which is a demuxer) won't
respect --no-audio anymore. It will probably work as expected, but it
will still open an audio device etc. - this is because track selection
is now always done with the runtime track switching mechanism. Maybe
the TV code could be updated to do proper runtime switching, but I
can't test this stuff.
The audio parser was needed only by the "old" demuxers, and
demux_rawaudio. All other demuxers output already parsed packets.
demux_rawaudio is usually for raw audio, so using a parser with it
doesn't usually make sense. But you can also force it to read
compressed formats with fixed packet sizes, in which case the parser
would have been used. This use case is probably broken now, but you
will be able to do the same thing with libavformat demuxers.
Delete demux_avi, demux_asf, demux_mpg, demux_ts. libavformat does
better than them (except in rare corner cases), and the demuxers have
a bad influence on the rest of the code. Often they don't output
proper packets, and require additional audio and video parsing. Most
work only in --no-correct-pts mode.
Remove them to facilitate further cleanups.
STREAM_CTRL_GET_METADATA will be used to poll for streamcast metadata.
Also add DEMUXER_CTRL_UPDATE_INFO, which could in theory be used by
demux_lavf.c. (Unfortunately, libavformat is too crappy to read metadata
mid-stream for mp3 or ogg, so we don't implement it.)
The filter chain and the video ouputs have config() functions. They are
strictly limited to transfering the video size and format. Other
parameters (like color levels) have to be transferred separately.
Improve upon this by introducing a separate set of reconfig() functions,
which use mp_image_params to carry format parameters. This struct
contains all image format related parameters from config(), plus
additional parameters such as colorspace.
Change vf_rotate to use it, as well as vo_opengl. vf_rotate is just
an example/test case, but vo_opengl will need it later.
The intention is also to get rid of VOCTRL_SET_YUV_COLORSPACE. This
information is now handed to the VOs via reconfig(). The getter,
VOCTRL_GET_YUV_COLORSPACE, will still be needed though.
Old code used to use libass' recoding feature, which is a copy of the
old MPlayer code. We dropped that a few commits ago. Unfortunately,
this made it impossible to load some subtitle files, like UTF-16 files.
Make .ass loading respect -subcp again. We do this by recoding the
probe buffer to UTF-8, and then trying to load it normally. (Yep.)
Since UTF-16 in particular will effectively half the probe buffer size,
double the probe size.
demux_libass.c allows us to make subtitle format detection part of the
normal file loading process. libass has no probe function, but trying to
load the start of a file (the first 4 KB) is good enough. Hope that
libass can even handle random binary input gracefully without printing
stupid log messages, and that the libass parser doesn't accept too many
non-ASS files as input.
This doesn't handle the -subcp option correctly yet. This will be fixed
later.
The default correct-pts mode depended on which demuxer was opened last.
Often this is the subtitle demuxer. The correct-pts mode should be
decided on the demuxer for video instead.
subreader.c (before this commit renamed to demux_subreader.c) was
special cased to the -sub option. The plan is using the normal demuxer
codepath for all subtitle formats (so we can prefer libavformat demuxers
for most formats).
There are some subtle changes. The probe size is restricted to 32 KB
(instead of unlimitted + giving up after 100 lines of input). For
formats like MicroDVD, the video FPS isn't used anymore, because it's
not available on the subtitle demuxer level. Instead, hardcode it to
23.976 FPS (libavformat seems to do the same). The user can probably
still use -sub-fps to fix the timing. Checking the file extension for
".utf"/".utf8"/".utf-8" is simply removed (seems worthless, was in the
way, and I've never seen this anywhere).
Simpler, reduces the amount of copying.
We still have to malloc+memcpy the probe buffer though, because padding
with FF_INPUT_BUFFER_PADDING_SIZE is required by libavformat.
If a subtitle is external, read it completely and add all subtitle
events in advance when the subtitle track is selected. This is done
for text subtitles only. (Note that subreader.c and subtitles loaded
with libass are different and don't have anything to do with this
commit.)
Seems like a completely unnecessary complication. Instead, always add a
1 byte padding (could be extended if a caller needs it), and clear it.
Also add some documentation. There was some, but it was outdated and
incomplete.
