If the packet read function returns, and EOF was detected, and a seek
was issued in the meantime, then don't use the EOF result. The seek will
be processed later, and reset the EOF state anyway.
The main effect is probably that we don't return EOF to the decoders
(which the playback core resets before issuing the seek), and that we
won't log an EOF message.
Not important, but slightly more correct.
When switching tracks, we normally have the problem that data gets lost
due to readahead buffering. (Which in turn is because we're stubborn and
instruct the demuxers to discard data on unselected streams.) The
demuxer layer has a hack that re-reads discarded buffered data if a
stream is enabled mid-stream, so track switching will seem instant.
A somewhat similar problem is when all tracks of an external files were
disabled - when enabling the first track, we have to seek to the target
position.
Handle these with the same mechanism. Pass the "current time" to the
demuxer's stream switch function, and let the demuxer figure out what to
do. The demuxer will issue a refresh seek (if possible) to update the
new stream, or will issue a "normal" seek if there was no active stream
yet.
One case that changes is when a video/audio stream is enabled on an
external file with only a subtitle stream active, and the demuxer does
not support rrefresh seeks. This is a fuzzy case, because subtitles are
sparse, and the demuxer might have skipped large amounts of data. We
used to seek (and send the subtitle decoder some subtitle packets
twice). This case is sort of obscure and insane, and the fix would be
questionable, so we simply don't care.
Should mostly fix#3392.
This commit adds an --audio-channel=auto-safe mode, and makes it the
default. This mode behaves like "auto" with most AOs, except with
ao_alsa. The intention is to allow multichannel output by default on
sane APIs. ALSA is not sane as in it's so low level that it will e.g.
configure any layout over HDMI, even if the connected A/V receiver does
not support it. The HDMI fuckup is of course not ALSA's fault, but other
audio APIs normally isolate applications from dealing with this and
require the user to globally configure the correct output layout.
This will help with other AOs too. ao_lavc (encoding) is changed to the
new semantics as well, because it used to force stereo (perhaps because
encoding mode is supposed to produce safe files for crap devices?).
Exclusive mode output on Windows might need to be adjusted accordingly,
as it grants the same kind of low level access as ALSA (requires more
research).
In addition to the things mentioned above, the --audio-channels option
is extended to accept a set of channel layouts. This is supposed to be
the correct way to configure mpv ALSA multichannel output. You need to
put a list of channel layouts that your A/V receiver supports.
It used not to work - but now it apparently does. Not sure when that got
fixed in FFmpeg, but there's no longer a reason to keep this hack.
This also gets rid of the check for the read_seek2 field, which is not
part of the public API.
Since the libavformat API is crap, we have to apply tons of heuristics
to check whether seeking will work. (No, checking it at seek time isn't
going to work either, because if a seek fails, the demuxer will be in an
undefined state. Because the libavformat API is crap.)
I've got a broken webm that fails to seek correctly with "--start=0".
The problem is that every index entry points to 1 byte before cluster
start (!!!). demux_mkv tries to resync to the next cluster, but since it
already has read 2 bytes with ebml_read_id(), it doesn't get the first
cluster, but the following one. Actually, it can be any amount of bytes
from 1-4, whatever happens to look valid at this essentially random byte
position.
Improve this by resyncing from the original position, instead of the one
after the EBML element ID has been attempted to be read.
The file shows the following headers:
| + Muxing application: google at 177
| + Writing application: google at 186
Indeed, the file was downloaded with youtube-dl. I can only guess that
Google got it completely wrong.
demux_playlist.c recognizes if the source stream points to a directory,
and adds its directory entries. Until now, only 1 level was added.
During playback, further directory entries could be resolved as
directory paths were "played".
While this worked fine, it lead to frequent user confusion, because
playlist resuming and other things didn't work as expected. So just
recursively scan everything.
I'm unsure whether it's a good fix, but at least it gets rid of the
complaints. (And probably will add others.)
Now it will always be able to seek back to the start, even if the index
is sparse or misses the first entry.
This can be achieved by reusing the logic for incremental index
generation (for files with no index), and start time probing (for making
sure the first block is always indexed).
AVFormatContext.codec is deprecated now, and you're supposed to use
AVFormatContext.codecpar instead.
Handle this for all of the normal playback code.
Encoding mode isn't touched.
This was changed in 2014, so I suppose users will usually have a FFmpeg
release which includes the corresponding upstream change. If not, well
too bad for those MicroDVD-obsessed users.
Also don't try to retrieve the default framerate as exported by the
demuxer, and instead hardcode it and trust it won't ever change. this
avoids that we have to deal with a larger mess in the codecpar commit.
I don't trust it one bit, and it's a bother with the codecpar change.
If it turns out to be important for some file formats, it could be
added back (or FFmpeg fixed).
This reverts commit 503c6f7fd6.
There are situations where some decoders (MF apparently) always require
a timestamp. Also, this makes bitrate estimation more granular than
necessary. It seems it's better to try to detect fiels with broken
default durations explicitly instead. Or maybe something should be
added to smooth audio timestamps after filters.
Instead of having a separate for each, which also requires separate
additional caching in the demuxer. (The demuxer adds an indirection,
since STREAM_CTRLs are not thread-safe.)
Since this includes the cache speed, this should fix#3003.
SEEK_HR is interpreted by demux_mkv.c, and enables subtitle preroll by
prefetching additional subtitle pakcets which might overlap with the
seek destination. This should make the case work when segment boundaries
fall into the middle of subtitle events.
This still usually leaves a flicker of at least 1 frame on start,
because dec_sub.c does not ensure that enough subtitles are read before
rendering after a segment switch. (Probably a WONTFIX.)
This is simpler, because it doesn't have to wait from both threads for
synchronization.
Apart from being simpler/cleaner, this serves vague plans to stop/start
the demuxer thread itself automatically on demand (for the purpose of
reducing unneeded resource usage).
This pause stuff is bothersome and is needed only for a few corner-
cases. This commit removes it from the demuxer public API and replaces
it with a demux_run_on_thread() function and refactors the code which
needed demux_pause(). The next commit will change the implementation.
Commit 503c6f7f essentially removed timestamps from "laces" (Block sub-
divisions), which means many audio packets will have no timestamp.
There's no reason why bitrate calculation can't just delayed to a point
when the next timestamp is known.
Fixes#2903 (no audio bitrate with mkv files).
stream->info can be NULL if it's the cache wrapper. To be fair,
stream->info is considered private API anyway. So don't access it, but
check the URL instead.
This reverts commit af66fa8fa5.
