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?)
Kevin Mitchell
wm4