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.
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.
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.
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).
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.
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
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.
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.
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
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.
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.
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.
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.
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.
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.
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.
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.
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.
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).
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)
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().
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.
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.
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.
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.
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.
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.
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
sub_reset() was called on cycling subtitle tracks and on seeking. Since
we don't want that subtitles disppear on cycling, sd_lavc.c didn't clear
its internal subtitle queue on reset, which meant that seeking with PGS
subtitles could leave the subtitle on screen (PGS subtitles usually
don't have a duration set).
Call it only on seeking, so we can also strictly clear the subtitle
queue in sd_lavc.
(This still can go very wrong if you disable a subtitle, seek, and
enable it again - for example, if used with libavformat that uses "SSA"
style demuxed ASS subtitle packets. That shouldn't happen with newer
libavformat versions, and the user can "correct" it anyway by executing
a seek while the subtitle is selected.)
Until recently, vo_opengl could be accessed from a single thread only,
due to the OpenGL API context being thread-specific. This issue doesn't
exist anymore, because VOs run on their own thread. This means we can
simply lock/unlock the playloop instead of doing something complicated
to get the playloop thread to execute our code.
This ran adjust_sync() on every playloop iteration, instead of every
newly decoded frame. It seems this was idempotent in the common case,
but the code was originally designed to be run once only, so restore
that.
The previous commit broke these things, and fixing them is separate in
this commit in order to reduce the volume of changes.
Move the image queue from the VO to the playback core. The image queue
is a remnant of the old way how vdpau was implemented, and increasingly
became more and more an artifact. In the end, it did only one thing:
computing the duration of the current frame. This was done by taking the
PTS difference between the current and the future frame. We keep this,
but by moving it out of the VO, we don't have to special-case format
changes anymore. This simplifies the code a lot.
Since we need the queue to compute the duration only, a queue size
larger than 2 makes no sense, and we can hardcode that.
Also change how the last frame is handled. The last frame is a bit of a
problem, because video timing works by showing one frame after another,
which makes it a special case. Make the VO provide a function to notify
us when the frame is done, instead. The frame duration is used for that.
This is not perfect. For example, changing playback speed during the
last frame doesn't update the end time. Pausing will not stop the clock
that times the last frame. But I don't think this matters for such a
corner case.
The VO is run inside its own thread. It also does most of video timing.
The playloop hands the image data and a realtime timestamp to the VO,
and the VO does the rest.
In particular, this allows the playloop to do other things, instead of
blocking for video redraw. But if anything accesses the VO during video
timing, it will block.
This also fixes vo_sdl.c event handling; but that is only a side-effect,
since reimplementing the broken way would require more effort.
Also drop --softsleep. In theory, this option helps if the kernel's
sleeping mechanism is too inaccurate for video timing. In practice, I
haven't ever encountered a situation where it helps, and it just burns
CPU cycles. On the other hand it's probably actively harmful, because
it prevents the libavcodec decoder threads from doing real work.
Side note:
Originally, I intended that multiple frames can be queued to the VO. But
this is not done, due to problems with OSD and other certain features.
OSD in particular is simply designed in a way that it can be neither
timed nor copied, so you do have to render it into the video frame
before you can draw the next frame. (Subtitles have no such restriction.
sd_lavc was even updated to fix this.) It seems the right solution to
queuing multiple VO frames is rendering on VO-backed framebuffers, like
vo_vdpau.c does. This requires VO driver support, and is out of scope
of this commit.
As consequence, the VO has a queue size of 1. The existing video queue
is just needed to compute frame duration, and will be moved out in the
next commit.
The function video_decode_and_filter(), called between initializing the
local vf variable and using it, can actually destroy and recreate the
filter. Thus, the vf variable turns into a dangling pointer if that
happens.
Could be observed with: --hwdec=vda --deinterlace=yes --vf=yadif
(Also happens with vdpau/vaapi.)
This code was sending a string to a different thread, and then
deallocated the string shortly after, which means most of the time
the other thread was accessing a dangling pointer.
It's possible that this is the cause for #1002.
Completely useless, and could accidentally be enabled by cycling
framedrop modes. Just get rid of it.
But still allow triggering the old code with --vd-lavc-framedrop, in
case someone asks for it. If nobody does, this new option will be
removed eventually.
Trying to jump chapters in a gile that has no chapters does nothing,
not even show a warning. This is confusing. The reason is that the
"add chapter" command will just bail out completely if the property
is unavailable.
