| Commit message (Collapse) | Author | Age | Files | Lines |
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Instead of having more global state just do it on-the-fly.
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This adds cache as a possible path for mpv to internally pick
(~/.cache/mpv for non-darwin unix-like systems, the usual config
directory for everyone else). For gpu shader cache and icc cache,
controlling whether or not to write such files is done with the new
--gpu-shader-cache and --icc-cache options respectively. Additionally,
--cache-on-disk no longer requires explicitly setting the --cache-dir
option. The old options, --cache-dir, --gpu-shader-cache-dir, and
--icc-cache-dir simply set an override for the directory to save cache
files. If unset, then the cache is saved in XDG_CACHE_HOME.
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A pain point for some users is the fact that watch_later is stored in
the ~/.config directory when it's really not configuration data. Roughly
2 years ago, XDG_STATE_DIR was added to the XDG Base Directory
Specification[0] and its description, user-specific state data, actually
perfectly matches what watch_later data is for. Let's go ahead and use
this directory as the default for watch_later. This change only affects
non-darwin unix-like systems (i.e. Linux, BSDs, etc.). The directory
doesn't move for anyone else.
Internally, quite a few things change with regards to the path
selection. If the platform in question does not have a statedir concept,
then the path selection will simply return "home" instead (old
behavior). Fixes #9147.
[0]: https://gitlab.freedesktop.org/xdg/xdg-specs/-/commit/4f2884e16db35f2962d9b64312917c81be5cb54b
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It no longer has any users, as the last ones (subtitle decoders)
were switched to mp_set_avctx_codec_headers.
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They should not be modifying the argument, so clearly marking it
as const makes sure we don't do it in the future as well as allows
for read-only optimizations.
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c78482045444c488bb7948305d583a55d17cd236 introduced a bool option type
as a replacement for the flag type, but didn't actually transition and
remove the flag type because it would have been too much mundane work.
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Most sources don't need config.h.
The inclusion only leads to lots of unneeded recompilation if the
configuration is changed.
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Previously, if log-file was set not via a CLI option (e.g. set via
mpv.conf or other config file, or set from a script init phase),
then meaningful early log messages were thrown away because the log
file name was unknown initially.
Such early log messages include the command line arguments, any
options set from mpv.conf, and possibly more.
Now we store up to 5000 early messages before the log file name is
known, and flush them once/if it becomes known, or destroy this
buffer once mpv init is complete.
The implementation is similar and adjacent, but not identical, to an
existing early log system for mpv clients which request a log buffer.
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In debug mode the macro causes an assertion failure.
In release mode it works differently and tells the compiler that it can
assume the codepath will never execute. For this reason I was conversative
in replacing it, e.g. in mpv-internal code that exhausts all valid values
of an enum or when a condition is clear from directly preceding code.
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So far there was no way to sync video to display and have audio sync to
video without changes in pitch.
With this option the audio does not get resampled (pitch change) and
instead the corrected audio speed is applied to audio filters.
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This has been a long standing annoyance - ffmpeg is removing
sizeof(AVPacket) from the API which means you cannot stack-allocate
AVPacket anymore. However, that is something we take advantage of
because we use short-lived AVPackets to bridge from native mpv packets
in our main decoding paths.
We don't think that switching these to `av_packet_alloc` is desirable,
given the cost of heap allocation, so this change takes a different
approach - allocating a single packet in the relevant context and
reusing it over and over.
That's fairly straight-forward, with the main caveat being that
re-initialising the packet is unintuitive. There is no function that
does exactly what we need (what `av_init_packet` did). The closest is
`av_packet_unref`, which additionally frees buffers and side-data.
However, we don't copy those things - we just assign them in from our
own packet, so we have to explicitly clear the pointers before calling
`av_packet_unref`. But at least we can make a wrapper function for
that.
The weirdest part of the change is the handling of the vtt subtitle
conversion. This requires two packets, so I had to pre-allocate two in
the context struct. That sounds excessive, but if allocating the
primary packet is too expensive, then allocating the secondary one for
vtt subtitles must also be too expensive.
