| Commit message (Collapse) | Author | Age |
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In particular, don't allow to add any external subtitle tracks in idle
mode. This make no sense and would just lead to leaks or worse.
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Because that might be a bad idea.
Note that remote playlists still can use any protocol marked with
is_safe and is_network, because the case of http-hosted playlists
containing URLs using other streaming protocols is not unusual.
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The event was copied early, and wasn't released if it was rejected
instead of being added to the event queue. Fix by copying the event at a
point when it's certainly added to the event queue.
The dup_event_data() function is merely moved.
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Until now, you had to use --load-unsafe-playlists or --playlist to get
playlists loaded. Change this and always load playlists by default.
This still attempts to reject unsafe URLs. For example, trying to invoke
libavdevice pseudo-demuxer is explicitly prevented. Local paths and any
http links (and some more) are always allowed.
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Probably no observable effect, but it's more correct. Setting audio to
EOF could have bad effects otherwise (anywhere the player logic for
example decides whether EOF was reached, and such).
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This inserts an automatic conversion filter if a Matroska file is marked
as 3D (StereoMode element). The basic idea is similar to video rotation
and colorspace handling: the 3D mode is added as a property to the video
params. Depending on this property, a video filter can be inserted.
As of this commit, extending mp_image_params is actually completely
unnecessary - but the idea is that it will make it easier to integrate
with VOs supporting stereo 3D mogrification. Although vo_opengl does
support some stereo rendering, it didn't support the mode my sample file
used, so I'll leave that part for later.
Not that most mappings from Matroska mode to vf_stereo3d mode are
probably wrong, and some are missing.
Assuming that Matroska modes, and vf_stereo3d in modes, and out modes
are all the same might be an oversimplification - we'll see.
See issue #1045.
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The player didn't quit when seeking past EOF in audio-only mode while
paused. The only case when we don't want to quit is when the last video
frame is displayed while paused.
This logic was probably broken a while ago, but I'm not exactly sure.
CC: @mpv-player/stable
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bstr.c doesn't really deserve its own directory, and compat had just
a few files, most of which may as well be in osdep. There isn't really
any justification for these extra directories, so get rid of them.
The compat/libav.h was empty - just delete it. We changed our approach
to API compatibility, and will likely not need it anymore.
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Signed-off-by: wm4 <wm4@nowhere>
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Add a mechanism to the client API code, which allows the player core to
query whether a client API event is needed at all. Use it for the cache
update.
In this case, this is probably a pure microoptimization; but the
mechanism will be useful for other things too.
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Don't attempt to resync after speed changes. Note that most other cases
of audio reinit (like switching tracks etc.) still resync, but other
code paths take care of setting the audio_status accordingly.
This restores the old behavior of not trying to fix audio desync, which
was probably changed with commit 261506e3.
Note that the code as of now wasn't even entirely correct, since the A/V
sync values are slightly shifted. The dsync depends on the audio buffer
size, so a larger buffer size will show more extreme desync. Also see
mplayer2 commit 213a224e, which should fixed this - it was not merged
into mpv, because it disabled audio for too long, resulting in a worse
user experience. This is similar to the issue this commit attempts to
fix.
Fixes: #1042 (probably)
CC: @mpv-player-stable
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The display name is always recomputed, so we can always toss the input
name.
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This can occur if the directory does not have any files in it which
causes files to never be non-NULL for qsort.
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Remove the hardcoded wait time of 2 seconds. Instead, adjust the wait
time each time we unpause: if downloading the data took longer than its
estimated playback time, increase the amount of data we wait for. If
it's shorter, decrease it.
The +/- is supposed to avoid oscillating between two values if the
elapsed time and the wait time are similar. It's not sure if this
actually helps with anything, but it can't harm.
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Use the "native" underrun detection, instead of guessing by a low cache
duration. The new underrun detection (which was added with the original
commit) might have the problem that it's easy for the playloop to miss
the underrun event. The underrun is actually not stored as state, so if
the demuxer thread adds a new packet before the playloop happens to see
the state, it's as if it never happened. On the other hand, this means
that network was fast enough, so it should be just fine.