This was because it exited when it couldn't get the property type.
Instead of exiting, just don't enter the code that needs the type.
(I'm not sure when this behavior changed. I consider it a regression.
It was probably caused by changes to the chapter code, which perhaps
started returning UNAVAILABLE instead of OK if there are no chapters.)
The client API exports this state via events already, but maybe it's
better to explicitly provide this property in order to facilitate use on
OSD and similar cases.
This is delayed by 300ms - before that, the status doesn't change. I
feel like it would too annoying if the status line would "flicker" on
normal seek by quickly showing and hiding the indicator.
If this code is not skipped, encoding (or dumping with --ao=pcm) will
attempt to adjust video timing to audio. Since another commit (0cce8fe6)
already avoids writing audio ahead, this didn't slow down encoding to
realtime, but it was still significantly slower.
This change should actually remove all extra sleeping.
This builds but both the libmpv example and the cplayer block infinitely when
building libmpv. That's because we wait inifinitely in `dispatch_sync` as
there's no event loop in the main thread that allows for libdispatch to work..
Whiel we are at it, we should probably investigate how to use mp_dispatch
instead since it is a little lower level and could give us higher control in
building and event loop.
The subtitle timing logic always used the demuxer's start time as video
offset. This made external subtitle files "just work" with file formats
like TS, which usually have a non-0 start time. But it was wrong for
subtitles muxed with the TS, so adjust the time offset explicitly with
external files only.
Although disabling both video and audio is surely an obscure corner
case, it's allowed, and we don't want the demuxer to skip arbitrary
packets.
Basically, make the heuristic for checking interleaved files affect
external files only.
Handle --term-playing-msg at a better place.
Move MPV_EVENT_TICK hack into a separate function. Also add some words
to the client API that you shouldn't use it. (But better leave breaking
it for later.)
Handle --frames and frame_step differently. Remove the mess from the
playloop, and do it after frame display. Give up on the weird semantics
for audio-only mode (they didn't make sense anyway), and adjust the
manpage accordingly.
If seeks take very long, it's better not to freeze up the display.
(This doesn't handle the case when decoding video frames is extremely
slow; just if hr-seek is used, or the demuxer is threaded and blocks on
network I/O.)
Internally, there are two mechanisms which can trigger property
notification as used with "observed" properties in the client API.
The first mechanism associates events with a group of properties that
are potentially changed by a certain event. mp_event_property_change[]
declares these associations, and maps each event to a set of strings.
When an event happens, the set of strings is matched against the list of
observed properties of each client. Make this more efficient by
comparing bitsets of events instead. This way, only a bit-wise "and" is
needed for each observed property. Even better, we can completely skip
clients which have no observed properties that match.
The second mechanism just updates individual properties explicitly by
name. Optimize this by using the property index instead. It would be
nice if we could reuse the first mechanism for the second one, but
there are too many properties to fit into a 64 bit mask.
(Though the limit on 64 events might get us into trouble later...)
Almost nothing was left of it.
The only thing this commit actually removes is support for reading
input commands from stdin. But you can emulate this via:
--input-file=/dev/stdin --input-terminal=no
However, this won't work on Windows. Just use a named pipe.
Playing audio files with embedded cover art broke due to some of the
recent changes. Treat video EOF properly, and don't burn the CPU.
Disable hrseek for video in attached picture mode, since the decoder
will always produce a new image, which makes hrseek never terminate.
Fixes#970.
In encoding mode, the AO pretends to be infinitely fast (it will take
whatever we write, without ever rejecting input). Commit 261506e3 broke
this somehow. It turns out an old hack dealing with this was accidentally
dropped.
This is the hunk of code whose semantics were (partially) dropped:
if (mpctx->d_audio && (mpctx->restart_playback ? !video_left :
ao_untimed(mpctx->ao) && (mpctx->delay <= 0 ||
!video_left)))
{
int status = fill_audio_out_buffers(mpctx, endpts);
// Not at audio stream EOF yet
audio_left = status > -2;
}
This if condition is pretty wild, and it looked like it was pretty much
for audio-only mode, rather than subtle handling for encoding mode.
"Internal" events were added in the previous commits to leverage the
client API property mechanism, without making weird properties public.
But they were sent to clients too (and returned by mpv_wait_event()).