The stream EOF flag should only be set when trying to read past the end
of the file (relatively similar to unix files). Always clear the EOF
flag on seeking. Trying to set it "properly" (depending whether data is
available at seek destination or not) might be an ok idea, but would
require attention to too many special cases. I suspect before this
commit (and in MPlayer etc. too), the EOF flag wasn't handled
consistently when the stream position was at the end of the file.
Fix one special case in ebml.c and stream_skip(): this function couldn't
distinguish between at-EOF and past-EOF either.
This function was called in various places. Most time, it was used
before a seek. In other cases, the purpose was apparently resetting
the EOF flag. As far as I can see, this makes no sense anymore. At
least the stream_reset() calls paired with stream_seek() are completely
pointless. A seek will either seek inside the buffer (and reset the
EOF flag), or do an actual seek and reset all state.
This branch heavily refactors the subtitle code (both loading and
rendering), and adds support for a few new formats through FFmpeg.
We don't remove any of the old code yet. There are still some subtleties
related to subreader.c to be resolved: code page detection & conversion,
timing post-processing, UTF-16 subtitle support, support for the -subfps
option. Also, SRT reading and loading ASS via libass should be turned
into proper demuxers. (SRT is needed because Libav's is gravely broken,
and we want ASS loading via libass to cover full libass format support.
Both should be demuxers which are probed _before_ libavformat, so that
all subtitles can be loaded through the demuxer infrastructure, and
libavformat subtitles don't need to be treated in a special way.)
Before this, subtitle packets were returned as data ptr/len pairs, and
mplayer.c got the rest (pts and duration) directly from the demuxer
data structures. Then mplayer.c reassembled the packet data structure
again.
Pass packets directly instead. The mplayer.c side stays a bit awkward,
because the (now by default unused) DVD path keeps getting in the way.
In demux.c there's lots of weird stuff (3 functions that read packets,
really?), but we want to keep the code equivalent for now to avoid
hitting weird issues and corner cases.
Makes WebVTT actually work.
Also simplify the logic for setting duration. Only the subtitle path
uses the packet duration, so the checks for STREAM_SUB as well as the
keyframe flag are redundant.
Apparently duration and convergence_duration are the same thing, but
convergence_duration was added as Matroska-specific hack to get a higher
value range (int vs. int64_t) with high resolution Matroska timebases.
For us it doesn't matter, because double floats are used for timestamps
and durations.
This means subassconvert.c is split in sd_srt.c and sd_microdvd.c. Now
this code is involved in the sub conversion chain like sd_movtext is.
The invocation of the converter in sd_ass.c is removed.
This requires some other changes to make the new sub converter code work
with loading external subtitles. Until now, subtitles loaded via
subreader.c was assumed to be in plaintext, or for some formats, in ASS
(except in -no-ass mode). Then these were added to an ASS_Track. Change
this so that subtitles are always in their original format (as far as
decoders/converters for them are available), and turn every sub event
read by subreader.c as packet to the dec_sub.c subtitle chain.
This removes differences between external/demuxed and -ass/-no-ass code
paths further.
Internally, stream_dvd.c returned DEMUXER_TYPE_MPEG_PS, and the same
value was hardcoded to enforced usage of demux_lavf in demux.c. But
"-demuxer mpegps" basically did the same, so that switch was broken
for this format. Undo this and don't request a demuxer in stream_dvd.c.
demux_lavf.c is (probably) good enough to probe correctly with DVD.
Otherwise, we'd actually have to do something completely different to
force the libavformat demuxer.
Make the sub decoder stuff independent from sh_sub (except for
initialization of course). Sub decoders now access a struct sd only,
instead of getting access to sh_sub. The glue code in dec_sub.c is
similarily independent from osd.
Some simplifications are made. For example, the switch_id stuff is
unneeded: the frontend code just has to make sure to call osd_changed()
any time subtitles are switched.
This is also preparation for introducing subtitle converters. It's much
cleaner to completely separate demuxer header/renderer glue/decoders
for this purpose, especially since sub converters might completely
change how demuxer headers have to be interpreted.
Also pass data as demux_packets. Currently, this doesn't help much, but
libavcodec converters might need scary stuff like packet side data, so
it's perhaps better to go with passing packets.
