Make the VF/VO/AO option parser available to audio filters. No audio
filter uses this yet, but it's still a quite intrusive change.
In particular, the commands for manipulating filters at runtime
completely change. We delete the old code, and use the same
infrastructure as for video filters. (This forces complete
reinitialization of the filter chain, which hopefully isn't a problem
for any use cases. The old code forced reinitialization too, but it
could potentially allow a filter to cache things; e.g. consider loaded
ladspa plugins and such.)
Doing "mpv --vo=opengl:lscale=help" now lists possible scalers and
exits. The "backend" suboption behaves similar. Make the "stereo"
suboption a choice, instead of using magic integer values.
These two options were supported by ALSA and OSS only. Further, their
values were specific to the respective audio systems, so it doesn't make
sense to keep them as top-level options.
This changes how device names are handled. Before this commit, device
names were mangled in strange ways to avoid clashing with the option
parser syntax. "." was replaced with ",", and "=" with ":" (the user had
to do the inverse to get the correct device name).
The "new" option parser has multiple ways to escape option strings, so
we don't need this confusing hack anymore.
Add an explicit note to the manpage as well.
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.
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.
Add this option, which lets users set the cache size without forcing it
even when playing from the local filesystem.
Also document the default value explicitly.
The Matroska linked segments case is slightly simplified: they can
never come from network (mostly because it'd be insane, and we can't
even list files from network sources), so the cache will never be
enabled automatically.
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_vstream.c in particular was actually dependent on the network
code, and didn't compile anymore.
Cleanup the protocol list in mpv.rst, and add some missing ones
supported by libavformat to stream_lavf.c.
This commit removes the "old" networking code in favor of libavformat's
code.
The code was still used for mp_http, udp, ftp, cddb. http has been
mapped to libavformat's http support since approximately 6 months ago.
udp and ftp have support in ffmpeg (though ftp was added only last
month). cddb support is removed with this commit - it's probably not
important and rarely used if at all, so we don't care about it.
percent-pos was an integer (0-100). Sometimes higher precision is
wanted, but the property is this way because fractional parts would
look silly with normal OSD usage. As a compromise, make percent-pos
double (i.e. includes fractional parts), but print it as integer.
So ${percent-pos} is like an integer, but not ${=percent-pos}.
Use the video decoder chroma location flags and render chroma locations
other than centered. Until now, we've always used the intuitive and
obvious centered chroma location, but H.264 uses something else.
FFmpeg provides a small overview in libavcodec/avcodec.h:
-----------
/**
* X X 3 4 X X are luma samples,
* 1 2 1-6 are possible chroma positions
* X X 5 6 X 0 is undefined/unknown position
*/
enum AVChromaLocation{
AVCHROMA_LOC_UNSPECIFIED = 0,
AVCHROMA_LOC_LEFT = 1, ///< mpeg2/4, h264 default
AVCHROMA_LOC_CENTER = 2, ///< mpeg1, jpeg, h263
AVCHROMA_LOC_TOPLEFT = 3, ///< DV
AVCHROMA_LOC_TOP = 4,
AVCHROMA_LOC_BOTTOMLEFT = 5,
AVCHROMA_LOC_BOTTOM = 6,
AVCHROMA_LOC_NB , ///< Not part of ABI
};
-----------
The visual difference is literally minimal, but since videophiles
apparently consider this detail as quality mark of a video renderer,
support it anyway. We don't bother with chroma locations other than
centered and left, though.
Not sure about correctness, but it's probably ok.
This adds support for libquvi 0.9.x, and these features:
- start time (part of youtube URL)
- youtube subtitles
- alternative source switching ('l' and 'L' keys)
- youtube playlists
Note that libquvi 0.9 is still in development. Although this seems to
be API stable now, it looks like there will be a 1.0 release, which is
supposed to be the next stable release and the actual successor of
libquvi 0.4.x.
Should we actually get into trouble for unproper handling of
frame-based subtitle formats, this might be the simplest way to
work this around. Also is a bit more intuitive than -subfps, which
might use an unknown, misdetected, or non-sense video FPS.
Still pretty silly, though.
This code was once part of subreader.c, then traveled to libass, and now
made its way back to the fork of the fork of the original code, MPlayer.
It works pretty much the same as subreader.c, except that we have to
concatenate some packets to do auto-detection. This is rather annoying,
but for all we know the actual source file could be a binary format.
Unlike subreader.c, the iconv context is reopened on each packet. This
is simpler, and with respect to multibyte encodings, more robust.
Reopening is probably not a very fast, but I suspect subtitle charset
conversion is not an operation that happens often or has to be fast.
Also, this auto-detection is disabled for microdvd - this is the only
format we know that has binary data in its packets, but is actually
decoded to text. FFmpeg doesn't really allow us to solve this properly,
because a) the input packets can be binary, and b) the output will be
checked whether it's UTF-8, and if it's not, the output is thrown away
and an error message is printed. We could just recode the decoded
subtitles before sd_ass if it weren't for that.
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.
This fixes the -subfps option (which unfortunately is still useful),
and fixes minor annoying timing errors (which unfortunately still
happen).
Note that none of these affect ASS or image subtitles. ASS is specially
handled: libass loads subtitles as ASS_Track. There are no actual
packets passed around, and sd_ass just uses the ASS_Track.
Disable the --sub-no-text-pp option. It's misleading now and always was
completely useless.
Basically rewrite all the code supporting the cache (i.e. anything other
than the ringbuffer logic). The underlying design is untouched.
Note that the old cache2.c (on which this code is based) already had a
threading implementation. This was mostly unused on Linux, and had some
problems, such as using shared volatile variables for communication and
uninterruptible timeouts, instead of using locks for synchronization.
This commit does use proper locking, while still retaining the way the
old cache worked. It's basically a big refactor.
Simplify the code too. Since we don't need to copy stream ctrl args
anymore (we're always guaranteed a shared address space now), lots of
annoying code just goes away. Likewise, we don't need to care about
sector sizes. The cache uses the high-level stream API to read from
other streams, and sector sizes are handled transparently.
