In all_formats mode, we've ignored what --ytdl-format did so far, since
we've converted the full format list, instead of just the formats
selected by youtube-dl.
But we can easily restore --ytdl-format behavior: just mark the selected
tracks as default tracks.
A parameter that is actually used is removed from the param_names[]
array, so we can report unused parameters. This also happened on
duplicate parameters, so adjust the warning to make it less confusing.
(In any case, you're not supposed to provide duplicate parameters.)
vbr and abr are the video and audio bitrates. Sometimes there is a weird
mix of any of them available, but in these cases, it's not good to fall
back to tbr if a specific track has no vbr/abr.
For example, the alphabetic site provides tbr only for the muxed
fallback stream, but using tbr would make the primitive mpv hls_bitrate
selection pick the compatibility stream for audio, because it appears to
have a higher bitrate than the other audio-only streams (because the
bitrate includes video). So we must not use tbr in this case.
On the other hand, formats coming from youtube-dl HLS master playlist
use will only have tbr set.
So as a heuristic, use the tbr only if it's the only bitrate available
in any track entry.
I don't think the skip_muxed option was overlay useful. While it was
nice to filter out the low quality muxed versions (as it happens on the
alphabetic site, I suspect it's compatibility stuff), it's not really
necessary, and just makes for another tricky and rarely used
configuration option. (This was different before muxed tracks were also
delay-loaded, and including the muxed versions slowed down loading.)
Add the force_all_formats option instead, which handles the HLS case.
Set it to true because they are also delay-loaded now, and don't slow
down startup as much.
Until now, delay-loading was for files with single tracks only
(basically what DASH and HLS like to expose, so adaptive streaming and
codec selection becomes easier - for sites, not for us). But they also
provide some interleaved versions, probably for compatibility. Until
now, we were forced to eagerly load it (making startup slightly slower).
But there is not much missing. We just need a way to provide multiple
metadata entries, and use them to represent each track.
A side effect is now that the "track_meta" header can be used for normal
EDL files too.
RTSP supports seeking, but at least the libavformat implementation makes
this dependent on runtime behavior. So you have to perform a seek, and
check if it fails. But even if you do this, the stream is interrupted
and restarted, and there seem to be other issues.
Assume that RTSP with unknown duration means it's a live stream, and
disable seeking in this case, as suggested by the issue reporter.
Fixes: #7472
Now this was stupid. To seek a source, it obviously has to be opened...
so just don't try to seek any unused source. If the track is actually
selected during playback, a seek to the correct position is performed
anyway.
These have ->segmented set (so the codec can be initialized properly),
but have no segment start or end times. This code was (probably) the
only thing which didn't handle this case.
If all streams were delay loaded, there was actually no duration present
at all in the EDL metadata. So the length was considered unknown by the
player frontend.
See manpage additions. We would have to extend delay_open to support
multiple sub-tracks (for audio and video), and we'd still don't know (?)
whether it might contain more than one stream each (thinking of HLS
master streams). And if it's a true interleaved file (such as a "normal"
mp4 file provided as fallback for more primitive players), we'd either
have to signal such "bundled" tracks, or waste bandwidth.
This restructures a lot. The if/else tree in add_single_video for format
selection was a bit annoying, so it's split into separate if blocks,
where it checks each time whether a URL was determined yet.
If a "format" has both audio and video codec set, it might contain both
audio and video. all_format assumes that each format is just a quality
variant containing a single track.
This seems to happen with sites that provide a HLS master URL.
youtube-dl tends to "butcher" it, and the result isn't very ideal. I
guess HLS "renditions" simply don't map well to youtube-dl's output
format and what mpv expects. Playing master HLS directly is also less
than ideal, because of libavformat's stupid probing.
Fix this by not using the delay-opening mechanism if it appears like we
detected such a case. Add a metadata override to set the track titles to
"muxed-N", to indicate that they form a single unit. (Mostly helpful for
testing.)
ytdl_hook.lua can do this with all_formats and when delay_open is used,
and if the source stream actually contains both audio and video. In this
case, it might accidentally hide a media type completely, or waste
bandwidth (if the stream has true interleaved audio/video). So it's
important to warn.
This is just a more convenient way to start IPC client scripts per mpv
instance.
Does not work on Windows, although it could if the subprocess and IPC
parts are implemented (and I guess .exe/.bat suffixes are required).
Also untested whether it builds on Windows. A lot of other things are
untested too, so don't complain.
Pretty worthless I guess. I only tested one site (and 2 videos), it's
somewhat likely that it will break with other sites. Even if you leave
the option disabled (the default).
Slightly related to #3548. This will allows you to use the bitrate
stream selection mechanism, that was added for HLS, with normal videos.
While paused, the decoders typically stop reading data from the demuxer.
But for some reason, the file size is returned as a public field in
struct demuxer (wat...), and updated only when the packet reading
function is called. This caused the file size property to always return
the same value when paused, even though the demuxer thread was reading
new data, and the internal file size was updated.
Fix with a simple hack.
Instead of every packet. Doing it every packet led to the performance
regression mentioned in the fstat() commit. This should now be over, but
out of being careful, don't query the file size that often. This is only
used for user interface things, so this should not cause any problems.