Achieve this by polling. Will be used by the OSC. Basically a bad hack -
but the point is that the mpv core itself is in the best position to
improve this later.
This is perfectly allowed, but was ignored, because it's a corner case.
It doesn't actually wait for other clients to be destroyed, but on the
other hand I think there's no way to have other clients before
initialization.
CC: @mpv-player/stable
Basically move the code from playloop.c to video.c. The new function
write_video() now contains the code that was part of run_playloop().
There are no functional changes, except handling "new_frame_shown"
slightly differently. This is done so that we don't need new a new
MPContext field or a return value for write_video() to signal this
condition. Instead, it's handled indirectly.
This also reduces some code duplication with other parts of the code.
The changfe is mostly cosmetic, although there are also some subtle
changes in behavior. At least one change is that the big desync message
is now printed after every seek.
Frames buffered in filters weren't flushed, so on EOF, the last frames
were dropped, depending on how much filters buffered. Oops.
Test case: "mpv something.jpg --vf=buffer"
In situations when the demuxer reports EOF, but immediately "recovers"
after that and returns new data, it could happen that audio sync was
skipped. Deal with this by actually entering the EOF state, instead of
assuming this will happen later.
Some files have the first audio much later into the video (for whatever
reasons). Instead of appending large amounts of silence to the audio
buffer (and refusing to sync if the audio to append is "too large"),
just wait until enough video has played.
Regression since commit 261506e3. Internally speaking, playback was
often not properly terminated, and the main part of handle_keep_open()
was just executed once, instead of any time the user tries to seek. This
means playback_pts was not set, and the "current time" was determined by
the seek target PTS.
So fix this aspect of video EOF handling, and also remove the now
unnecessary eof_reached field.
The pause check before calling pause_player() is a lazy workaround for
a strange event feedback loop that happens on EOF with --keep-open.
If an imprecise seek is issues while a precise seek is ongoing,
don't wait up to 300ms (herustistic which usually improves user
experience), but instead let it cancel the seek.
Improves responsiveness of the OSC after the previous commit.
Note that we don't do this on "default-precise" seeks, because we
don't know if they're going to be precise or not.
It probably happens relatively often that the first packet (or even the
first N packets) of a stream will fail to decode, but decoding will
eventually succeed at a later point. Before commit 261506e3, this was
handled by an explicit retry loop (although this was also for other
purposes), but with then was changed to abort on the first error. This
makes it impossible to decode some audio streams.
Change this so that errors are ignored for the first 50 packets, which
should make it equivalent to the old code.
If you for example use --audio-file, disable the external track, seek,
and enable the external track again, the playback position of the
external file was off, and you would get major A/V desync. This was
actually supposed to work, but broke at some time ago (probably commit
2b87415f). It didn't work, because it attempted to seek the stream if it
was already selected, which was always true due to
reselect_demux_streams() being called before that.
Fix by putting the initial selection and the seek together.
Seeking in .ts files (and some other formats) is too unreliable, so
there's a separate code path for this case. But it breaks hr-seek.
Maybe hr-seek could actually be enabled in this case if we're careful
enough about timestamp resets, but for now nothing changes.
If the actual PTS is not known yet right after a seek, the "time-pos"
property will just return the seek target PTS. For this purpose, trigger
a change event to make the client API update the "time-pos" and related
properties. (MPV_EVENT_TICK triggers this update.)
Commit 261506e3 made constant seeking feel slower, because a subtle
change in the restart logic makes it now waste time showing another
video frame. The slowdown is about 20%.
(Background: the seek logic explicitly waits until a video frame is
displayed, because this makes it easier for the user to search for
something in the video. Without this logic, the display would freeze
until the user stops giving seek commands.)
Fix this by letting the seek logic issue another seek as soon as the
first video frame is displayed. This will prevent it from showing a
(useless, slow) second frame. Now it seems to be as fast as before the
change.
One side-effect is that the next seek happens after the first video
frame, but _before_ audio is restarted. Seeking is now silent. I guess
this is ok, so we don't do anything about it. Actually, I think whether
this happens is probably random; the seeking logic simply doesn't make
this explicit, so anything can happen.
This commit makes audio decoding non-blocking. If e.g. the network is
too slow the playloop will just go to sleep, instead of blocking until
enough data is available.
For video, this was already done with commit 7083f88c. For audio, it's
unfortunately much more complicated, because the audio decoder was used
in a blocking manner. Large changes are required to get around this.