The reverted commit caused AVCodecContext.channel_layout to be set,
while requesting stereo downmix will make libavcodec output a stupid
message:
ac3: Channel layout '5.1' with 6 channels does not match specified number of channels 2: ignoring specified channel layout
The same happens with --demuxer=lavf (without this change too).
I'm not quite sure what acrobatics are required to shut up libavcodec,
but for now revert the commit. It was a rather minor, almost cosmetic
issue, which I consider less important than clean CLI terminal output.
Ever since a change in mplayer2 or so, relative seeks were translated to
absolute seeks before sending them to the demuxer in most cases. The
only exception in current mpv is DVD seeking.
Remove the SEEK_ABSOLUTE flag; it's not the implied default. SEEK_FACTOR
is kept, because it's sometimes slightly useful for seeking in things
like transport streams. (And maybe mkv files without duration set?)
DVD seeking is terrible because DVD and libdvdnav are terrible, but
mostly because libdvdnav is terrible. libdvdnav does not expose seeking
with seek tables. (Although I know xbmc/kodi use an undocumented API
that is not declared in the headers by dladdr()ing it - I think the
function is dvdnav_jump_to_sector_by_time().) With the current mpv
policy if not giving a shit about DVD, just revert our half-working seek
hacks and always use dvdnav_time_search(). Relative seeking might get
stuck sometimes; in this case --hr-seek=always is recommended.
If a stream is marked as EOF (due to no packets found in reach), then we
need to wakeup the decoder. This is important especially if no packets
are found at the start of the file, so the A/V sync logic actually
starts playback, instead of waiting for packets that will never come.
(It would randomly start playback when running the playback loop due to
arbitrary external events like user input.)
Commit 943f76e6, which already tried this, was very stupid: it didn't
actually override the samplerate for Opus, but overrode it for all
codecs other than Opus. And even then, it failed to use the overridden
samplerate. (Sigh...)
Fixes relative seeks. Without this, a seek back could skip so much data
that the seek would effectively jump forward. (Or insert silence for
files with video.)
There's the question whether the frontend should do this instead (by
using information from the decoders), but for now this seems more
proper.
demux_mkv.c does this already, sort of.
libavformat doesn't for seeks in .ogg (aka .opus), but might be doing it
for mkv. Seems to be a mess as well.
I think the conclusion is that AV_PKT_DATA_SKIP_SAMPLES is misdesigned
(at least for some formats), and an alternative mechanism using
durations would be better. (Combining it with a proper timebase would
keep sample-accuracy.)
This is achieved indirectly by deslecting all streams for the non-
current segment (and if the segment doesn't share the demuxer with the
currently active one).
Restores functionality added with commit 46bcdb70.
This uses a different method to piece segments together. The old
approach basically changes to a new file (with a new start offset) any
time a segment ends. This meant waiting for audio/video end on segment
end, and then changing to the new segment all at once. It had a very
weird impact on the playback core, and some things (like truly gapless
segment transitions, or frame backstepping) just didn't work.
The new approach adds the demux_timeline pseudo-demuxer, which presents
an uniform packet stream from the many segments. This is pretty similar
to how ordered chapters are implemented everywhere else. It also reminds
of the FFmpeg concat pseudo-demuxer.
The "pure" version of this approach doesn't work though. Segments can
actually have different codec configurations (different extradata), and
subtitles are most likely broken too. (Subtitles have multiple corner
cases which break the pure stream-concatenation approach completely.)
To counter this, we do two things:
- Reinit the decoder with each segment. We go as far as allowing
concatenating files with completely different codecs for the sake
of EDL (which also uses the timeline infrastructure). A "lighter"
approach would try to make use of decoder mechanism to update e.g.
the extradata, but that seems fragile.
- Clip decoded data to segment boundaries. This is equivalent to
normal playback core mechanisms like hr-seek, but now the playback
core doesn't need to care about these things.
These two mechanisms are equivalent to what happened in the old
implementation, except they don't happen in the playback core anymore.
In other words, the playback core is completely relieved from timeline
implementation details. (Which honestly is exactly what I'm trying to
do here. I don't think ordered chapter behavior deserves improvement,
even if it's bad - but I want to get it out from the playback core.)
There is code duplication between audio and video decoder common code.
This is awful and could be shareable - but this will happen later.
Note that the audio path has some code to clip audio frames for the
purpose of codec preroll/gapless handling, but it's not shared as
sharing it would cause more pain than it would help.
FFmpeg can generate such files. It's unclear whether they're allowed by
Matroska. mkvinfo shows packet timestamps in both forms (one of them
must be a bug), and at last libavformat's demuxer treats timestamps
as signed.
This covers source files which were added in mplayer2 and mpv times
only, and where all code is covered by LGPL relicensing agreements.
There are probably more files to which this applies, but I'm being
conservative here.
A file named ao_sdl.c exists in MPlayer too, but the mpv one is a
complete rewrite, and was added some time after the original ao_sdl.c
was removed. The same applies to vo_sdl.c, for which the SDL2 API is
radically different in addition (MPlayer supports SDL 1.2 only).
common.c contains only code written by me. But common.h is a strange
case: although it originally was named mp_common.h and exists in MPlayer
too, by now it contains only definitions written by uau and me. The
exceptions are the CONTROL_ defines - thus not changing the license of
common.h yet.
codec_tags.c contained once large tables generated from MPlayer's
codecs.conf, but all of these tables were removed.
From demux_playlist.c I'm removing a code fragment from someone who was
not asked; this probably could be done later (see commit 15dccc37).
misc.c is a bit complicated to reason about (it was split off mplayer.c
and thus contains random functions out of this file), but actually all
functions have been added post-MPlayer. Except get_relative_time(),
which was written by uau, but looks similar to 3 different versions of
something similar in each of the Unix/win32/OSX timer source files. I'm
not sure what that means in regards to copyright, so I've just moved it
into another still-GPL source file for now.
screenshot.c once had some minor parts of MPlayer's vf_screenshot.c, but
they're all gone.
Slightly helps with timeline stuff, like EDL. There is no need to keep
network (or even just disk I/O) busy for all segments at the same time,
because 1. the data won't be needed any time soon, and 2. will probably
be discarded anyway if the stream is seeked when segment is resumed.
Partially fixes#2692.
demux_lavf.c leaked the complete subtitle data if it was put through
iconv.
lavc_conv.c leaked AVCodecContext.subtitle_header (set by libavcodec),
which is fixed by using avcodec_free_context(). It also leaked the
subtitle that was decoded last.
UTF-16 subtitles are special in that they are usually read by
libavformat directly, even though they are not in UTF-8. This is
explicitly handled convert_charset() and skips conversion to UTF-8.
