| Commit message (Collapse) | Author | Age | Files | Lines |
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Whoever thought this was a good idea should be punched.
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Also remove the ability to disable deinterlacing at runtime. You can
still disable deinterlacing at runtime by using the ``D`` key and its
automatical filter insertion/removal.
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As discussed on IRC.
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The "run" input command does fork+exec to spawn child processes. But it
doesn't cleanup the child processes, so they are left as zombies until
mpv terminates. Leaving zombie processes around is not very nice, so
employ a simple trick to let pid 1 take care of this: we fork twice, and
when the first fork exits, the second fork becomes orphaned and becomes
pid 1's child. It becomes pid 1's responsibility to cleanup the process.
The advantage is that we don't need extra logic to cleanup the spawned
process, which could have an arbitrary lifetime.
This is e.g. described here: http://yarchive.net/comp/zombie_process.html
Also use _exit() instead of exit(). It's not really sane to run cleanup
handlers (atexit() etc.) inside a forked process.
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This is needed so that new processes (created with fork+exec) don't
inherit open files, which can be important for a number of reasons.
Since O_CLOEXEC is relatively new (POSIX.1-2008, before that Linux
specific), we #define it to 0 in io.h to prevent compilation errors on
older/crappy systems. At least this is the plan.
input.c creates a pipe. For that, add a mp_set_cloexec() function (which
is based on Weston's code in vo_wayland.c, but more correct). We could
use pipe2() instead, but that is Linux specific. Technically, we have a
race condition, but it won't matter.
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See the changes in input.rst for explanations.
Technically speaking, this also gets rid of some undefined behavior:
passing NULL as a vararg (execl()) is always a bug.
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Makes packaging a bit simpler.
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These were confined to the video path, but resetting them even if no
video is available shouldn't really matter. Always resetting them makes
the logic easier to follow, I think.
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Recently, the check was moved, so it was printed only for source video
PTS (since that's easier, and filters should normally behave sane). But
it turns out it's trivial to print a warning in the filter case too by
reusing the code that normally checks for PTS forward jumps without
needing any additional code, so, fine, restore warning in this case.
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Feature request from github issue #376.
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Introduce a mp_cmd_def struct to define commands, instead of using
mp_cmd for this. This way each command parameter can be a m_option,
instead of m_option plus some more stuff.
Define the ARG_ macros directly in terms of the OPT_ macros. Not sure if
this makes it easier to read (maybe not, even if it looks simpler), but
at least it makes it easier to add other option types.
Another idea was adding a name for each parameter (so you could have
named parameters), but not today.
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The hr-seek code assumes that when seeking the demuxer, the first image
decoded after the seek will have a PTS exactly equal to the demuxer seek
target time, or before that target time. Incorrect timestamps,
implicitly dropped initial frames, or broken files/demuxers can all
break this assumption, and lead to hr-seek missing the seek target.
Generally, this is not much a problem (the user won't notice being off
by one frame), but it really shows when using the backstep feature. In
this case, backstepping would simply hang.
Add a simple hack that basically forces a minimal value for the --hr-
seek-demuxer-offset option (which is 0 by default) when doing a
backstep-seek. The chosen minimum value is arbitrary. There's no perfect
value, though in general it should perhaps be slightly longer than the
frametime, which the chosen value is more than enough for typical
framerates.
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Using --start with files that use DTS only, or which simply have broken
PTS timestamps, would incorrectly drop frames and possibly not execute
the seek correctly.
Add yet another heuristic to detect this. The intent is that --start and
hr-seeks in general should work correctly, but in order to keep things
fast, we still want to allow frame dropping during hr-seek if there are
no problems doing so. Do this by disabling frame dropping by default,
but re-enabling it if there are no problems found for a while. As a
consequence, --start might be somewhat slower, but normal user
interaction should remain as fast as before.
Note that there's something subtle about the added code: the
has_broken_packet_pts field is checked even before the first packet is
fed to dec_video.c, so the field must not be set to 0 right on start.
It's not initially set to 0 anyway, because the heuristic requires
decoding some images before enabling frame drop anyway.
Note 2: it's not clear whether frame dropping during hr-seek really
helps; I didn't benchmark it.
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The d_video->pts field was a bit strange. The code overwrote it multiple
times (on decoding, on filtering, then once again...), and it wasn't
really clear what purpose this field had exactly. Replace it with the
mpctx->video_next_pts field, which is relatively unambiguous.
Move the decreasing PTS check to dec_video.c. This means it acts on
decoder output, not on filter output. (Just like in the previous commit,
assume the filter chain is sane.) Drop the jitter vs. reset semantics;
the dec_video.c determined PTS never goes backwards, and demuxer
timestamps don't "jitter".
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Also get rid of the PTS check _after_ filters. This means if there's a
video filter which unsets PTS, no warning will be printed. But we assume
that all filters are well-behaved enough by now.
