If no packets are queued, the readahead time is obviously 0.
If the end time is smaller than the start time, the problem is probably
that audio and video start at slightly different times - report 0 in
this case too.
Do this because seeing "???" as readahead time is a bit annoying.
Instead, use the native-endian alias, and switch the wayland format
depending on the target platform's endian.
This drops support for swapped-endian formats, but I think that is ok.
Not only are the affected formats rather ancient and backwards, but
using swapped formats probably does not make any sense for performance
either.
Untested.
These formats are still supported; you just can't reference them via a
defined constants directly. They are now handled via the generic
passthrough.
(If you want to use such a format, you either have to add the entry
back, or use AV_PIX_FMT_* directly.)
This is a rather radical change: instead of maintaining a whitelist of
FFmpeg formats we support, we automatically support all formats.
In general, a format which doesn't have an explicit IMGFMT_* name will
be converted to a known format through libswscale, or will be handled
by code which can treat pixel formats in a generic way using the pixel
format description, like vo_opengl.
AV_PIX_FMT_UYYVYY411 is a special-case. It's packed YUV with chroma
subsampling by 4 in both directions. Its component order is documented
as "Cb Y0 Y1 Cr Y2 Y3", meaning there's one UV sample for 4 Y samples.
This means each pixel uses 1.5 bytes (4 pixels have 1 UV sample, so
4 bytes + 2 bytes). FFmpeg can actually handle this format with its
generic mechanism in an extremely awkward way, but it doesn't work for
us. Blacklist it, and hope no similar formats will be added in the
future.
Currently, the AV_PIX_FMT_*s allowed are limited to a numeric value of
500. More is not allowed, and there are some fixed size arrays that need
to contain any possible format (look for IMGFMT_END dependencies).
We could have this simpler by replacing IMGFMT_* with AV_PIX_FMT_*
through the whole codebase. But for now, this is better, because we
can compensate for formats missing in Libav or older FFmpeg versions,
like AV_PIX_FMT_RGB0 and others.
FFmpeg has only a AV_PIX_FMT_FLAG_BE flag, not a LE one, which causes
problems for us: we want to have the LE flag too, so code can actually
detect whether a format is non-native endian. Basically, we want to
reconstruct the LE/BE suffix all AV_PIX_FMT_*s have.
Doing this is hard due to the (messed up) way AVPixFmtDescriptor works.
The worst is AV_PIX_FMT_RGB444: this group of formats describe an
endian-independent access (since no component actually spans 2 bytes,
you only need byte accesses with a fixed offset), so we have to go
through some pain.
This influences the demuxer readahead display. If a stream has reached
EOF, we want to ignore it for the purpose of this calculation.
Note that if a stream contains no packets, it still should cause the
value 0s to be displayed (unless it's EOF), because that's just the
actual situation.
This was relying on the fact that timestamps will always be numerically
larger than MP_NOPTS_VALUE, but the trick didn't actually work for
MP_PTS_MIN. Be a bit more sincere, and don't rely on this anymore. This
fixes the comparison, and avoids the readahead amount displaying as
"???" in some situations (since one of the values was NOPTS).
In this case, we didn't find any new packets for this stream, even
though we've read ahead as much as possible. (If reading ahead in this
case, the "Too many packets in the demuxer packet queues" error is
normally printed.)
If we do consider this an underrun, handle_pause_on_low_cache() will
pause and show the "buffering" state, which is not useful.
Could also happen on very bad interleaving.
This mechanism was introduced for Opus, and allows correct skipping of
"preroll" data, as well as discarding trailing audio if the file's
length isn't a multiple of the audio frame size.
Not sure how to handle seeking. I don't understand the purpose of the
SeekPreRoll element.
This was tested with correctness_trimming_nobeeps.opus, remuxed to mka
with mkvmerge v7.2.0. It seems to be correct, although the reported file
duration is incorrect (maybe a mkvmerge issue).