There was a bug due to not resetting the file position: if conversion
happens, the actual stream is replaced with a memory stream containing
the converted data, but if conversion is skipped, the original stream
with the wrong file position is kept.
Fix by always opening a memory stream. (We _could_ seek back, but there
is a slight possibility of additional failure due to unseekable
streams.)
Also, don't enter conversion if the subtitle is detected as UTF-8
either.
Fixes#2700.
This is mainly a refactor. I'm hoping it will make some things easier
in the future due to cleanly separating codec metadata and stream
metadata.
Also, declare that the "codec" field can not be NULL anymore. demux.c
will set it to "" if it's NULL when added. This gets rid of a corner
case everything had to handle, but which rarely happened.
This slightly changes behavior when seeking with external audio/subtitle
tracks if transport streams and mpeg files are played, as well as
behavior when seeking with such external tracks.
get_main_demux_pts() is evil because it always blocks on the demuxer (if
there isn't already a packet queued). Thus it could lock up the player,
which is a shame because all other possible causes have been removed.
The reduced "precision" when seeking in the ts/mpeg cases (where
SEEK_FACTOR is used, resulting in byte seeks instead of timestamp seeks)
might lead to issues. We should probably drop this heuristic. (It was
introduced because there is no other way to seek in files with PTS
resets with libavformat, but its value is still questionable.)
There are a lot of incorrectly encoded subtitles with .ass extension
and non-ass subtitles (srt, ssa) with such extension, so we need to
try codepage detection even for .ass.
Signed-off-by: wm4 <wm4@nowhere>
Slightly change how it is decided when a new packet should be read.
Switch to demux_read_packet_async(), and let the player "wait properly"
until required subtitle packets arrive, instead of blocking everything.
Move distinguishing the cases of passive and active reading into the
demuxer, where it belongs.
Just so I can remove a few lines from dec_sub.c.
This is slightly inelegant, as the whole subtitle file has to be read
into memory, converted at once in memory, and then provided to
libavformat in an awkward way by creating a memory stream instead of
using demuxer->stream. It also won't be possible to force the charset on
subtitles in binary container formats - but this wasn't exposed before,
and we just hope this won't be ever needed. (One motivation was fixing
broken files with non-UTF8 muxed.) It also won't be possible to change
the charset on the fly, but this was not exposed either.
Always preroll by default if the cue (index) information indicates
overlapping subtitles.
Increase the amount of maximum data it will skip to get such subtitles
to 10 seconds. Since the index information can reliably tell whether
reading earlier is needed, the maximum should be rarely actually used,
thus we can set it high. On the other hand, the "old" prerolling
mechanism always has to skip the maximum amount of data; thus the method
using the index gets its own option to control the maximum amount of
data to skip.
(As more and more files With newer mkvtoolnix versions are muxed, and
with this new and hopefully sane default established, these options can
probably be removed in the future.)
Since commit 6d9cb893, subtitle state doesn't survive timeline switches
(ordered chapters etc.). So there is no point in caching the state per
sh_stream anymore (which would be required to deal with multiple
segments). Move the cache to struct track.
(Whether it's worth caching the subtitle state just for the situation
when subtitle tracks get reselected is questionable. But for now, it's
nice to have the subtitles immediately show up when reselecting a
subtitle.)
The demuxer infrastructure was originally single-threaded. To make it
suitable for multithreading (specifically, demuxing and decoding on
separate threads), some sort of tripple-buffering was introduced. There
are separate "struct demuxer" allocations. The demuxer thread sets the
state on d_thread. If anything changes, the state is copied to d_buffer
(the copy is protected by a lock), and the decoder thread is notified.
Then the decoder thread copies the state from d_buffer to d_user (again
while holding a lock). This avoids the need for locking in the
demuxer/decoder code itself (only demux.c needs an internal, "invisible"
lock.)
Remove the streams/num_streams fields from this tripple-buffering
schema. Move them to the internal struct, and protect them with the
internal lock. Use accessors for read access outside of demux.c.
Other than replacing all field accesses with accessors, this separates
allocating and adding sh_streams. This is needed to avoid race
conditions. Before this change, this was awkwardly handled by first
initializing the sh_stream, and then sending a stream change event. Now
the stream is allocated, then initialized, and then declared as
immutable and added (at which point it becomes visible to the decoder
thread immediately).
This change is useful for PR #2626. And eventually, we should probably
get entirely of the tripple buffering, and this makes a nice first step.
Commit 127da161 was not properly tested either - it did nothing, and
just made it use the video bitstream aspect ratio determined by
libavformat (which isn't always the correct one).
MPlayer traditionally always used the display aspect ratio, e.g. 16:9,
while FFmpeg uses the sample (aka pixel) aspect ratio.
Both have a bunch of advantages and disadvantages. Actually, it seems
using sample aspect ratio is generally nicer. The main reason for the
change is making mpv closer to how FFmpeg works in order to make life
easier. It's also nice that everything uses integer fractions instead
of floats now (except --video-aspect option/property).
Note that there is at least 1 user-visible change: vf_dsize now does
not set the display size, only the display aspect ratio. This is
because the image_params d_w/d_h fields did not just set the display
aspect, but also the size (except in encoding mode).
This can happen if the file references a track, but does not specify
an INDEX 01 for it. This would cause mpv to just segfault due to
dereferencing the null pointer as a string.
A file causing this was observed in the wild by
ExactAudioCopy v0.99pb4 for a disk that contained a data track at the
end.
Slightly simpler, and removes the need to pre-read all subtitle packets.
This still does the subtitle charset conversion on the packet level
(instead converting when parsing the file), so in theory this still
could provide a way to change the charset at runtime. But maybe even
this should be removed, as FFmpeg is somewhat likely to get its own
charset detection and conversion mechanism in the future. (Would have
to keep the subtitle file in memory to allow changing the charset on
the fly, I guess.)
All of these are supported by FFmpeg now. It was disabled by default
too (with FFmpeg).
If compiled against Libav, mpv will lose the ability to read some
subtitle formats (but the most important ones, srt and ass, still should
work).
This has no reason to be there. Put the functionality into another
function instead. While we're at it, also adjust for possible accuracy
issues with high bit depth YUV (matters for rendering subtitles into
screenshots only).
This is another regression of the recently added start time probing. If
a seek is executed after opening the file (but before reading any
packets), the first block is discarded instead of indexed. If there are
no other keyframes in the file, seeking will fail completely.
Fix it by seeking to the cluster start if there aren't any index entries
yet. This will read the skipped packet again.
Fixes#2498.
Most of this is explained in the DOCS additions.
This gives us slightly more sanity, because there is less interaction
between the various parts. The goal is getting rid of the video_offset
entirely.