The whole playback restart mechanism must be turned into a statemachine,
especially since it has close interactions with video restart. Lots of
video code is thus also changed.
(For the record, I don't think switching this code to threads would
make this conceptually easier: the code would still have to deal with
external input while blocked, so these in-between states do get visible
[and thus need to be handled] anyway. On the other hand, it certainly
should be possible to modularize this code a bit better.)
This will probably cause a bunch of regressions.
Actually free the old mmap region when readding an overlay of the same
ID without removing it before. (This is explicitly documented as
working.)
Replace the OSD atomically. Before this commit, the overlays were
removed and then readded to avoid synchronization problems.
Simplify the code: now there is no weird mapping between index and ID.
The OSD sub-bitmap list still needs to be prepared to skip unused IDs
(since each sub-bitmap list entry must be in use), but the code for this
is relatively separated now.
Fixes issue #956.
Currently entries are added after the current playlist element. This is kinda
confusing, more so given that "loadfile append" appends at the end of the
playlist.
Don't return an EOF code if there's still buffered data.
Also, don't call demux_stream_eof() in the playloop. There's probably
nothing wrong with it, but it's cleaner not to use it.
Also give AD_EOF its own value, so that a decoding error doesn't drain
audio by causing an EOF condition.
"loadfile filename append-play" will now always append the file to the
playlist, and if nothing is playing yet, start playback. I don't want to
change the semantics of "append" mode, so a new mode is needed.
Probably fixes issue #950.
This allows using external subtitle files with e.g. transport stream
files that don't start at time 0.
Note that if the .ts file has timestamp resets, everything goes south.
But I guess this was already the case before, unless there are external
subtitle files that also include timestamp resets, which is unlikely.
(On the other hand, you could for example expect that it works with
embedded DVB subtitles, that were somehow captured from the same stream
and use the same timestamps.)
The video flushing logic was broken: if there are no more packets,
decode_image() will feed flush packets to the decoder. Even if an image
was produced, it will return the demuxer EOF state, and since commit
7083f88c, this EOF state is returned to the caller, which is incorrect.
Revert this part of the change, and explicitly check for VD_WAIT (the
bogus change was intended to forward this error code to the caller).
Also, turn the "r < 1" into something equivalent that doesn't rely on
the exact value of VD_EOF. "r < 0" is ok, because at least here, errors
are always negative.
Move a condition somewhere else, which makes it conceptually simpler.
Also, the assignment to full_audio_buffers removed with this commit was
dead, and its value never used.
Fatal errors in the vidoe chain (such as failing to initialize the video
chain) disable video decoding. Restart the playloop, instead of just
continuing the current iteration.
The resulting behavior should be the same, but it gets rid of possible
corner cases.
Broken by commit 1301a907. This commit added demuxer threading, and
changed some other things to make them simpler and more orthogonal. One
of these things was ntofications about streams that appear during
playback. That's an obscure corner case, but the change made handling of
it as natural as normal initialization.
This didn't work for two reasons:
1. When playing an ordered chapters file where the initial segment was
not from the main file, its streams were added to the track list. So
they were printed twice, and switching to the next segment didn't work,
because the right streams were not selected.
2. EDL, CUE, as well as possibly certain Matroska files don't have any
data or tracks in the "main" demuxer, so normally the first segment is
picked for the track list. This was simply broken.
Fix by sprinkling the code with various hacks.
This called demux_flush(), but that doesn't make any sense with an
asynchronously running demuxer. It would just keep reading and add new
packets again. Explicitly pause the demuxer, so that this can't happen.
Also, when flushing, data will be missing, so the decoders should
always be reinitialized, even if the operation fails.
Commit dc00b146, which disables polling by default, missed another
instance of polling: when the player pauses automatically on low cache.
This could lead to apparent freezes when playing network streams.
In my opinion this is not really necessary, since there's only a single
user of update_video(), but others reading this code would probably hate
me for using magic integer values instead of symbolic constants.
This should be a purely cosmetic commit; any changes in behavior are
bugs.
Instead of blocking on the demuxer when reading a packet, let packets be
read asynchronously. Basically, it polls whether a packet is available,
and if not, the playloop goes to sleep until the demuxer thread wakes it
up.
Note that the player will still block for I/O, because audio is still
read synchronously. It's much harder to do the same change for audio
(because of the design of the audio decoding path and especially
initialization), so audio will have to be done later.