This change is not conditional as heap allocated AVPackets were
available for years and years before the deprecation.
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This way another library can be logged before FFmpeg without it
looking weird.
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Add xoshiro as a PRNG implementation instead of relying
on srand() and rand() from the C standard library. This,
in particular, lets us avoid platform-defined behavior with
respect to threading.
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The video sync logic for mpv lies completely within its core at
essentially the highest layer of abstraction. The problem with this is
that it is impossible for VOs to know what video sync mode mpv is
currently using since it has no access to the opts. Because different
video sync modes completely changes how mpv's render loop operates, it's
reasonable that a VO may want to change how it renders based on the
current mode (see the next commit for an example).
Let's just move the video sync option to mp_vo_opts. MPContext, of
course, can still access the value of the option so it only requires
minor changes in player/video.c. Additionally, move the VS_IS_DISP
define from to player/core.h to common/common.h. All VOs already have
access to common/common.h, and there's no need for them to gain access
to everything that's in player/core.h.
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FFmpeg recently split version.h into version.h and
version_major.h, and no longer automatically includes
version.h in avcodec.h (and the other libraries). This
should allow mpv to build against ffmpeg git master. See
FFmpeg/ffmpeg@f2da2e1458b76a1d6c068673430b46cf2850bc51
These headers are not new, so their inclusion should not affect
backwards compatibility.
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Adds support for the meson build system as well as a bit of
documentation. Compatibility with the existing waf build is
maintained.
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Though, only when the output format is matroska, to avoid muxing errors.
This is quite useful when the input has ASS subtitles, as they tend to
rely on embedded fonts.
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I've looked and studied the flow in the recorder, and to my
understanding, the packet queue is moot after the initial sync, maybe
even then - but that's beyond me right now.
With the previous choice to mux trailing packets whatever the case, this
doesn't result in any new ill effects (and some missing packets at the
end is no big deal).
Notably, since we don't have to hold onto the packets after we get
muxing, we'll never run into any issues with veeery long GOPs filling up
our queue (resulting in dropped packets and much user chagrin).
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For muxing, FFmpeg expects the FLAC extradata to be just the bare STREAMINFO,
and passing the full FLAC extradata (fLaC header and block size, with any
additional channel layout metadata) will result in malformed output, as
ffmpeg will simply prefix another fLaC header in front.
This can be considered to be a bug.
FFmpeg's own demuxers only store the STREAMINFO, hence the naivety, while
our common source of FLAC streams, the matroska demuxer, holds onto the
full extradata. It has been deemed preferable to adjust the extradata upon
muxing, instead of in the demuxer (ffmpeg's FLAC decoder knows to read
the full fLaC extradata).
This fixes muxing FLAC streams, meaning recorder.c or dump-cache.
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In commit f7678575a5d7f598abf267cb303e0a74db276f27, wm4 chooses to mux
all remaining packets when mp_recorder_mark_discontinuity() is called and
adds a call to mux_packets(). However, it is called only after flush_packets(),
which clears the packets before they can be muxed out. This has no ill
effects per se - recordings end on keyframes, as before - but judging from
his commit message, the intention explicitly was to output the inter
frames, since long GOPs can mean several seconds of missing content from
the output. So, clear the stream packet queues only after the final mux.
Also, flushing can mean both discarding and committing. What a country!
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Avoiding blindly copying the codec_tag between different formats allows
us to mux packets from, say, mpegts streams to matroska, making the
recorder (dump-cache) much more usable as unsupported codec_tags can
make the muxer reject the streams.
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FFmpeg recently changed these to be const on their side.
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if log-file and really-quiet options were used together it could lead to
a completely empty log-file. this is unexpected because we need the
log-file option to work in all cases and produces at least a log of
verbosity -v -v. this is a regression of commit
a600d152d21ef398eb72b008ee3fe266696eb638
move the really quiet check back up, so it's set before the evaluation
of the actual log level, where check for log file, terminal, etc take
place.
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More readable, similar to what --log-file will use (although the
terminal code shows microseconds and uses less left padding).