The simplification extends to the user API. In particular, we don't need
to fix missing parts in the API, such as the lack for a seek command
that seeks relatively to the start time. All these things are now
transparent.
(If someone really wants to know the real timestamps/start time, new
properties would have to be added.)
This loaded external .ass files via libass. libavformat's .ass reader is
now good enough, so use that instead.
Apparently libavformat still doesn't support fonts embedded into text
.ass files, but support for this has been accidentally broken in mpv for
a while anyway. (And only 1 person complained.)
While it seemed like a pretty good idea at first, it's just a dead end
and works only in the simplest cases. While it may or may not help
slightly with audio sync mode, the display-sync mode already compensates
this in a better way. The main issue is that timestamps at this layer
are not in order, so it can look at single timestamps only.
av_free_packet() got finally deprecated. Use av_packet_unref() instead,
which has almost the same semantics, has existed for a while, and is
available in all FFmpeg and Libav versions we support.
This makes the bitrate properties unavailable, instead of
returning 0 when:
1. No track is selected, or
2. Not enough packets have been read to have a bitrate estimate yet
MKV files can very well start with timestamps other than 0. While mpv
has support for such files in general, and demux_lavf enables this
feature, demux_mkv didn't export a start time.
Implement this by simply reading the first cluster timestamp. This in
turn is done by reading 1 block. While we don't need the block for this
prupose at all, it's the easiest way to get the cluster timestamp read
correctly without code duplication. In theory this could be wrong, and
a packet could start at a much later time, but in practice this won't
happen.
This commit also adds an option to disable this feature. It's not
documented because nobody should use it. (But I happen to have a need
for this.)
This affects the subtitle preroll mode during seeking. It could matter
somewhat with insane files with ten-thousands of subtitle events, which
now seem to pop up, and will avoid packet queue overflow.
FFmpeg supports all formats the old subreader code does, and is better
at it. On the other hand, subreader.c's probing is bad and can lead to
false positives easily.
Make handling of metadata slightly more generic, and add reading of the
"PERFORMER" fields. There are some more fields, but for now let's leave
it at this.
TRACK-specific PERFORMER fields have to be read from the per-chapter
metadata (somewhat obscure).
Fixes#2328.
This AVPacket field was a hack against the fact that the duration field
was merely an int (too small for things like subtitle durations). Newer
libavcodec drops this field and makes duration 64 bit.
While unknown lengths are supported in some important cases like
segments and clusters, they are not for small and complex metadata
elements like the track list. Such elements are simply rejected.
This case was caught by the size sanity check below, but the message is
misleading and wrong.
(There are likely no files in the wild which require support for this.
The sample file I've seen was muxed by libavformat, but in a case where
it aborted when writing the header. Clearly a broken file.)
Add a simplistic heuristic for detecting broken indexes. This includes
indexes with very few elements (apparently libavformat sometimes writes
such indexes, or used to), and indexes with broken timestamps.
The latter was apparently produced by very old HandBrake versions:
| + Muxing application: libmkv 0.6.1.2
| + Writing application: HandBrake 0.9.1
These broken files seem to be common enough that libavformat added a
workaround for them in 2008 (and maybe again in 2015). Apparently all
timestamps are multiplied with the file's tc_scale twice, and FFmpeg
attempts to fix them. We should throw away the whole thing.
Actually, this never happened, because there's logic for ignoring
duplicate header elements (which includes the seek index). This is
mostly for robustness and readability.
Instead, allow reading 2KB only. This seems to be sufficient for
libarchive to recognize zip, 7z, rar, tar. Good enough.
This is implemented by creating an in-memory stream with a copy of
the file header. If libarchive succeeds opening this, the actual
stream is opened.
Allowing unlimited reading could break unseekable streams, such as
playing from http servers with no range request support or pipes.
Also, we try not to read too much data in the first probe pass. Some
slow network streams like shoutcast services could make probing much
slower if we allow it to read too much. In the second probing pass,
actually allow 200KB.
Things like .gz etc., which have no real file header. A mixed bag,
because it e.g. tends to misdetect mp3 files as compressed files or
something (of course it has no mp3 support - I don't know as what it
detects them). But requested by someone (or maybe not, I'm not sure
how to interpret that).
This works similar to the existing .rar support, but uses libarchive.
libarchive supports a number of formats, including zip and (most of)
rar.
Unfortunately, seeking does not work too well. Most libarchive readers
do not support seeking, so it's emulated by skipping data until the
target position. On backwards seek, the file is reopened. This works
fine on a local machine (and if the file is not too large), but will
perform not so well over network connection.
This is disabled by default for now. One reason is that we try
libarchive on every file we open, before trying libavformat, and I'm not
sure if I trust libarchive that much yet. Another reason is that this
breaks multivolume rar support. While libarchive supports seeking in
rar, and (probably) supports multivolume archive, our support of
libarchive (probably) does not. I don't care about multivolume rar, but
vocal users do.
Instead, force everyone to use the metadata struct and set a "title"
field. This is only a problem for the timeline producers, which set up
chapters manually. (They do this because a timeline is a separate
struct.)
This fixes the behavior of the chapter-metadata property, which never
returned a "title" property for e.g. ordered chapters.
This doesn't work too well if sections of the file change to a different
framerate. It lowers our chances to guess the correct FPS in the display
sync code.
For normal playback, this (probably) doesn't help that much anyway,
except that the "estimated-vf-fps" property will regress in the simplest
mkv case. This will be fixed with the next commit.
The now disabled code will probably be removed; it's not useful anymore.
Add --demuxer-max-packets and --demuxer-max-bytes, which control the
maximum size of the packet queue. These can be helpful to avoid
excessive memory usage.
Memory usage is the reason why there's a limit in the first place. If a
file is more or less broken, and audio and video don't line up, the
decoders will fill up the packet queue trying to read more audio or
video, and the maximum sizes are required to avoid unbounded memory
allocation. Being able to override the maximum sizes is useful; either
for restricting memory usage further, or enlarging the sizes when
attempting to play various broken files.
Remove --demuxer-readahead-packets and --demuxer-readahead-bytes. These
were a bit useless. They could force a minimum packet queue size, but
controlling the queue size with --demuxer-readahead-secs is much nicer.
It's fairly certain nobody ever used these options.
That just makes no sense, but seems to be a somewhat common user error.
The detection is not perfect. It's conceivable that EXT-X-... headers
are used in normal m3u playlists. After all, HLS playlists are by
definition a compatible extension to m3u playlists, as stupid as it
sounds.
Instead of opening a stream and then a demuxer, do both at once with
demux_open_url().
