| Commit message (Collapse) | Author | Age | Files | Lines |
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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.
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Before this commit, the cache was franken-hacked on top of the stream
API. You had to use special functions (like cache_stream_fill_buffer()
instead of stream_fill_buffer()), which would access the stream in a
cached manner.
The whole idea about the previous design was that the cache runs in a
thread or in a forked process, while the cache awa functions made sure
the stream instance looked consistent to the user. If you used the
normal functions instead of the special ones while the cache was
running, you were out of luck.
Make it a bit more reasonable by turning the cache into a stream on its
own. This makes it behave exactly like a normal stream. The stream
callbacks call into the original (uncached) stream to do work. No
special cache functions or redirections are needed. The only different
thing about cache streams is that they are created by special functions,
instead of being part of the auto_open_streams[] array.
To make things simpler, remove the threading implementation, which was
messed into the code. The threading code could perhaps be kept, but I
don't really want to have to worry about this special case. A proper
threaded implementation will be added later.
Remove the cache enabling code from stream_radio.c. Since enabling the
cache involves replacing the old stream with a new one, the code as-is
can't be kept. It would be easily possible to enable the cache by
requesting a cache size (which is also much simpler). But nobody uses
stream_radio.c and I can't even test this thing, and the cache is
probably not really important for it either.
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Some code in mplayer.c did stuff like accessing (dvd_priv_t *)st->priv.
Do this indirectly by introducing STREAM_CTRL_GET_DVD_INFO. This is
extremely specific to DVD, so it's not worth abstracting this further.
This is a preparation for turning the cache into an actual stream, which
simply wraps the cached stream. There are other streams which are
accessed in the way DVD was, at least TV/radio/DVB. We assume these
can't be used with the cache. The code doesn't look thread-safe or fork
aware.
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The core didn't use these fields, and use of them was inconsistent
accross AOs. Some didn't use them at all. Some only set them; the values
were completely unused by the core. Some made full use of them.
Remove these fields. In places where they are still needed, make them
private AO state.
Remove the --abs option. It set the buffer size for ao_oss and ao_dsound
(being ignored by all other AOs), and was already marked as obsolete. If
it turns out that it's still needed for ao_oss or ao_dsound, their
default buffer sizes could be adjusted, and if even that doesn't help,
AO suboptions could be added in these cases.
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Currently every single AO was implementing it's own ringbuffer, many times
with slightly different semantics. This is an attempt to fix the problem.
I stole some good ideas from ao_portaudio's ringbuffer and went from there.
The main difference is this one stores wpos and rpos which are absolute
positions in an "infinite" buffer. To find the actual position for writing /
reading just apply modulo size.
The producer only modifies wpos while the consumer only modifies rpos. This
makes it pretty easy to reason about and make the operations thread safe by
using barriers (thread safety is guaranteed only in the Single-Producer/Single-
Consumer case).
Also adapted ao_coreaudio to use this ringbuffer.
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This allows having properties like time-pos in the window title update
properly. There is a danger of this causing significant CPU usage,
depending on the properties used and the window manager.
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Instead of implicitly changing the window title on config(), do it as
part of the new VOCTRL.
At first I wanted to make all VOs use the VOCTRL argument directly, but
on a second thought it appears vo_get_window_title() is much more useful
for some (namely, if the window is created lazily on first config()).
Not all VOs are changed. Wayland and OSX have to follow.
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%k is not very standard. The manpage notes them as conforming to
"Olson's timezone package", and it's not standard C89, C99 or POSIX.
mingw doesn't provide it, and even some of the smaller Linux libcs
don't have support.
Use %H instead. This gives slightly different results, but I think
this is ok. Difference in behavior between these summarized:
%k: "single digits are preceded by a blank"
%H: "range 00 to 23"
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Use the recently introduced screensaver VOCTRLs to control the
screensaver in the X11 backend. This means the behavior when paused
changes: the old code always kept the screensaver disabled, but now the
screensaver is reenabled on pausing.
Rename the --stop-xscreensaver option to --stop-screensaver and make it
more generic. Now it affects all backends that respond to the
screensaver VOCTRLs.
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This is slightly better because VOCTRL_RESUME/VOCTRL_PAUSE are usually
needed by VOs to know whether video is actually being played (for
whatever reason), and they wouldn't be passed to the backend's VOCTRL
handler, like vo_x11_control().
Also try to make sure that these flags (both pause state and screensaver
state) are set consistently in some corner cases. For example, it seems
enabling video in the middle of playing a file while the player is
paused would not set the paused flag.
If codec initialization fails, destroy the VO instead of keeping it
around to make sure the state is consistent.
Framestepping is implemented by unpausing the player for the duration of
a frame. Remove the special handling of VOCTRL_PAUSE/RESUME in these
cases. It was most likely needed because these VOCTRLs used to be
important for screen redrawing (blatant guess), which is now handled
completely differently. The only potentially bad side-effect is that the
screensaver will be disabled/reenabled for the duration of one frame.
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This workaround sucks. avio just does not support "-" - and ffmpeg's
command line binaries work around it. FOUR TIMES. DIFFERENTLY.