Add a generic mechanism to the VO to relay "extra" events from VO to
player. Use it to notify the core of window resizes, which in turn will
be used to mark all affected properties ("window-scale" in this case) as
changed.
(I refrained from hacking this as internal command into input_ctx, or to
poll the state change, etc. - but in the end, maybe it would be best to
actually pass the client API context directly to the places where events
can happen.)
Add a comment about this to avoid confusing users of this function. The
parameter is essentially unused, but exists so that we don't need to
add extra APIs if the need for it arises.
Instead of defining a separate data structure in the core.
For some odd reason, demux_chapter exported the chapter time in
nano-seconds. Change that to the usual timestamps (rename the field
to make any code relying on this to fail compilation), and also remove
the unused chapter end time.
Note that you can't pass .cue or .edl files to it, at least not yet.
Requested in context of allowing to specify custom chapters. For that
to work well, we probably need to add some sort of chapter metadata
pseudo-demuxer.
Using the --playlist option is no longer recommended.
A while ago, mpv rewrote all playlist parsers and added some minimal
security mechanisms (like not allowing local file access or unsafe
protocols in remote playlists). Further, mpv can load playlists by
passing them as normal file arguments, without the option.
Now, --playlist is needed only in these situations:
1) loading plaintext files
2) disabling additional security mechanisms
(e.g. using a remote playlist to play local files)
NSDisableScreenUpdates came to hunt me in the end and when mpv was paused, it
did wait for a frame that never came (because of interaction with the live
resizing code)!
This was shown only if decoder-framedropping was enabled, and only if at
least 50 frames were dropped by it. Since drop_frame_cnt used to mean
"number of late frames", this code made sense, but this is not the case
anymore: drop_frame_cnt can be even 0, all while video gets hopelessly
behind audio.
One problem with this is that short desync spikes (which usually can
probably dealt with) will also cause this message to be shown. If it
gets triggered too often, the code will need to be adjusted.
This shouldn't use the host's 'ar' when building static libs. It only
worked until now because Linux 'ar' is usually built with PE support.
Couldn't confirm whether it works, because this dumb crap is just
broken when cross-compiling to mingw.
Thanks to STREAM_CTRL_HAS_AVSEEK, we actually know whether CTRL_AVSEEK
is implemented at all, and we can avoid a blocking wait on the cache if
demux_lavf sends CTRL_AVSEEK even if it won't wait. I'm hoping this
can't currently happen, but why hope if we can explicitly prevent it.
It'll make us more robust against future changes in libavformat.
For example, if --force-window is used, and video is switched off during
playback, then you need to redecide the rendering method to get subs
displayed correctly.
Do this by moving the state setup code into a function, and call it on
every frame.
Apparently this can "sometimes" return an error. In my opinion, this
should never return an error: neither the semantics of the function,
nor the ALSA documentation or ALSA sample code seem to indicate that
a failure is to be expected. I'm not perfectly sure about this though
(I blame ALSA being a weird, big, underdocumented API).
Since it causes problems for some users, and since there is really no
reason why we should abort on such an error, turn it into a warning.
Fixes#1231.
If you played e.g. an audio-only file and something bad happened that
interrupted playback, the exit message could say "No files played".
This was awkward, so show a different message in this case.
Also overhaul how the exit status is reported in order to make this
easier. This includes things such as not reporting a playback error
when loading playlists (playlists contain no video or audio, which
was considered an error).
Not sure if I'm happy with this, but for now it seems like a slight
improvement.
Basically, this will mark the demuxer as seekable with rtmp* and mmsh
protocols. These protocols have network-level time seeking, and whether
you can seek on the byte level does not matter.
Until now, seeking was typically only enabled because of the cache, and
a (nonsensical) warning was shown accordingly.
It still could happen that the server doesn't actually support thse
requests (or simply rejects them), so this is somewhat imperfect.