This requires some awkward additions to demuxer_params, because there
are some weird features associated with opening the main file. E.g. the
relatively useless --stream-capture features requires enabling capturing
on the stream before the demuxer is opened, but on the other hand
shouldn't be done on secondary files like external subtitles.
Also relatively bad: since demux_open_url() returns just a demuxer
pointer or NULL, additional error reporting is done via demuxer_params.
Still, at least conceptually, it's ok, and simpler than before.
Nobody wanted to restore this, so it gets the boot.
If anyone still wants to volunteer to restore menu support, this would
be welcome. (I might even try it myself if I feel masochistic and like
wasting a lot of time for nothing.) But if it does get restored, it
should be done differently. There were many stupid things about how it
was done. For example, it somehow tried to pull mp_nav_events through
all the layers (including needing to "buffer" them in the demuxer),
which was needlessly complicated. It could be done simpler.
This code was already inactive, so this commit actually changes nothing.
Also keep in mind that normal DVD/BD playback still works.
The user probably doesn't want these. Conveniently, this also skips the
unwanted "." and ".." entries.
(This code is triggered if the input stream is a directory - and it's in
demux_playlist.c because it's convenient.)
Handle a relatively recently introduced hack, that allows FLAC audio to
have arbitrary channel layouts, instead of just the predefined fixed
ones. This is actually supported by FFmpeg, but since the demuxer
(instead of the decoder) handles this in FFmpeg, we need to add special-
code to our mkv demuxer.
(The way FFmpeg does this seems a bit backwards, since now every demuxer
for a format that can handle FLAC needs to contain this logic as well.)
The FLAC hack is relatively terrible: we need to parse the FLAC headers,
look for a VorbisComment, parse the VorbisComment, and then retrieve
the magic WAVEFORMATEXTENSIBLE_CHANNEL_MASK entry. But the hack is
officially endorsed, as the official FLAC tools use it. (Although I
couldn't find a trace of it in the format specification. Should I be
surprised?)
Extend the --demuxer-mkv-probe-video-duration behavior to work with
files that are partial and are missing an index. Do this by finding a
cluster 10MB before the end of the file, and if that fails, just read
the entire file. This is actually pretty trivial to do and requires only
5 lines of code.
Also add a mode that always reads the entire file to estimate the video
duration.
Until now, if a stream wasn't seekable, but the stream cache was enabled
(--cache), we've enabled seeking anyway. The idea was that at least
short seeks would typically fall within the cache. And if not, the user
was out of luck and terrible things happened. In other words, it was
unreliable.
Be stricter about it and remove this behavior. Effectively, this will
for example disable seeking in piped data.
Instead of trying to be clever, add an --force-seekable option, which
will always enable seeking if the user really wants it.
If the EditionFlagOrdered is set, chapters without ChapterTimeEnd make
no sense. Ordered chapters will play the chapters in the order they
appear, but will play the ranges the chapters cover. So if the end time
is missing, the range is incomplete and it's not clear what should be
played. If you assume the start of the next chapter as end time, the
ordered flag will have no observable effect, so that's not a useful
assumption.
This fixes playback of a file which (apparently) had the
EditionFlagOrdered set accidentally, with normal chapters.
At least Matroska files have a "forced" flag (in addition to the
"default" flag). Export this flag. Treat it almost like the default
flag, but with slightly higher priority.
The "FrameRate" element is probably deprecated (it's greyed out in the
"spec", and described as "Informational only" in bold). Normally files
use DefaultDuration. In fact, the FrameRate field was preferred over
DefaultDuration for determining framerate if present. Do not do this and
rely on DefaultDuration only.
Also, if no framerate is set, do not assume PAL (25 FPS). Such a
fallback makes little sense and will cause more problems than it solves.
Integer and float elements are encoded as a sequence of bytes prefixed
by a variable-length encoded length specifier. If the length is 0, then
there is no data. Whether this is valid or not is not really clear, but
some sample files which do this have surfaced. It's not particularly
hard to handle this, so just do it.
Use char* for strings instead of bstr (data ptr + length pair). Matroska
actually (probably) allows "padding" strings with \0 bytes, so using
normal C strings instead of byte strings is more appropriate.
MPlayer traditionally had completely separate sh_ structs for
audio/video/subs, without a good way to share fields. This meant that
fields shared across all these headers had to be duplicated. This commit
deduplicates essentially the last remaining duplicated fields.
Always use the already existing extradata[_len] variable, instead of the
awkward switch between manually changed extradata and falling back to
passing through extradata at the end.
That's how mime types are.
(This makes redirection with a specific HLS URL work, because some idiot
thought it'd be a great idea to spell the mime type as
"application/x-mpegURL".)
The only decoders I could find and which (possibly) require this field
are codecs which can be used via VfW only, and realaudio sipr. For VfW
we still passthrough this field.
Native Matroska codec support has to map the Matroska codec IDs to
libavcodec ones, and also has to undo codec-specific Matroska
strangeness, such as restoring AAC extradata and realaudio handling. The
VfW codec support doesn't need it, because AVI maps well enough to
libavcodec conventions (possibly because AVI was a dominant codec when
libavcodec was created). But there's still some need for generic codec
handling, such as enabling parsers and messing with various codec
parameters.
Separate these two, and move the parts which are guaranteed not to be
needed by VfW to the if-else tree that handles the VfW case
("A_MS/ACM"), making the cases exclusive.
(This should probably be done more radically, since it's very unlikely
that we should or have to mess with the VfW parameters at all - they
should just be passed through to the decoder.)
This removes the last traces of the old MPlayer FourCC-based codec
mapping code. Forcing all codec IDs through a FourCC table and then
back to codec names was confusing at best, so this is a nice cleanup.
Handling of PCM (non-VfW case) is redone to some degree.
Handling of AC3 is moved below realaudio handling, since "A_REAL/DNET"
is apparently AC3, and we must not skip realaudio-specific handling.
(It seems unlikely that anything would actually break, but on the other
hand I don't have any A_REAL/DNET samples for testing.)
Instead of explicitly matching all the specific AAC codec names, just
match them all as prefix.
Some codecs don't need special handling other than their mapping
entries, so they fall away (like Vorbis and Opus).
The prores check in mkv_parse_and_add_packet() is not strictly related
to this, but is done for consistency with the wavpack check above.
The existing code avoided doing this for some codecs. I see no point in
this, and it seems the original reason this exists was due to some
cleanup in 2007. libavformat doesn't do this. So just drop it.