For the sake of leaving the code compact, abuse another thing that is
updated only every second (speed statistics).
It appears using lseek() to seek to the end and back to determine file
size is inefficient in some cases.
With CIFS, this restores the performance regression that happened when
the stream cache was removed (which called read() from a thread). This
is probably faster than the old code too, because it's the seeking that
was slowing down CIFS.
According to the user who tested this, the size caching does not help
with fstat() (although it did with the old method).
Fixes: #7408, #7152
The previous method for this sucked: for every launched detached
process, it started a thread, which then would leak if the launched
process didn't end before the player uninitialized. This was very racy
(although I bet the race condition wouldn't trigger in a 100 years), and
wasteful (threads aren't a cheap resource).
Implement it for POSIX directly. posix_spawn() has no direct support for
this, so we need to do it ourselves with fork(). We could probably do it
without fork(), and attempt to collect the PID in another thread. But
then we'd either have a waiting thread again, or we'd need to do an
unsafe waitpid(-1, ...) call. (POSIX process management sucks so badly,
how did they even manage this. Hopefully I'm just missing something, but
I'm not.) So now we depend on both posix_spawn() _and_ fork(), isn't it
fun?
Also call setsid(), to essentially detach the child process from the
terminal. (Otherwise it can receive various signals from the terminal,
which is probably not what you want.) posix_spawn() adds
POSIX_SPAWN_SETSID in newer POSIX releases, but we don't want to rely on
this yet.
The posix_spawnp() call is duplicated, but this is better than somehow
trying to unify the code paths.
Only somewhat tested, so enjoy the bugs.
Introduce mp_subprocess() and related definitions. This is a bit more
flexible than the old stuff. This may or may not be used for a more
complicated feature that involves starting processes, and which would
require more control.
Only port subprocess-posix.c to this API. The player still uses the
"old" API, so for win32 and dummy implementations, the new API is simply
not available, while for POSIX, the old APIs are emulated on top of the
new one. I'm hoping the win32 code can be ported as well, so the ifdefs
in subprocess.c can be dropped, and the player can (if convenient or
needed) use the new API.
Filters that fail to create are not supposed to do this. Generally it
should happen in process() only.
This fixes the previous commit. If a filter could not be created, it
"trashed" the wrapper filter with the failure. (Even if the wrapper were
to handle were to handle failures of sub-filter, it couldn't handle init
failure because it cannot call mp_filter_set_error_handler() before the
newly created filter is actually returned.)
Fixes: #7465 (attempt 2)
If hw decoding is used, but no hw deinterlacer is available, even though
we expect that it is present, fall back to using hw-download and yadif
anyway. Until now, it was over if the hw filter was somehow missing; for
example, yadif_cuda apparently requires a full Cuda SDK, so it can be
missing, even if nvdec is available. (Whether this particular case works
was not tested with this commit.)
Fixes: #7465
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.
Works as ad-filter. I had some more plans, for example replacing
matching text with different text, but for now it's dropping matches
only. There's a big warning in the manpage that I might change
semantics. For example, I might turn it into a primitive sed.
In a sane world, you'd probably write a simple script that processes
downloaded subtitles before giving them to mpv, and avoid all this
complexity. But we don't live in a sane world, and the sooner you learn
this, the happier you will be. (But I also want to run this on muxed
subtitles.)
This is pretty straightforward. We use POSIX regexes, which are readily
available without additional pain or dependencies. This also means it's
(apparently) not available on win32 (MinGW). The regex list is because I
hate big monolithic regexes, and this makes it slightly better.
Very superficially tested.
Until now, filter_sdh was simply a function that was called by sd_ass
directly (if enabled).
I want to add another filter, so it's time to turn this into a somewhat
more general subtitle filtering infrastructure.
I pondered whether to reuse the audio/video filtering stuff - but better
not. Also, since subtitles are horrible and tend to refuse proper
abstraction, it's still messed into sd_ass, instead of working on the
dec_sub.c level. Actually mpv used to have subtitle "filters" and even
made subtitle converters part of it, but it was fairly horrible, so
don't do that again.
In addition, make runtime changes possible. Since this was supposed to
be a quick hack, I just decided to put all subtitle filter options into
a separate option group (=> simpler change notification), to manually
push the change through the playloop (like it was sort of before for OSD
options), and to recreate the sub filter chain completely in every
change. Should be good enough.
One strangeness is that due to prefetching and such, most subtitle
packets (or those some time ahead) are actually done filtering when we
change, so the user still needs to manually seek to actually refresh
everything. And since subtitle data is usually cached in ASS_Track (for
other terrible but user-friendly reasons), we also must clear the
subtitle data, but of course only on seek, since otherwise all subtitles
would just disappear. What a fucking mess, but such is life. We could
trigger a "refresh seek" to make this more automatic, but I don't feel
like it currently.
This is slightly inefficient (lots of allocations and copying), but I
decided that it doesn't matter. Could matter slightly for crazy ASS
subtitles that render with thousands of events.
Not very well tested. Still seems to work, but I didn't have many test
cases.