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These used the suffix _resync_stream, which is a bit misleading. Nothing
gets "resynchronized", they really just reset state.
(Some audio decoders actually used to "resync" by reading packets for
resuming playback, but that's not the case anymore.)
Also move the function in dec_video.c to the top of the file.
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This used a really weird idiom: a loop that iterates only once, so you
can use break; to jump out of the block.
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These packets have to be explicitly dropped, because usually libavcodec
uses 0-sized packets to flush delayed frames, meaning just passing
through these packets would have bad consequences.
Normally, libavformat doesn't output 0-sized packets anyway. But I don't
want to take any chances, so don't delete it, and just move it out of
the way to demux.c.
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Solve this by passing through the language to the player, which then
uses the generic subtitle selection code to make a choice.
Fixes #367.
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If the value for --cache-on-pause is larger than --cache-min, and the
cache runs below --cache-on-pause, but above --cache-min, the logic
would demand to pause the player and then unpause immediately again.
This doesn't make much sense, and alternating the pause state in each
playloop iteration has negative consequences. Add an explicit check to
avoid this situation.
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This means the code that tries to figure out the timestamp from
demuxer and decoder output is now all in dec_video.c. We set the
final timestamp on the returned image (mp_image.pts), as well as
the d_video->pts field.
The way the player uses d_video->pts field is still a bit messy. Maybe
this could be cleaned up later.
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Having the DTS directly can be useful for restoring PTS values.
The avi file format doesn't actually store PTS values, just DTS. An
older hack explicitly exported the DTS as PTS (ignoring the [I assume]
genpts generated non-sense PTS), which is not necessary anymore due to
this change.
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Now the --no-correct-pts mode is like the normal mode, just with
different timestamp calculations. The semantics should be about the
same as before this commit.
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Before this commit, this mode estimated the frame time by subtracting
successive packet PTS values. This is complete non-sense for video
codecs which use reordering. The code compensated frame times for these
non-sense using the FPS value, but confused the rest of the player with
non-sense jumping around timestamps. So, all in all this mode is not
very useful.
Repurpose this mode for fixed frame rate playback. This gives almost the
same behavior as the old mode with forced framerate (--fps option). The
result is simpler and often more robust.
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Instead of passing the PTS as separate field, pass it as part of the
usual data structures. Basically, this removes strange artifacts from
the API. (It's not finished, though: the final decoded PTS goes through
strange paths, and filter_video() finally overwrites the decoded
mp_image's pts field with it.)
We also stop using libavcodec's reordered_opaque fields, and use
AVPacket.pts and AVFrame.pkt_pts. This is slightly unorthodox, because
these pts fields are not "really" opaque anymore, yet we treat them as
such. But the end result should be the same, and reordered_opaque is
marked as partially deprecated (it's not clear whether it's really
deprecated).
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This normally shouldn't happen. It does happen with VfW-muxed mkv files,
and would normally happen with avi files (except that we force the
correct association mode using an explicit hack for avi).
This is usually prepended by warnings like:
Decreasing video pts: 0.125000 < 0.167000
and after the switch, there should be no warnings anymore.
Background: avi likes to use DTS for timestamps instead of PTS.
Basically, there are no proper timestamps in the file, only frame
numbers. And Matroska is insane and stores the made-up DTS instead
of a proper PTS. This will be handled properly later.
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Demuxer might access stream even when closing. For now, this is not
a real problem (because it didn't actually happen), but it's cleaner.
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This is not needed anymore, because we decided that the PAR of the
decoded video matters, and not the PAR of the filtered video that
arrives at the VO.
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This was way too misleading. osd.c merely calls the subtitle renderers,
instead of actually dealing with subtitles.
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In theory, we can't really do this, because we don't know when a spdif
frame ends. Spdif transports compressed audio through audio setups that
were originally designed for PCM only (which includes the audio filter
chain, the AO API, most audio output APIs, etc.), and to reach this
goal, spdif pretends to be PCM. Compressed data frames are padded with
zeros, until a certain data rate is reached, which corresponds to a
pseudo-PCM format with 2 bytes per sample and 2 channels at 48000 Hz.
Of course an actual spdif frame is significantly larger than a frame
of the PCM format it pretends to be, so cutting audio data on frame
boundaries (as according to the pseudo-PCM format) merely yields an
incomplete and broken frame, not audio that plays for the desired
duration.
However, sending an incomplete frame might still be much better than the
current behavior, which simply ignores --end/--length (but still lets
the video end at the exact end time).
Should this result in trouble with real spdif receivers, this commit
probably has to be reverted.
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If the --fps option was given (MPOpts->force_fps), the demuxer FPS value
was overwritten with the forced value. This was fine, since the demuxer
value wasn't needed anymore. But with the recent changes not to write to
the demuxer stream headers, we don't want to do this anymore. So
maintain the (forced/updated) FPS value in dec_video->fps.
