This should actually be rather safe - we already check whether the
estimated value jitters less than the (possibly untrustworthy) nominal
one. Remove a "safety" check that disabled this code for small
deviations, and make it trigger sooner into playback. Also lower the log
level of messages about using the estimated display FPS down to verbose.
Normally there's another mechanism for smoothing out minor estimation
differences, but that is not good enough here.
This possibly improves behavior as reported in #3433, which can be
reproduced with --vo=null:fps=48.426 --display-fps=48 (though it doesn't
consider the jitter introduced by a real VO).
Doing this required synchronizing with the VO thread, which could lead
to audio dropouts if the VO was frozen (which can happen in practice if
e.g. an opengl_cb user is not doing what the API demands).
Add a way to send asynchronous VOCTRLs, and use that for the playback
state. In theory, it would be better to make this status update a
several function and to "merge" several queued update, but that would be
slightly more effort/code, and the update is so infrequent that the
merging would never happen anyway.
The change to vo_destroy() is to make sure all queued asynchronous
reuqests are finished before making the vo_thread exit.
Even though it's only used on MS Windows, it's run on any platform with
any VO, which makes this worse.
run_control() dereferences an uint32_t as int. Whether this is allowed
depends on what uint32_t is typedefed to (dereferencing an unsigned int
as int should be fine). Fix it by always using int. The uint32_t type
never really made sense.
There are situations where the resampler is destroyed and recreated
during playback. If recreating the resampler unexpectedly fails, the
filter function is supposed to return an error. This wasn't done
correctly, because get_out_samples() accessed the resampler before the
check. Move the check up to fix this.
When fetching the playlist property, playlist_entry_from_index would be
called for each playlist entry, which traversed a linked list to get the
entry corresponding to the specified index. This was very slow for large
playlists. Since get_playlist_entry is called for each index in order,
it can avoid a full traversal of the linked list by using the next
pointer on the previously requested entry.
Instead of passing through double float timestamps opaquely, pass real
timestamps. Do so by always setting a valid timebase on the
AVCodecContext for audio and video decoding.
Specifically try not to round timestamps to a too coarse timebase, which
could round off small adjustments to timestamps (such as for start time
rebasing or demux_timeline). If the timebase is considered too coarse,
make it finer.
This gets rid of the need to do this specifically for some hardware
decoding wrapper. The old method of passing through double timestamps
was also a bit questionable. While libavcodec is not supposed to
interpret timestamps at all if no timebase is provided, it was
needlessly tricky. Also, it actually does compare them with
AV_NOPTS_VALUE. This change will probably also reduce confusion in the
future.
This affects A-B loops and --loop-file, and audio. Instead of dropping
audio by resetting the AO, try to make it seamless by not sending data
after the loop point, and after the seek send new data without a reset.
The code actually kept going out of EOF mode into resync mode back into
EOF mode when the playloop had to wait after an audio EOF caused by the
endpts. This would break seamless looping (as added by the next commit).
Apply endpts earlier, to ensure the filter_audio() function always
returns AD_EOF in this case.
The idiotic ao_buffer makes this an amazing pain in the ass.
Instead of letting it keep decoding by trying to find a new frame,
"plug" the frame queue by not removing it. (Or actually, by putting
it back instead of discarding it.)
Matters for seamless looping (following commits), and possibly some
other corner cases.
The added function vf_unread_output_frame() is a bit of a sin, but still
reasonable, since its implementation is trivial.
The --image-display-duration option controls how long an image is
displayed. It's also possible to display the image forever (until manual
user interaction stops playback).
With this, the core drops the old method to "drain" video (i.e. waiting
for the last frame duration on end of playback). Instead, we reuse
MPContext.time_frame. The old mechanism was disabled for non-images
anyway.
Fixes#3425.
When an ogg track upodates metadata, we have to perform a complicated
runtime update due to the demux.c architecture. A detail was broken and
an array was allocated with the previous number of streams, which
usually led to invalid memory write accesses at least on the first
update.