It's well possible that we've always ended up invoking the
AV_CODEC_ID_MPEG1VIDEO codec, but it's hard to tell. Mangling everything
through FourCCs (and then back) makes it hard to analyze. Also,
libavformat's Matroska demuxer uses AV_CODEC_ID_MPEG2VIDEO here, so it
should be quite safe to do anyway.
Inherited from MPlayer times, we used FourCCs to identify video codecs.
This was later changed to libavcodec codec names (which made life a
whole lot simpler). But demux_mkv still uses FourCCs a lot.
Change this for video. It's pretty simple, because some preparation was
done in the past. We just have to replace some "internal" FourCCs with
different handling.
One potentially complicated issue is that there is no natural way to
set the sh->format (AVCodecContext.codec_tag) field anymore. Most
decoders do not need it, though mjpeg is an exception.
Note that the AVI compatibility code still requires codec mappings, but
these are provided by FFmpeg. Also, the audio code is not changed.
For the MKV_V_MPEG2 -> mpeg1video thing see next commit.
Vobsubs come as .idx/.sub pair of files. The .idx file is the one that
should be opened, but the name of the .sub file is unknown. We can now
make our own guess what the name of that file is. In particular, improve
support with URLs (as these can have the file extension in the middle of
the filename string if there are HTTP parameters).
Note that this works only with newer ffmpeg versions, because the
recently added sub_name demuxer option is used for this.
This reads the "CUESHEET" tag, and attempts to parse it as .cue data. If
any is found, the cue tracks are added as chapters.
This reuses the parser written for demux_cue.c.
Fixes#1957.
The options don't change, but they're now declared and used privately by
demux_mkv.c. This also brings with it a minor refactor of the subpreroll
seek handling - merge the code from playloop.c into demux_mkv.c. The
change in demux.c is pretty much equivalent as well.
This change allows forward seeking even if there are no more video
keyframes in forward direction. This helps with files that e.g. encode
cover art as a single video frame (within a _real_ video stream - ffmpeg
seems to like to produce such files). Seeking backwards will still jump
to the nearest video frame, so this improvement has limited use.
The old code didn't do this because of the logic the min_diff variable
followed. Instead of somehow using the timestamp of the last packet read
for min_diff, use the first index entry for it. This actually makes it
fall back to the first/last index entry as the (removed) comment claims.
Note that last_pts is basically random at this point (because the
demuxer can be far ahead of playback position), so this didn't make
sense in the first place.
On EOF, this stopped reporting the actual cache duration, and just
signalled unknown duration. Fix this and keep reporting whatever is left
in the packet queue.
This reverts commit 5438a8b3. The commit doesn't give a good explanation
as to why it is needed, but I guess it was because the reporting was
imperfect (it switched between unknown or 0, and the correct duration).
This also removes a line added in commit 848546f2. The line is
ds->active = false;
The "active" flag basically says that data from this stream is actively
needed, and it's used to calculate the minimum data that can actually be
played (approximately). If this were ignored, a sparse subtitle stream
would set the cache duration to 0s. The commit message adding the line
says "actually does nothing, but in theory it's cleaner". Well, screw
it.
Remove the old implementation for these properties. It was never very
good, often returned very innaccurate values or just 0, and was static
even if the source was variable bitrate. Replace it with the
implementation of "packet-video-bitrate". Mark the "packet-..."
properties as deprecated. (The effective difference is different
formatting, and returning the raw value in bits instead of kilobits.)
Also extend the documentation a little.
It appears at least some decoders (sipr?) need the
AVCodecContext.bit_rate field set, so this one is still passed through.
If a directory is encountered, replace it with its contents in the
internal playlist.
This is messed into demux_playlist.c, because why not. STREAMTYPE_DIR
could be avoided by unconditonally trying opendir() in demux_playlist.c,
but it seems nicer not to do weird things like calling it on real files.
This does not work on Windows, because msvcrt is retarded.
We handle picking out font attachments by mime type ourselves in a
higher level, so we really just want to use the mimetype. Also, Matroska
is currently the only code in libavformat which uses the fonts at all,
and we can drop use of the codec IDs completely.
With a recent cleanup, rar support was stuffed into demux_playlist.c
(because "opening" rar files pretty much just lists archive contents and
adds them to a playlist using a special rar:// protocol, which will
actually access the rar file contents).
Since demux_playlist.c is probed _after_ demux_lavf.c (and should/must
be), libavformat was given the chance to detect DTS streams embedded
within the rar file. This is not really what we want, and a regression
what happened before rar listing was moved to demux_playlist.c.
Fix it by moving the rar listing into its own pseudo-demuxer, and let ir
probe before demux_lavf.c.
(Yes, this feature still has users.)
Trying to handle such video is almost worthless, but it was requested by
at least 2 users.
If there are no timestamps, enable byte seeking by setting
ts_resets_possible. Use the video FPS (wherever it comes from) and the
audio samplerate for timing. The latter was already done by making the
first packet emit DTS=0; remove this again and do it "properly" in a
higher level.
Check async abort notification. libavformat already do something
equivalent.
Before this commit, the demuxer could enter resync mode (and print silly
warning messages) when the stream stopped returning data because of an
abort.
A user reported a webm stream that couldn't be played. The issue was
that this stream 1. was on an unseekable HTTP connection, and 2. had a
SeekHead element (wtf?). The code reading the SeekHead marked the
element as unreadable too early: although you can't seek in the stream,
reading the header elements after the SeekHead read them anyway. Marking
them as unreadable only after the normal header reading fixes this.
(The way the failing stream was setup was pretty retarded: inserting
these SeekHead elements makes absolutely no sense for a stream that
cannot be seeked.)
Fixes#1656.
This reverts commit c8f49be919.
Not needed anymore; fixed in all supported FFmpeg releases. Though I
could not test again, because all sample files are gone (oops).
Use the (relatively new) libavformat image format probing functionality,
instead of letting demux_mf guess by file extension and MIME type.
The libavformat support is weird, though. Traditionally, it uses an
absolutely terrible hack to detect images by extension, _and_ (which is
the horrible part) will randomly interpret parts of the filename as
specifiers for matching by number. So something like '%03d' will be
interpreted as placeholder for a frame number. The worst part is that
such character sequences can be perfectly valid and common in http URLs.
This is known as "image2" demuxer. The newer support, which probes by
examining the file header, is split into several format-specific
demuxers with names ending in "_pipe". So we check for such a name
suffix. (At this point we're doing fine-grained hacking around ffmpeg
weirdness, so a clean solution is impossible anyway until upstream
changes.)
It was possible to make the player play local files by putting rar://
links into remote playlists, and some other potentially unsafe things.