Also, should it happen that we don't know the cached range (the
ts_duration < 0 case), just wait until the demuxer goes idle (i.e.
read_packet() decides to stop). This pretty much should affect broken or
unusual files only, and there might be various things that could go
wrong. But it's more robust in the normal case: this situation also
happens when no packets have been read yet, and we don't want to
consider this as reason to resume playback.
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The cache percentage was useless. It showed how much of the total stream
cache was in use, but since the cache size is something huge and
unrelated to the bitrate or network speed, the information content of
the percentage was rather low.
Replace this with printing the duration of the demuxer-cached data, and
the size of the stream cache in KB.
I'm not completely sure about the formatting; suggestions are welcome.
Note that it's not easy to know how much playback time the stream cache
covers, so it's always in bytes.
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The "buffering" logic was active even if the stream cache was disabled.
This is contrary to what the manpage says. It also breaks playback
because of another bug: the demuxer cache is smaller than 2 seconds,
and thus the resume condition never becomes true.
Explicitly run this code only if the stream cache is enabled. Also, fix
the underlying problem of the breakage, and resume when the demuxer
thread stops reading in any case, not just on EOF.
Broken by previous commit. Unbreaks playback of local files.
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Add the --cache-secs option, which literally overrides the value of
--demuxer-readahead-secs if the stream cache is active. The default
value is very high (10 seconds), which means it can act as network
cache.
Remove the old behavior of trying to pause once the byte cache runs
low. Instead, do something similar wit the demuxer cache. The nice
thing is that we can guess how many seconds of video it has cached,
and we can make better decisions. But for now, apply a relatively
naive heuristic: if the cache is below 0.5 secs, pause, and wait
until at least 2 secs are available.
Note that due to timestamp reordering, the estimated cached duration
of video might be inaccurate, depending on the file format. If the
file format has DTS, it's easy, otherwise the duration will seemingly
jump back and forth.
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When video format changes, the frame before the frame with the new
format sets video_status briefly to STATUS_DRAINING. This caused the
code to handle the EOF case to kick in, which just pauses the player
when trying to step past the last frame. As a result, trying to
framestep over format changes resulted in pausing the player.
Fix by testing against the correct status.
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This shouldn't change anything functionally.
Change the A/V desync message. --framedrop is enabled by default now, so
the text must be changed a little. I've never heard of audio outputs
messing up A/V sync recently, so remove that part.
Remove the unused ao_pts field.
Reorder 2 A/V sync related expressions so that they look the same.
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The "x " prefix annoyed some users.
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Seems some programs were still relying on it. Whatever, it's not hard to
support.
CC: @mpv-player/stable
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Commit 846257da introduced an accidental feature: if you kept seeking
(so playback never really resumes), the audio would never be played.
This was nice, but commit 4c25b000 accidentally removed it again (due
to the video_next_pts being earlier available than it used to be, so
audio could be played before the player executed the next queued seek).
Implicitly reintroduce the old behavior again by not decoding a second
video frame immediately. Usually, the second frame is used to compute
the frame duration needed to for accurate framedropping, but since the
first frame after a seek is never dropped, we don't need this.
Now the video code will queue the new frame to the VO immediately, and
since fill_audio_out_buffers() is called in the playloop before
write_video() and execute_queued_seek(), it never gets the chance to
enter STATUS_READY, and seeks will be silent.
This also has a nice side-effect: since the second frame is not decoded
and filtered, seeking becomes slightly faster (back to the same level
as with framedrop disabled).
It seems this still sometimes plays a period of audio when keeping a
seek key down. In my tests, this appeared to happen because the seek
finished before the next key repeat was sent.
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In theory, timestamps can be negative, so we shouldn't just return -1
as special value.
Remove the separate code for clearing decode buffers; use the same code
that is used for normal seek reset.
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Now that we use the symbols from the font, we should also
actually use the font.
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Commit 5afc025c broke this. The reason is that mpctx->delay is updated
when a new video frame is added. This value is also needed to resync
audio, but it will be for the wrong PTS. They must be consistent with
each other, and if they aren't, initial sync will be off by N video
frames, which results at least in worse user experience.
This can be reproduced by for example heavily switching between normal
and 2x speed, or similar.