Mouse cursor handling, --heartbeat-cmd, and OSD messages basically
relied on polling. For this reason, the playloop always used a small
timeout (not more than 500ms).
Fix these cases, and raise the timeout to 100 seconds. There is no
reason behind this number; for this specific purpose it's as close to
infinity as any other number.
On MS Windows, or if vo_sdl is used, the timeout remains very small.
In these cases the GUI code doesn't do proper event handling in the
first place, and fixing it requires much more effort.
getch2_poll() still does polling, because as far as I'm aware no event-
based way to detect this state change exists.
This adds a thread to the demuxer which reads packets asynchronously.
It will do so until a configurable minimum packet queue size is
reached. (See options.rst additions.)
For now, the thread is disabled by default. There are some corner cases
that have to be fixed, such as fixing cache behavior with webradios.
Note that most interaction with the demuxer is still blocking, so if
e.g. network dies, the player will still freeze. But this change will
make it possible to remove most causes for freezing.
Most of the new code in demux.c actually consists of weird caches to
compensate for thread-safety issues (with the previously single-threaded
design), or to avoid blocking by having to wait on the demuxer thread.
Most of the changes in the player are due to the fact that we must not
access the source stream directly. the demuxer thread already accesses
it, and the stream stuff is not thread-safe.
For timeline stuff (like ordered chapters), we enable the thread for the
current segment only. We also clear its packet queue on seek, so that
the remaining (unconsumed) readahead buffer doesn't waste memory.
Keep in mind that insane subtitles (such as ASS typesetting muxed into
mkv files) will practically disable the readahead, because the total
queue size is considered when checking whether the minimum queue size
was reached.
There was confusion about what should go into audio pts calculation and
what not (mainly due to the audio push thread). This has been fixed by
using the playing - not written - audio pts (which properly takes into
account the ao's buffer), and incrementing the samples count only by the
amount of samples actually taken from the buffer (unfortunately this
now forces us to keep the lock too long for my taste).
Until now, changing the properties showed the VO colorspace parameters
on OSD. This didn't work quite well, because it showed the VO parameters
_before_ the change. This is because at least one video frame with the
new parameters has to be shown, and this doesn't happen right after
changing the property, but a bit later.
Also fix a random typo in unrelated code.
The final goal is all mp_msg calls produce complete lines. We want this
because otherwise, race conditions could corrupt the terminal output,
and it's inconvenient for the client API too. This commit works towards
this goal. There's still code that has this not fixed yet, though.
demux_seek() actually doesn't return seek success. Instead, it fails if
the demuxer is flagged as unseekable (but this is checked explicitly at
the beginning of this function), or if the seek target PTS is
MP_NOPTS_VALUE (which can never happen).
No reason to wait until the audio has been played. This isn't a problem
with gapless audio disabled, and since gapless is now default, this
behavior might be perceived as regression.
CC: @mpv-player/stable
The original goal was just adding backtraces, however making the code
safe (especially wrt. to out of memory Lua errors) was hard. So this
commit also restructures error handling to make it conceptually simpler.
Now all Lua code is run inside a Lua error handling, except the calls
for creating and destroying the Lua context, and calling the wrapper C
function in a safe way.
The new error handling is actually conceptually simpler and more
correct, because you can just call any Lua function at initialization,
without having to worry whwther it throws errors or not.
Unfortunately, Lua 5.2 removes lua_cpcall(), so we have to emulate it.
There isn't any way to emulate it in a way that works the same on 5.1
and 5.2 with the same semantics in error cases, so ifdeffery is needed.
The debug.traceback() function also behaves weirdly differently between
the Lua versions, so its output is not as nice as it could be (empty
extra line).
This should be unneeded, and the packet position is already sufficient
for this case.
Accessing the stream position directly is going to be a problem when the
stream is accessed from another thread later.
DVD and Bluray (and to some extent cdda) require awful hacks all over
the codebase to make them work. The main reason is that they act like
container, but are entirely implemented on the stream layer. The raw
mpeg data resulting from these streams must be "extended" with the
container-like metadata transported via STREAM_CTRLs. The result were
hacks all over demux.c and some higher-level parts.
Add a "disc" pseudo-demuxer, and move all these hacks and special-cases
to it.
It's unlikely that files with multiple audio tracks and with replaygain
actually happen, but this change might help avoid minor corner cases
with later changes.