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If you encode to e.g. an audio-only format, then video is disabled
automatically. This also takes care of the very cryptic error message.
It says "[vo/lavc] codec for video not found". Sort of true, but
obscures the real problem if it's e.g. an audio-only format.
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I don't see the point of this. Not doing it may defer an error to later.
That's OK? For now, it seems better to reduce the encoding internal API.
If someone can demonstrate that this is needed, I might reimplement it
in a different way.
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I guess the audio timestamp corruption problem is probably solved now.
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This replaces the two buffers (ao_chain.ao_buffer in the core, and
buffer_state.buffers in the AO) with a single queue. Instead of having a
byte based buffer, the queue is simply a list of audio frames, as output
by the decoder. This should make dataflow simpler and reduce copying.
It also attempts to simplify fill_audio_out_buffers(), the function I
always hated most, because it's full of subtle and buggy logic.
Unfortunately, I got assaulted by corner cases, dumb features (attempt
at seamless looping, really?), and other crap, so it got pretty
complicated again. fill_audio_out_buffers() is still full of subtle and
buggy logic. Maybe it got worse. On the other hand, maybe there really
is some progress. Who knows.
Originally, the data flow parts was meant to be in f_output_chain, but
due to tricky interactions with the playloop code, it's now in the dummy
filter in audio.c.
At least this improves the way the audio PTS is passed to the encoder in
encoding mode. Now it attempts to pass frames directly, along with the
pts, which should minimize timestamp problems. But to be honest, encoder
mode is one big kludge that shouldn't exist in this way.
This commit should be considered pre-alpha code. There are lots of bugs
still hiding.
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Force them into a more consistent naming schema.
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This simply printf()s a concatenation of the provided string and the
relevant escape sequences. No idea what exactly defines this escape
sequence (is it just a xterm thing that is now supported relatively
widely?), and this simply uses information provided on the linked github
issue.
Not much of an advantage over --term-status-msg, though at least this
can have a lower update frequency. Also I may consider setting a default
value, and then it shouldn't conflict with the status message.
Fixes: #1725
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Not sure if generally useful; the following commit uses it.
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Sometimes it's helpful to override this for specific mp_log instances,
because in some specific circumstances you just want to suppress log
file noise you never want to see.
-1 is an allowed value (for suppressing MSGL_FATAL==0). It looks like
the libplacebo wrapper still does this wrong, so it will probably
trigger UB in some cases. I guess I don't care, though.
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Who write this dumb shit¹? It didn't handle 1<<31, and was unnecessarily
slow.
¹ it was me
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In addition to stats.c being gross, I don't think master branch code
should be littered with debug code. But it's a helpful abomination.
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Add an infrastructure for collecting performance-related data, use it in
some places. Add rendering of them to stats.lua.
There were two main goals: minimal impact on the normal code and normal
playback. So all these stats_* function calls either happen only during
initialization, or return immediately if no stats collection is going
on. That's why it does this lazily adding of stats entries etc. (a first
iteration made each stats entry an API thing, instead of just a single
stats_ctx, but I thought that was getting too intrusive in the "normal"
code, even if everything gets worse inside of stats.c).
You could get most of this information from various profilers (including
the extremely primitive --dump-stats thing in mpv), but this makes it
easier to see the most important information at once (at least in
theory), partially because we know best about the context of various
things.
Not very happy with this. It's all pretty primitive and dumb. At this
point I just wanted to get over with it, without necessarily having to
revisit it later, but with having my stupid statistics.
Somehow the code feels terrible. There are a lot of meh decisions in
there that could be better or worse (but mostly could be better), and it
just sucks but it's also trivial and uninteresting and does the job. I
guess I hate programming. It's so tedious and the result is always shit.
Anyway, enjoy.
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May or may not help when dealing with playlist loading in scripts. It's
supposed to help with the mean fact that loading a recursive playlist
will essentially edit the playlist behind the API user's back.
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Looks like the recent change to this actually made it crash whenever
audio happened to be initialized first, due to not setting the
mux_stream field before the on_ready callback. Mess a way around this.