Redo the handling of it. Now the rar-redirector (the thing in
demux_playlist.c) sets disable_safety, which makes the player open any
playlist entries returned. This is fine, because it redirects to the
same file anyway (just with different selection/interpretation of the
contents). On the other hand, rar:// itself is now considered fully
unsafe, which means that it is ignored if found in normal playlists.
This warning wasn't overly helpful in the past, and warned against
perfectly fine code. But at least with recent gcc versions, this is the
warning that complains about assignments in if expressions (why???), so
we want to enable it.
Also change all the code this warning complains about for no reason.
Refactors an older hack, which for some reason used a more complicated
way. This generates the playlist representing the contents of the rar
file in demux_playlist.c. The pseudo-demuxer could easily be separate
from the the playlist parsers (and in fact there's almost no shared
code), but I don't think this obscure feature deserves a separate file.
Sample files created with:
rar a -v20000k -m0 files.rar file1.mkv file1.mkv
If the cache is enabled, the demuxer is closed and opened again (because
currently, the cache can not be enabled atfer data was already read).
The call for opening a new demuxer uses the same params struct, which
references the ctx->uids array. But there is a MP_TARRAY_GROW()
invocation somewhere on the way, which can reallocate the ctx->uids
array, making params.uids a dangling pointer.
This issue probably existed for a longer time, probably since 5cd33853
(slightly more obvious since f50b105d).
Again removes some indirections and extra arguments.
Also replace some memcpy/memmoves with assignments. (Assignments became
possible only later, when reference UIDs were turned into a struct.)
Should behave about the same, but reduces code some duplication with
seeking and reading a header element pointed to by a SeekHead. It also
makes behavior with incomplete files slightly better.
Remove coded_width and coded_height. This was originally added in commit
fd7dde40, when BITMAPINFOHEADER was killed. The separate fields became
redundant in commit e68f4be1. Remove them (nothing passed to the
decoders actually changes with _this_ commit).
Some of the hacks were not applied if the file format was forced. Commit
37a0c914 moved them to a table, which is checked with normal probing
only.
Fixes#1612 (DVD forces mpeg, which in turn has to export native stream
IDs specifically).
Do some code restructuring on the way. For example, the probescore can
simply be set to the correct initial value, instead of checking whether
it was set at all.
Whatever the hell that is. FFmpeg tries to open any files with .bin file
extension with this demuxer (unless it finds a better demuxer), and then
reads the whole damn file, along with spamming dumb crap.
Includes some logic for not starting the demuxer thread for fully read
subtitles. (Well, the cache will still waste _lots_ of resources, and
the cache always has to be created, because we don't know whether it'll
be needed _before_ opening the file.)
See #1597.
An attempt to make format-specifics more declarative. (In my opinion,
all of this should be either provided by libavformat, or should not be
needed.)
I'm still leaving many checks with matches_avinputformat_name(), because
they're so specific.
Also useful for the following commit.
The Matroska timeline code was the only thing which still used the
demuxer.type field. This field explicitly identifies a demuxer
implementation. The purpose of the Matroska timeline code was to reject
files that are not Matroska. But it already forces the Matroska format,
meaning loading will explicitly only use the Matroska demuxer. If the
demuxer can't open the file, no other demuxer will be tried, and thus
checking the field is redundant.
The change in demux_mkv_timeline.c removes the if condition, and
unindents the if body.
Only demux_cue and demux_edl used it. It's a weird field and doesn't
help with anything anymore - by now, it only saves a priv context in the
mentioned demuxers. Reducing the number of confusing things the demuxer
struct has is more important than minimizing the code.
Move the implementation, of which most was in tl_cue.c, to demux_cue.c.
Currently, this is illogical, because tl_cue.c still accesses MPContext.
This is going to change, and then it will be better if everything is in
demux_cue.c. This is only a separate commit to distinguish code movement
and actual work; the next commit will do the actual work.
Instead of accessing MPContext in player/timeline/*, create a separate
context struct, which the timeline loaders fill out. It turns out that
there's not much in the way too big MPContext that these need to access.
One major PITA is managing (and closing) the set of open demuxers. The
problem is that we need a list of all demuxers to make sure no unneeded
streams are enabled.
This adds a callback to the demuxer_desc struct, with the intention of
leaving to to the demuxer to call the right loader, instead of
explicitly checking the demuxer type and dispatching manually in common
code. I also considered making the timeline part of the demuxer state,
but decided against: it's too much of a mess wrt. memory management and
threading, and also doesn't make it clear who owns the child demuxers.
With the struct timeline decoupled from the demuxer state, it's at least
somewhat clear that the child demuxers are independent from the "main"
demuxer.
The actual changes to player/timeline/* are separated in the following
commits, because they're quite verbose. Some artifacts will be removed
later as soon as there's only 1 timeline loading mechanism.
The HLs protocol consists of a "playlist" main file, which mpv downloads
and passes to the HLS demuxer. The HLS demuxer actually requests segment
files containing media data on its own. The packets read from the
demuxer have a source file position set, but it's not from the main
file. This leads to a strange effect: as a last fallback, the player
will calculate the approximate playback position from the file
position/size ratio, and since the main file is tiny, this will always
show 100%. Fix this by resetting the packet file position.
This doesn't affect the case when HLS actually reports a duration.
If the previous subtitle packet is too far back, and the refresh seek
won't pick it up, and the packet never comes again. As a consequence,
the refresh mode was never stopped on the subtitle stream, which caused
all packets to be discarded.
Fix by assuming the file position is monotonically increasing; then it
will resume even if a packet _after_ the intended resume point is
returned. This introduces a new requirement on how the demuxer behaves.
(I'm not sure if mp4 actually satisfies this requirement in all cases.)
Fixes a regression introduced by commit f9f2e1cc.
This removes the delay when switching audio tracks in mkv or mp4 files.
Other formats are not enabled, because it's not clear whether the
demuxers fulfill the requirements listed in demux.h. (Many formats
definitely do not with libavformat.)
Background:
The demuxer packet cache buffers a certain amount of packets. This
includes only packets from selected streams. We discard packets from
other streams for various reasons. This introduces a problem: switching
to a different audio track introduces a delay. The delay is as big as
the demuxer packet cache buffer, because while the file was read ahead
to fill the packet buffer, the process of reading packets also discarded
all packets from the previously not selected audio stream. Once the
remaining packet buffer has been played, new audio packets are available
and you hear audio again.
We could probably just not discard packets from unselected streams. But
this would require additional memory and CPU resources, and also it's
hard to tell when packets from unused streams should be discarded (we
don't want to keep them forever; it'd be a memory leak).
We could also issue a player hr-seek to the current playback position,
which would solve the problem in 1 line of code or so. But this can be
rather slow.