This also means that the printed size is always rounded to KBs, because
the cache properties are returned in KB. I think this doesn't matter
much. But if it does, the cache properties should probably changed to
return bytes in the first place.
Something like "char *s = ...; isdigit(s[0]);" triggers undefined
behavior, because char can be signed, and thus s[0] can be a negative
value. The is*() functions require unsigned char _or_ EOF. EOF is a
special value outside of unsigned char range, thus the argument to the
is*() functions can't be a char.
This undefined behavior can actually trigger crashes if the
implementation of these functions e.g. uses lookup tables, which are
then indexed with out-of-range values.
Replace all <ctype.h> uses with our own custom mp_is*() functions added
with misc/ctype.h. As a bonus, these functions are locale-independent.
(Although currently, we _require_ C locale for other reasons.)
Also allows it to deal with NULL return values, which currently is not
needed, but may or may not be required at some point in the future (what
if malloc fails).
It's better to keep the logic in one place.
Also drop that a broken config file aborts loading of the player. I
don't see much reason for this, and it inflates the code slightly.
Search $XDG_CONFIG_HOME and $XDG_CONFIG_DIRS for config files.
This also negates the need to have separate user and global variants of
mp_find_config_file()
Closes#864, #109.
Signed-off-by: wm4 <wm4@nowhere>
The main issue was actually that the OSD callback locked the subtitle
decoder, which does not necessarily work, because the OSD code is
already allowed to lock it. The state was already protected by
unsetting the callback (which involes the OSD lock). So, in summary,
this is probably just a cleanup.
Let the VOs draw the OSD on their own, instead of making OSD drawing a
separate VO driver call. Further, let it be the VOs responsibility to
request subtitles with the correct PTS. We also basically allow the VO
to request OSD/subtitles at any time.
OSX changes untested.
Instead of absuing m_option to store the property list, introduce a
separate type for properties. m_option is still used to handle data
types. The property declaration itself now never contains the option
type, and instead it's always queried with M_PROPERTY_GET_TYPE. (This
was already done with some properties, now all properties use it.)
This also fixes that the function signatures did not match the function
type with which these functions were called. They were called as:
int (*)(const m_option_t*, int, void*, void*)
but the actual function signatures were:
int (*)(m_option_t*, int, void*, MPContext *)
Two arguments were mismatched.
This adds one line per property implementation. With additional the
reordering of the parameters, this makes most of the changes in this
commit.
Does anyone actually use this?
For now, update it, because it's the only case left where an option
points to a global variable (and not a struct offset).
While I'm not very fond of "const", it's important for declarations
(it decides whether a symbol is emitted in a read-only or read/write
section). Fix all these cases, so we have writeable global data only
when we really need.
Convert all these commands to properties. (Except tv_last_channel, not
sure what to do with this.) Also, internally, don't access stream
details directly, but dispatch commands with stream ctrls.
Many of the new properties are a bit strange, because they're write-
only. Also remove some OSD output these commands produced, because I
couldn't be bothered to port these.
In general, this makes everything much cleaner, and will also make it
easier to e.g. move the demuxer to its own thread.
Don't bother updating input.conf, but changes.rst documents how old
commands map to the new ones.
Mostly untested, due to lack of hardware.
Basically, this allows gapless playback with similar files (including
the ordered chapter case), while still being robust in general.
The implementation is quite simplistic on purpose, in order to avoid
all the weird corner cases that can occur when creating the filter
chain. The consequence is that it might do not-gapless playback in
more cases when needed, but if that bothers you, you still can use
the normal gapless mode.
Just using "--gapless-audio" or "--gapless-audio=yes" selects the old
mode.
Since redundant wakeups are avoided now, it's easy to miss a wakeup when
creating/setting the pipe/callback after the client API was signalled.
If the client API is signalled, need_wakeup is set to true, and
wakeup_client skips writing to the pipe or calling the client API. That
this can happen is not very obvious to the client API, so trigger a
wakeup right on start in order to remove this special case.
Until now, availability of new log messages (through the mechanism
associated with mpv_request_log_messages()) did not wakeup the client
API properly. Commit 3b7402b5 was basically a hack to improve that
somewhat, but it wasn't a solution.
The main problem is that the client API itself is producing messages, so
the message callback would attempt to lock the client API lock,
resulting in a deadlock. Even if the lock was recursive, we'd run into
lock-order issues.