Fix by readding the video_next_pts field (keeping its use minimal,
instead of reverting the commit that removed it).
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If video reaches EOF, and audio is also EOF (or is otherwise not
meaningful, like audio disabled), then the playback position was briefly
set to 0. Fix this by not trying to use a bogus audio PTS.
CC: @mpv-player/stable (maybe)
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This simplifies the code, and fixes an odd bug: the second-last frame
was displayed for a very short duration if framedrop was enabled. The
reason was that basically the time difference between second-last and
last frame were skipped, because at this point EOF was already
signaled. Also see commit b0959488 for a similar issue in the
same code.
This removes the messiness of the next_frame 2-frame queue, and
strictly runs the "new frame" code when a frame is moved to the first
position of the queue, instead of somehow messing with return codes.
This also merges update_video() into video_output_image().
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Not really needed anymore. Code should be mostly equivalent.
Also get rid of some other now-unused or outdated things.
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No functional changes. init_vo() is now needed a bit further down, and
moving it keeps definition and use close. adjust_sync() will be used by
a function further up in one of the following commits.
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This is mostly equivalent, but simpler, and reduces some duplication.
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It was an unintended/accidental change.
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In particular, remove all the stupid debug printfs from the win code.
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For use as playlist navigation button in OSC, now the osd-font
carries all symbols needed by the OSC.
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Although it's probably safe for most VOs, there's no guarantee.
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Signed-off-by: wm4 <wm4@nowhere>
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After a new file is loaded, playback never starts instantly. Rather, it
takes some playloop iterations until initial audio and video have been
decoded, and the outputs have been (lazily) initialized. This means you
will get status line updates between the messages that inform of the
initialized outputs. This is a bit annoying and clutters the terminal
output needlessly.
Fix this by never printing the status line before playback isn't fully
initialized. Do this by reusing the --term-playing-msg code (which
prints a message once playback is initialized). This also makes sure the
status line _is_ shown during playback restart when doing seeks.
It's possible that the change will make the output more confusing if for
some reason is stuck forever initializing playback, but that seems like
an obscure corner case that never happens, so forget about it.
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print_status() is called at a later point anyway (and before sleeping),
so this code has little effect. This code was added in commit a4f7a3df5,
and I can't observe any problems with idle mode anymore.
Now print_status() is called from a single place only, within osd.c.
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This is probably a stupid idea, but it can't be denied that this
actually allows playing video without larger desync, even if video is
too slow.
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This mostly uses the same idea as with vo_vdpau.c, but much simplified.
On X11, it tries to get the display framerate with XF86VM, and limits
the frequency of new video frames against it. Note that this is an old
extension, and is confirmed not to work correctly with multi-monitor
setups. But we're using it because it was already around (it is also
used by vo_vdpau).
This attempts to predict the next vsync event by using the time of the
last frame and the display FPS. Even if that goes completely wrong,
the results are still relatively good.
On other systems, or if the X11 code doesn't return a display FPS, a
framerate of 1000 is assumed. This is infinite for all practical
purposes, and means that only frames which are definitely too late are
dropped. This probably has worse results, but is still useful.
"--framedrop=yes" is basically replaced with "--framedrop=decoder". The
old framedropping mode is kept around, and should perhaps be improved.
Dropping on the decoder level is still useful if decoding itself is too
slow.
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Apparently users prefer this behavior.
It was used for subtitles too, so move the code to calculate the video
offset into a separate function. Seeking also needs to be fixed.
Fixes #1018.
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The OSD is marked as changed, but the core isn't notified and this
doesn't immediately wakeup. (Possibly the OSD code should wakeup the
core instead, but maybe that woudl be overkill.)
Observed when using "mp.use_suspend = false" in the OSC, and pausing,
and moving the mouse pointer out of the window. The last part of the
fade remained visible for longer than intended.
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Code reorganized to make layouts exchangeable
alternative test layout can be tested with
layout=slimbox
in the OSC config
timers are now used to properly animate the fade out when the
player is paused
duplicate seeks are discarded again
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See additions to options.rst.
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wm4
Otto Modinos
Ryan Jacobs
Ben Boeckel
ChrisK2
Andrey Morozov