Also remove a stray unused variable from ao_lavc.c.
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While I'd like to keep it, I'm apparently the maintainer now, and I have
no idea what the heck some of this code does, so it's deprecated.
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Seems to crash hard if an error happens somewhere at init. Who cares.
Part of #7524.
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This should make dealing with some async. things easier.
It's intentionally not a globally unique ID.
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Should give a good deal more explicit control and insight over the
player state.
Some feel a bit pointless, and/or expose internal weirdness. However,
it's not like the existing weirdness didn't exist before, or can be made
go away. (In part, the weirdness is because certain in-between states
are visible. Hiding them would make things simpler, but less flexible.)
Maybe this actually gives users a better idea how the API _should_ look
like, too.
On a side note, this tries to really guarantee that mpctx->playing is
set between playback start/end. For that, the loadfile.c changes assume
that mpctx->playing is set (guaranteed by code above the change), and
that playing->filename is set (probably could never be false; was broken
before and actually would have crashed if that could ever happen; in any
case, also add an assert to playlist.c for this).
playlist_entry_to_index() now tolerates playlist_entrys that are not
part of the playlist. This is also needed for mpctx->playing.
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Change all OPT_* macros such that they don't define the entire m_option
initializer, and instead expand only to a part of it, which sets certain
fields. This requires changing almost every option declaration, because
they all use these macros. A declaration now always starts with
{"name", ...
followed by designated initializers only (possibly wrapped in macros).
The OPT_* macros now initialize the .offset and .type fields only,
sometimes also .priv and others.
I think this change makes the option macros less tricky. The old code
had to stuff everything into macro arguments (and attempted to allow
setting arbitrary fields by letting the user pass designated
initializers in the vararg parts). Some of this was made messy due to
C99 and C11 not allowing 0-sized varargs with ',' removal. It's also
possible that this change is pointless, other than cosmetic preferences.
Not too happy about some things. For example, the OPT_CHOICE()
indentation I applied looks a bit ugly.
Much of this change was done with regex search&replace, but some places
required manual editing. In particular, code in "obscure" areas (which I
didn't include in compilation) might be broken now.
In wayland_common.c the author of some option declarations confused the
flags parameter with the default value (though the default value was
also properly set below). I fixed this with this change.
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No need to be nice. Also hopefully breaks idiotic distro patches.
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Libav seems rather dead: no release for 2 years, no new git commits in
master for almost a year (with one exception ~6 months ago). From what I
can tell, some developers resigned themselves to the horrifying idea to
post patches to ffmpeg-devel instead, while the rest of the developers
went on to greener pastures.
Libav was a better project than FFmpeg. Unfortunately, FFmpeg won,
because it managed to keep the name and website. Libav was pushed more
and more into obscurity: while there was initially a big push for Libav,
FFmpeg just remained "in place" and visible for most people. FFmpeg was
slowly draining all manpower and energy from Libav. A big part of this
was that FFmpeg stole code from Libav (regular merges of the entire
Libav git tree), making it some sort of Frankenstein mirror of Libav,
think decaying zombie with additional legs ("features") nailed to it.
"Stealing" surely is the wrong word; I'm just aping the language that
some of the FFmpeg members used to use. All that is in the past now, I'm
probably the only person left who is annoyed by this, and with this
commit I'm putting this decade long problem finally to an end. I just
thought I'd express my annoyance about this fucking shitshow one last
time.
The most intrusive change in this commit is the resample filter, which
originally used libavresample. Since the FFmpeg developer refused to
enable libavresample by default for drama reasons, and the API was
slightly different, so the filter used some big preprocessor mess to
make it compatible to libswresample. All that falls away now. The
simplification to the build system is also significant.
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Cut the arbitrary offset, and document what unit/timesource it uses.
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This is a central lock (that is to stay and has no reason to go away),
and it was simply made global. This reduces complexity when the original
MPlayer code was changed from single thread + global state to a context
handle.
Having the global lock was still a bit silly if there were multiple mpv
instances in the process, because it would make the instances wait for
each other for no reason. So move it to the per-instance context, which
is trivial enough.
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