Solve this by using a separate lock for waiting and wakeup. Also, since
it's a natural addition, avoid redundant wakeups. This means the wakeup
callback as well as the wakeup pipe will be triggered only once until
the next mpv_wait_event() call happens.
This might make the wakeup callback be invoked in a reentrant way for
the first time, for example if a mpv_* function prints to a log. Adjust
the docs accordingly. (Note that non-reentrant beheavior was never
guaranteed - basically the wakeup callback is somewhat dangerous and
inconvenient.)
Also remove some traces of unneeded code. ctx->shutdown for one was
never set, and probably a leftover of an abandoned idea.
A bit verbose, but less misleading. In most cases, the API user probably
actually wants mpv_terminate_destroy() instead, so the less-useful
function shouldn't have a simnpler name anyway.
mpv_destroy() should perhaps better be called mpv_detach(), because it
destroys only the handle, not necessarily the player. The player is only
terminated if a quit command is sent.
This function quits automatically, and additionally waits until the
player is completely destroyed. It removes the possibility that the
player core is still uninitializing, while all client handles are
already destroyed. (Although in practice, the difference is usually not
important.)
Now a negative timeout mean an infinite timeout. This is similar to the
poll() system call. Apparently this is more intuitive and less confusing
than specifying a "very high" value as timeout if you want to wait
forever.
For callers that never pass negative timeouts, nothing changes.
Listening on messages currently uses polling (every time
mpv_wait_event() has no new events, the message buffer is polled and a
message event is possibly created). Improve this situation a bit, and
call the user-supplied wakeup callback.
This will increase the frequency with which the wakeup callback is
called, but the client is already supposed to be able to deal with this
situation. Also, as before, calling mpv_wait_event() from the wakeup
callback is forbidden, so the client can't read new messages from the
callback directly.
The wakeup pipe is written either. Since the wakeup pipe is created
lazily, we can't access the pipe handle without creating a race
condition or a deadlock. (This is actually very silly, since in practice
the race condition won't matter, but for now let's keep it clean.)
If an API user calls mpv_wait_event() and mpv_observe_property() before
mpv_initialize(), it could happen that a property was accessed before
initialization, which is not ok.
This was always intended this way, and even documented in client.h. Due
to an oversight it was never actually implemented.
The intention is that mpv embedded in applications and "real mpv" don't
conflict. An API user can undo this by setting the "config" option to
"yes", if using the user's mpv config is desired.
This simply writes the file name as a comment to the top of the watch later
config file.
It can be useful to the user for determining whether a watch later config file
can be manually removed (e.g. in case the corresponding media file has been
deleted) or not.
Until now, an error was reported only if the command couldn't be parsed.
Attempt to do more fine-grained reporting. This is not necessarily
perfect, but it's an improvement.
This code handles buggy AOs (even if all AOs are bug-free, it's good for
robustness). Move handling of it to the AO feed thread. Now this check
doesn't require magic numbers and does exactly what's it supposed to do.
The i_bps members of the sh_audio and dev_video structs are mostly used
for displaying the average audio and video bitrates. Keeping them in
bits-per-second avoids truncating them to bytes-per-second and changing
them back lateron.
When Lua itself prints errors such as:
Error: [string "mp.defaults"]:387: syntax error near 'function'
It's unclear why the location is prefixed with "string ". And for some
reason, it disappears if you prefix the name with '@'. I suppose this is
done for the sake of luaL_loadstring. With the '@' prefix added, the
output is now:
Error: mp.defaults:387: syntax error near 'function'
These are actually already included in osdep/io.h, but I think it's
cleaner to repeat them in the file where they are actually needed.
(osdep/io.h needs to have them for other reasons.)
Commit e2e450f9 started making use of luaL_register(), but OF COURSE
this function disappeared in Lua 5.2, and was replaced with a 5.2-only
alternative, slightly different mechanism.
So just NIH our own function. This is actually slightly more correct,
since it forces the user to call "require" to actually make the module
visible for builtin C-only modules other than "mp". Fix autoload.lua
accordingly.
We need this only because Lua's stdlib is so scarce. Lua doesn't intend
to include a complete stdlib - they confine themselves to standard C,
both for portability reasons and to keep the code minimal. But standard
C does not provide much either.
It would be possible to use Lua POSIX wrapper libraries, but that would
be a messy (and unobvious) dependency. It's better to implement the
missing functions ourselves, as long as they're small in number.