WHY DOESN'T AVIO DO THIS RIGHT TO BEGIN WITH?
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Whatever this was supposed to be originally, it doesn't have much value
anymore. It just forced ad_mpg123 to upmix mono to stereo by default
(the audio chain can do that). As an option, it was mostly useless and
misleading, so get rid of it.
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Raw MPEG streams can contain PTS discontinuities. While the playback
core has obvious code for handling PTS going backward, PTS going
forward was as far as I can see not handled.
This can be an issue with DVD playback. This wasn't caught earlier,
because DVD playback was just recently switched to demux_lavf, which
implies -no-correct-pts mode. This mode doesn't use PTS in the same way
as the normal playback mode, and as such was too primitive to be
affected by this issue.
Use the following heuristic to handle PTS forward jumps: if the PTS
difference between two frames is higher than 10 seconds, consider it a
reset. (Also, use MSGL_WARN for the PTS discontinuity warnings.)
In this particular case, the MPEG stream was going from pts=304510857
to pts=8589959849 according to ffprobe (raw timestamps), which seems a
bit strange.
Note that this heuristic breaks if the source video has unusually high
frame times. For example Rooster_Teeth_Podcast_191.m4a, an audio file
with a slide show encoded as MJPEG video track.
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Closes github issue #107. (Hopefully.)
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show_text "${metadata}" crashed if no metadata was set.
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Lower-fps content is left alone (NOT aligned to this fps); higher fps
content is decimated to this frame rate.
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This also removes the split between "mplayer" and "common" opts (common
opts used to be shared between mencoder and mplayer).
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There isn't really any reason why this should be in a separate source
file.
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show_chapters, show_tracks, and show_playlist are killed and replaced
with the properties chapter-list, track-list, and playlist. The code
and the output of these stays the same, this is just moving a lot of
code around and reducing the number of properties.
The "old" commands will still be supported for a while (to avoid making
everyone angry), so handle them with the legacy layer. Add something to
suppress printing the legacy warnings for these commands.
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Slightly better output when printing ${metadata}. Print each metadata
item as "name: value", instead of the raw list. It's still not very
great, though. The old format is still available through ${=metadata}
for things which dare to use the broken slave mode.
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Apparently this behavior is more intuitive/better to users.
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There's no reason why this should be forbidden.
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Makes 213ad5d behave has intended.
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This adds a the property 'clock', which returns the current
local time as the string hh:mm.
Additionally the keybinding 'shift' + 'o' was added to displaying
the clock as '[hh:mm]' .
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This commit addresses some issues with the users had with the previous
implementation in commit c39efb9. Here's the changes:
* Use Quartz Event Taps to remove Media Key events mpv handles from
the global OS X queue. This prevents conflicts with iTunes. I did this on
the main thread since it is mostly idling. It's the playloop thread that
actually does all the work so there is no danger of blocking the event tap
callback.
* Introduce `--no-media-keys` switch so that users can disable all of mpv's
media key handling at runtime (some prefer iTunes for example).
* Use mpv's bindings so that users can customize what the media keys do via
input.conf. Current bindings are:
MK_PLAY cycle pause
MK_PREV playlist_prev
MK_NEXT playlist_next
An additional benefit of this implementation is that it is completly handled
by the `macosx_events` file instead of `macosx_application` making the
project organization more straightforward.
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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.)
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Until now, this happened only when the -no-ass option was used. This
difference in behavior doesn't make much sense, so change it so that
whether -no-ass is used or not doesn't matter. (-no-ass enables the OSD
subtitle renderer, which has the terminal fallback, while the normal
path is video only.)
the changes in set_osd_subtitle() and reinit_video_chain() are for
resetting the state correctly when switching between video/no-video.
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Audio and video had their own (very similar) functions to initialize an
AVPacket (ffmpeg's packet struct) from a demux_packet (mplayer's packet
struct). Add a common function for these.
Also use this function for sd_lavc_conv. This is actually a functional
change, as some libavfilter subtitle demuxers add weird out-of-band
stuff as side-data.
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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.
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The -no-ass option used to disable all use of libass completely. This
doesn't work this way anymore, and the text subtitle path has an
inherent dependency on libass. Currently -no-ass does 3 things:
1. Strip tags and formatting on display, and use a separate renderer for
the result. (Which might be the terminal, or libass via OSD code.)
2. Not loading attached fonts from Matroska files.
3. Use subreader.c instead of libass for reading .ass files.
1. and 2. are ok and what the user (probably wants), but 3. doesn't
really make sense anymore. subreader.c reads .ass files just fine, but
then does some strange things to them (something about coalescing and
re-adding newlines?), leading to even more broken display with -no-ass.
Instead of fighting with subreader.c, just use libass as loader.
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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.
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Add a basic infrastructure for subtitle converters. These converters
work sort-of like decoders, except that they produce packets instead
of subtitle bitmaps. They are put in front of actual decoders.
Start with sd_movtext. 4 lines of code are blown up to a 55 lines file,
but fortunately this is not going to be that bad for the following
converters.
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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.
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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.
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Just pushing some code around.
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