The removed code in loadfile.c is probably redundant, and an artifact
from past refactorings.
Note that sub.c will now always use the demuxer FPS value, instead of
the user override value. I think this is fine, because it used the
demuxer's video size values too. (And it's rare that these values are
used at all.)
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The only reason why these structs were dynamically allocated was to
avoid recursive includes in stheader.h, which is (or was) a very central
file included by almost all other files. (If a struct is referenced via
a pointer type only, it can be forward referenced, and the definition of
the struct is not needed.) Now that they're out of stheader.h, this
difference doesn't matter anymore, and the code can be simplified.
Also sneak in some sanity checks.
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It's redundant.
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This used to be needed to access the generic stream header from the
specific headers, which in turn was needed because the decoders had
access only to the specific headers. This is not the case anymore, so
this can finally be removed again.
Also move the "format" field from the specific headers to sh_stream.
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Use sh_stream over sh_sub. Use dec_sub (and mpctx->d_sub) instead of the
stream header. This aligns the subtitle code with the recent audio and
video refactoring.
sh_sub still has the decoder context, though. This is because we want to
avoid reinit when switching segments with ordered chapters. (Reinit is
fast, except for creating the ASS_Renderer, which in turn triggers
fontconfig.) Not sure how much this matters, though, because the initial
segment switch will lazily initialize the decoder anyway.
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This is similar to the sh_audio commit.
This is mostly cosmetic in nature, except that it also adds automatical
freeing of the decoder driver's state struct (which was in
sh_video->context, now in dec_video->priv).
Also remove all the stheader.h fields that are not needed anymore.
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Commit 0cb9227a added this to the option list help output, but it looks
strange with sub-options.
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This means most code accessing this struct must now include hwdec.h
instead of dec_video.h. I just put it into dec_video.h at first because
I thought a separate file would be a waste, but it's more proper to do
it this way, as there are too many files which include dec_video.h only
to get the mp_hwdec_info definition.
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This was printed before something was decoded, and thus was not really
correct. Also, this code is hilariously broken:
/* Assume FOURCC if all bytes >= 0x20 (' ') */
if (sh_audio->format >= 0x20202020)
mp_msg(MSGT_IDENTIFY, MSGL_INFO,
"ID_AUDIO_FORMAT=%.4s\n", (char *)&sh_audio->format);
Time to kill it.
This information can be accessed through properties instead.
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sh_audio is supposed to contain file headers, not whatever was decoded.
Fix this, and write the decoded format to separate fields in the decoder
context, the dec_audio.decoded field. (Note that this field is really
only needed to communicate the audio format from decoder driver to the
generic code, so no other code accesses it.)
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Move all state that basically changes during decoding or is needed in
order to manage decoding itself into a new struct (dec_audio).
sh_audio (defined in stheader.h) is supposed to be the audio stream
header. This should reflect the file headers for the stream. Putting the
decoder context there is strange design, to say the least.
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This was forgotten when the parser for mplayer2 EDL files was removed.
Change the header of the mpv EDL format to include a '#', so a naive
parser could skip the header as comment. (Maybe this is questionable;
on the other hand, if it can be simpler, why not.)
Also, strip the header in demux_edl.c before passing on the data, so the
header check doesn't need to be duplicated in tl_mpv_edl.c.
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It was a bit too complicated and inconvenient, and I doubt anyone
actively used it. The mpv EDL format should cover all use cases.
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Questionable change from user perspective, but internally needed to
implement the next commit. Also useful for testing timeline stuff.
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Edit Decision Lists (EDL) allow combining parts from multiple source
files into one virtual file. MPlayer had an EDL format (which sucked),
which mplayer2 tried to improve with its own format (which sucked). As
logic demands, mpv introduces its very own format (which sucks).
The new format should actually be much simpler and easier to use, and
its implementation is simpler and smaller too.
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The intention of the existing code was trying to match demuxer-reported
stream IDs, instead of using possibly arbitrary ordering of the frontend
track list. But EDL files can consist of quite different files, for
which trying to match the stream IDs doesn't always make sense.
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This didn't have any consequences, other than suddenly reinitializing
video when it works again (such as with EDL timeline mixing video and
audio-only files).
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This could lead to (barely) audible artifacts with --af=scaletempo and
modified playback speed.
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The AF control commands used an elaborate and unnecessary organization
for the command constants. Get rid of all that and convert the
definitions to a simple enum. Also remove the control commands that
were not really needed, because they were not used outside of the
filters that implemented them.
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Some decoders used to read packets and decode data when calling
resync_audio_stream(). This required a special case in mp_seek() for
audio. (A comment mentions liba52, which is long gone; but until
recently ad_mpg123.c actually exposed this behavior.)
No decoder does this anymore, and resync_audio_stream() works similar
as resync_video_stream(). Remove the special case.
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