Stop using it in most places, and prefer STREAM_CTRL_GET_SIZE. The
advantage is that always the correct size will be used. There can be no
doubt anymore whether the end_pos value is outdated (as it happens often
with files that are being downloaded).
Some streams still use end_pos. They don't change size, and it's easier
to emulate STREAM_CTRL_GET_SIZE using end_pos, instead of adding a
STREAM_CTRL_GET_SIZE implementation to these streams.
Make sure int64_t is always used for STREAM_CTRL_GET_SIZE (it was
uint64_t before).
Remove the seek flags mess, and replace them with a seekable flag. Every
stream must set it consistently now, and an assertion in stream.c checks
this. Don't distinguish between streams that can only be forward or
backwards seeked, since we have no such stream types.
stream.start_pos was needed for optical media only, and (apparently) not
for very good reasons. Just get rid of it.
For stream_dvd, we don't need to do anything. Byte seeking was already
removed from it earlier.
For stream_cdda and stream_vcd, emulate the start_pos by offsetting the
stream pos as seen by the rest of mpv.
The bits in discnav.c and loadfile.c were for dealing with the code
seeking back to the start in demux.c. Handle this differently by
assuming the demuxer is always initialized with the stream at start
position, and instead seek back if initializing the demuxer fails.
Remove the --sb option, which worked by modifying stream.start_pos. If
someone really wants this option, it could be added back by creating a
"slice" stream (actually ffmpeg already has such a thing).
Small fixes for the OSC, seektooltip now enabled by default.
Option-parser now moved to separate package, can be used from
other scripts, see DOCS/man/en/lua.rst.
OSC config file location moved to lua-settings/osc.conf
The quit command has an optional argument that is used as exit code.
Extend that to the quit_watch_later command. Actually, unify the
implementations of the two commands.
Requested in #798.
Cover art is treated like video, but is not really video. In one case,
the audio sync code was accidentally still active. Fixes cover art
playback with --ao=null. (This is due to ao_null's latency emulation.
Although it's not very clear whether that is actually correct...)
vo_vdpau currently has a video queue larger than 1 entry, which causes
the video display code to never queue display the video frame. This is
because we consider cover art an endless stream of frames decoded from
the same source packet, and include special logic to actually only
decode and display 1 frame.
Also, make decode_image() also signal EOF in the cover art case.
Some options change from percentages to number of kilobytes; there are
no cache options using percentages anymore.
Raise the default values. The cache is now 25000 kilobytes, although if
your connection is slow enough, the maximum is probably never reached.
(Although all the memory will still be used as seekback-cache.)
Remove the separate --audio-file-cache option, and use the cache default
settings for it.
This is probably a good idea, because it would make it easier for
software embedding mpv to configure the mpv parts, without requiring the
host program to provide explicit mechanisms for this (other than calling
mpv_load_config_file()).
The code paths for setting options by string and by direct "raw" value
were too different, which resulted in some weird code. Make the code
paths closer to each other.
Also, use this to remove the weirdness in the mpv_set_option()
implementation.
Use the time as returned by mp_time_us() for mpthread_cond_timedwait(),
instead of calculating the struct timespec value based on a timeout.
This (probably) makes it easier to wait for a specific deadline.
When the player is paused, and video filters are changed, an exact seek
is executed to refresh the display. Increase the exactness of the seek
in this case; this reuses the code used for frame backstepping.
It might help in cases where seeking is very imprecise, such as with
transport streams.
If a property is notified as changed, and then again (before the change
notification is returned to the client), and the second change is a
sporadic change (i.e. nothing actually changed) and the change
notification was associated with with a data type, it could happen that
a change was "overlooked", because it would detect no change on the
second notification.
This is actually a pretty annoying corner case, due to the annoying way
we do things, so just store both the previously returned _and_ the newly
obtained property value. then we always compare with the user value to
check for a change, excluding any possibility of a missed change.
Note that we don't (can't/shouldn't) care if a value changes, and then
changes back; it's fine if that doesn't generate a notification. This is
due to how property notifications are supposed to be coalesced.
These are now equivalent to combining commands with the "cycle pause" or
"set pause" commands, and thus are not needed anymore. They were also
obscure and undocumented.
mpv supports per-file config files, basically filename+".conf". We use
a static buffer for the new filename, and if that buffer is too small,
we print a warning. This is confusing for e.g. long URLs, so just hide
the warning by default.
Why not dynamically allocate the buffer? Who cares.