So what we do in this commit instead is: we just seek back to the
position where our current packet buffer starts, and start demuxing from
this position again. This way we can get the "past" packets for the
newly selected stream. For streams which were already selected the
packets are simply discarded until the previous position is reached
again.
That latter part is the hard part. We really want to skip packets
exactly until the position where we left off previously, or we will skip
packets or feed packets to the decoder twice. If we assume that the
demuxer is deterministic (returns exactly the same packets after a seek
to a previous position), then we can try to check whether it's the same
packet as the one at the end of the packet buffer. If it is, we know
that the packet after it is where we left off last time.
Unfortunately, this is not very robust, and maybe it can't be made
robust. Currently we use the demux_packet.pos field as unique packet
ID - which works fine in some scenarios, but will break in arbitrary
ways if the basic requirement to the demuxer (as listed in the demux.h
additions) are broken. Thus, this is enabled only for the internal mkv
demuxer and the libavformat mp4 demuxer.
(libavformat mkv does not work, because the packet positions are not
unique. Probably could be fixed upstream, but it's not clear whether
it's a bug or a feature.)
Until now, some packets could return the same file position if they were
split off from a Matroska-level packet. This was perfectly fine, because
the file position isn't used for anything overly important (it uses it
to estimate playback position if no other information is available). The
following commit will use the demux_packet.pos field as unique ID (as a
simplification), so make the demuxer export more finegrained
information.
Also, the last_filepos field didn't have to be global, at least not
anymore.
Reindent the whole handle_realaudio() function, and make the surrouding
if block return early instead.
Also contains some cosmetics to the sipr swapping, which hopefully does
not change the semantics, but is untested (the kind of cosmetic changes
everyone loves so much). May the person responsible for sipr rot in
hell. (It was probably done to obfuscate the codec?)
Staring at the code, it doesn't look like the extra code for "normal"
audio is needed. Most of it looks like artifacts from the previous code
structure (much of it was added in the initial commit). I couldn't find
a sample that uses this code path to fully confirm this, though.
I suppose it could lead to subtle changes in behavior in presence of
realvideo files that change aspect radio. With the only sample I had
available, the behavior actually improved (azumi.mkv from the MPlayer
samples FTP; when starting playback in the middle it used the wrong
aspect ratio).
Appears to work, so we can drop some code. For some really odd reason,
the descrambling done on the timestamp requires millisecond units (due
to the "algorithm", not the libavcodec API).
Fixes vp9 missing timestamps. This requires a brand new libavcodec (the
patch for this was just applied to FFmpeg git master).
The timestamp mangling is applied to VP9 only. It'd probably work with
other codecs, but it's not needed. It could break in various ways, so
it has to be explicitly checked for every enabled codec.
Makes it somewhat more uniform, and breaks up the awfully deep nesting.
This implicitly changes multiple small details, rather than only moving
code around. In particular, this computes the packet fields first and
parses them afterwards, which is needed for the next commit.
Currently, audio packets are always filtered as a whole. Since demux_raw
output a 1 second long packet, this could lead to large delays when
applying softvol volume. It could be fixed by splitting the frames the
decoder outputs before filtering them (like the old filter code used
to), but since this didn't cause any other problems yet, I'm going with
the simpler fix.
Fixes#1558.
The only reason why cdda:// goes through this wrapper-demuxer is so that
we add chapters to it. Most things related to seeking apply only to
DVD/BD, and in fact broke CDDA sekkability.
Fixes#1555.
Might fix behavior with mkv files that use ordered chapters and have
cover art tags. In my opinion, this should actually have worked (because
cover art pseudo-tracks are strictly appended), but I don't have a
sample file to test at hand.
If a file is unseekable (consider e.g. a http server without resume
functionality), but the stream cache is active, the player will enable
seeking anyway. Until know, client API user couldn't know that this
happens, and it has implications on how well seeking will work. So add a
property which exports whether this situation applies.
Fixes#1522.
Repurpose demuxer->filetype for this. It used to be used to print a
human readable format description; change it to a symbolic format name
and export it as property.
Unfortunately, libavformat has its own weird conventions, which are
reflected through the new property, e.g. the .mp4 case mentioned in the
manpage.
Fixes#1504.
Pass through the seek flags to the stream layer. The STREAM_CTRL
semantics become a bit awkward, but that's still the least awkward
part about optical disc media.
Make demux_disc.c request relative seeks. Now the player will use
relative seeks if the user sends relative seek commands, and the
demuxer announces it wants these by setting rel_seeks to true. This
change probably changes seek behavior for dvd, dvdnav, bluray, cdda,
and possibly makes seeking useless if the demuxer-cache is set to
a high value.
Will be used in the next commit. (Split to make reverting the next
commit easier.)
Normally the player doesn't read from unselected streams, so this should
be a no-op. But unfortunately, some broken files can severely confuse
the player, and assign the same demuxer stream to multiple front-end
tracks. Then selecting one of the tracks would deselect the other track,
with the end result that the demuxer stream for the selected track is
deselected. This could happen with mkv files that use the same track
number (which is of course broken). timeline_set_part() sets the tracks
using demuxer_stream_by_demuxer_id(), using the broken non-unique IDs.
The observable effect was that the player never quit, because
demux_read_packet_async() told the caller to wait some longer for new
packets. Fix by returning EOF instead.
Fixes#1481.
Reading IDs must be checked too. This was basically forgotten in commit
f3a978cd. Also set the *length parameter for ebml_parse_length() in some
error cases, which _really_ should happen.
Fixes#1461.
Apparently, originally this code was meant to be able to read past the
buffer somewhat, which is why the buffer allocation was padded by 8
byte. This is unclean and confuses valgrind. This probably could have
crashed with certain invalid files too.
Also revert the change added with 10a2f69; it should be not needed
anymore.
The VP9 codec parser has a bug: it doesn't set the data/size pointers
passed to it. As I understand, it must always do this, and in fact, if
it doesn't some libavcodec generic code would be in trouble too.
This helps with #1448, but is not the full fix for it. The codec parser
must be fixed in libavcodec itself.
Removes an annoying "No video PTS! Making something up." warning.
Mark it as keyframe, which is needed to prevent strange behavior with
PNG. Also, don't leak the picture data.
For some codecs, we need to invoke a codec parser (because libavcodec
will run into trouble otherwise). This was done based on the Matroska
codec field.
But this ignores handling of vfw-muxed files, which use a pseudo-codec
to signal presence of vfw structures, which we must unmangle to get the
real codec. Handle this by rearranging the code.
This fixes at least mp3-in-mkv for vfw-muxed files; typically old files.