See github commit comment on commit b9ba9a89.
The touched code is for seek resets and such - we simply want to reset
the entire resample state. But I noticed after a seek a tiny bit of
audio is missing (mpv's audio sync code inserted silence to compensate).
It turns out swr_drop_output() either does not reset some internal state
as we expect, or it's designed to drop not only buffered samples, but
also future samples.
On the other hand, libavresample's avresample_read(), does not have this
problem. (It is also pretty explicit in what it does - return/skip
buffered data, nothing else.)
Is the libswresample behavior a bug? Or a feature? Does nobody even
know? Who cares - use the hammer to unfuck the situation. Destroy and
deallocate the libswresample context and recreate it. On every seek.
Change the last parameter from a bool to an int, which is supposed to
take bit-flags. The at this point only flag is MPSEEK_FLAG_DELAY, which
replaces the previous bool parameter. The old false parameter becomes 0,
the old true parameter becomes MPSEEK_FLAG_DELAY.
Since the old "immediate" parameter is now essentially inverted, two
coalesced immediate and delayed seeks end up as delayed instead of
immediate. This change doesn't matter, since there are no relative
immediate seeks anyway.
The accepts_packet packet callback is supposed to deal with subtitle
decoders which have only a small queue of current subtitle events (i.e.
sd_lavc.c), in case feeding it too many packets would discard events
that are still needed.
Normally, the number of subtitles that need to be preserved is estimated
by the rendering pts (get_bitmaps() argument). Rendering lags behind
decoding, so normally the rendering pts is smaller than the next video
frame pts, and we simply discard all subtitle events until the rendering
pts.
This breaks down in some annoying corner cases. One of them is seeking
backwards: the VO will still try to render the old PTS during seeks,
which passes a high PTS to the subtitle renderer, which in turn would
discard more subtitles than it should. There is a similar issue with
forward seeks. Add hacks to deal with those issues.
There should be a better way to deal with the essentially unknown
"rendering position", which is made worse by screenshots or rendering
with vf_sub. At the very least, we could handle seeks better, and e.g.
either force the VO not to re-render subs after seeks (ugly), or
introduce seek sequence numbers to distinguish attempts to render
earlier subtitles when a seek is done.
If the PEAK tag is invalid, return an error.
Make the error signalling conventions more uniform by strictly returning
a negative value on error, and treating >=0 as success.
The demuxer layer usually doesn't log per-stream information, and even
the replaygain information was logged only if it came from tags.
So log it in af_volume instead.
...and ignore it. The main purpose is for retrieving per-track
replaygain tags. Other than that per-track tags are not used or accessed
by the playback core yet.
The demuxer infrastructure is still not really good with that whole
synchronization thing (at least in part due to being inherited from
mplayer's single-threaded architecture). A convoluted mechanism is
needed to transport the tags from demuxer thread to user thread. Two
factors contribute to the complexity: tags can change during playback,
and tracks (i.e. struct sh_stream) are not duplicated per thread.
In particular, we update the way replaygain tags are retrieved. We first
try to use per-track tags (common in Matroska) and global tags
(effectively formats like mp3). This part fixes#3405.
With the previous commit, ao_alsa.c now has 3 possible ways to pause
playback. Actually all 3 of them need get_delay() to fake its return
value, so don't duplicate that code.
Also much of the code looks a bit questionable when considering
inconsistent pause/resume calls from outside, so ignore redundant calls.
push.c does not handle this automatically, and AOs using push.c have to
handle it themselves. Also, ALSA is low-level enough that it needs
explicit support in user code. At least I haven't found any option that
does this.
We still can get away relatively cheaply by abusing underflow-handling
for this. ao_alsa.c already configures ALSA to handle underflows by
playing silence. So we purposely induce an underflow when opening the
device, as well as when pausing or resetting the device.
This introduces minor misbehavior: it doesn't account for the additional
delay the initial silence adds, unless the device has fully played the
fragment of silence when the player starts sending data to it. But
nobody cares.