Subtitle files are opened in mplayer.c, not using the demuxer
infrastructure in general. Pretend that this is not the case (outside of
the loading code) by opening a pseudo demuxer that does nothing. One
advantage is that the initialization code is now the same, and there's
no confusion about what the difference between track->stream,
track->sh_sub and mpctx->sh_sub is supposed to be.
This is a bit stupid, and it would be much better if there were proper
subtitle demuxers (there are many in recent FFmpeg, but not Libav). So
for now this is just a transition to a more proper architecture. Look
at demux_sub like an artifical limb: it's ugly, but don't hate it - it
helps you to get on with your life.
This unifies the subtitle rendering path. Now all subtitle rendering
goes through sd_ass.c/sd_lavc.c/sd_spu.c.
Before that commit, the spudec.h functions were used directly in
mplayer.c, which introduced many special cases. Add sd_spu.c, which is
just a small wrapper connecting the new subtitle render API with the
dusty old vobsub decoder in spudec.c.
One detail that changes is that we always pass the palette as extra
data, instead of passing the libdvdread palette as pointer to spudec
directly. This is a bit roundabout, but actually makes the code simpler
and more elegant: the difference between DVD and non-DVD dvdsubs is
reduced.
Ideally, we would just delete spudec.c and use libavcodec's DVD sub
decoder. However, DVD playback with demux_mpg produces packets
incompatible to lavc. There are incompatibilities the other way around
as well: packets from libavformat's vobsub demuxer are incompatible to
spudec.c. So we define a new subtitle codec name for demux_mpg subs,
"dvd_subtitle_mpg", which only sd_spu can decode.
There is actually code in spudec.c to "assemble" fragments into complete
packets, but using the whole spudec.c is easier than trying to move this
code into demux_mpg to fix subtitle packets.
As additional complication, Libav 9.x can't decode DVD subs correctly,
so use sd_spu in that case as well.
The new wavpack packet format (see previous commit) doesn't work with
older libavcodec versions, so disable the new code in this case.
The version numbers are only approximate, since the libavcodec version
wasn't bumped with the wavpack change, but it's close enough.
Libav introduced a silent API breakage by changing what wavpack packets
the libavcodec decoder accepts. Originally the libavcodec codec accepted
Matroska-style wavpack packets. Libav commit 9b6f47c removed this
capability from the libavcodec code, and added code to libavformat's
Matroska demuxer to "rearrange" wavpack packets. Since demux_mkv still
sent Matroska-style packets, playback failed.
Fix this by "rearranging" packets in demux_mkv as well by copying
libavformat's code. (The best kind of fix.)
Tested with [CCCP]_Mega_Lossless_Audio_Test.mkv, as well as with a
sample generated by mkvmerge.
The core deselected all streams on initialization, and then selected the
streams it actually wanted. This was no problem for
demux_mkv/demux_lavf, but old demuxers (like demux_asf) could lose some
packets. The problem is that these demuxers can buffer some data on
initialization, which then is flushed on track switching. Fix this by
explicitly avoiding deselecting a wanted stream.
When AAC is streamed over HTTP, using libavformat defaults is
pathetically slow. One solution for that is skipping probing and using
the mimetype to identify that it's AAC instead. This is what we did
before this commit (and ffmpeg does it too, but their logic is too
"inaccessible" for mpv).
This is still pretty fragile though. Make it a bit more robust by
requiring minimal probing. A probescore of 25 is reached after feeding
2 KB to libavformat (instead of > 500 KB for the normal probescore), so
use that. This is done only when streaming AAC from HTTP to reduce the
possibility of weird breakages for other formats.
Also reduce analyzeduration. The default analyzeduration will make
libavformat read lots of data, which makes playback start slow. So we
set analyzeduration to a low value. On the other hand, doing that for
other formats is risky, because there are unspecified effects with
certain "strange" formats (like transport streams). So we do this only
if we're streaming AAC from HTTP as well.
tl;dr libavformat is shit for media players
This can control whether demux_lavf should use the HTTP mime type to
determine the format, instead of probing the data with the libavformat
API. Do this to allow easier debugging in case the mimetype is
incorrect. (This is done only for AAC streams right now.)
